commit ad473b259b0e80a3c6fab8b3ad4eb694947f8be3 Author: John K. Luebs Date: Sun May 27 18:34:36 2012 -0400 Last upstream release diff --git a/Doxyfile b/Doxyfile new file mode 100644 index 0000000..b8d4bd4 --- /dev/null +++ b/Doxyfile @@ -0,0 +1,1200 @@ +# Doxyfile 1.4.3 + +# This file describes the settings to be used by the documentation system +# doxygen (www.doxygen.org) for a project +# +# All text after a hash (#) is considered a comment and will be ignored +# The format is: +# TAG = value [value, ...] +# For lists items can also be appended using: +# TAG += value [value, ...] +# Values that contain spaces should be placed between quotes (" ") + +#--------------------------------------------------------------------------- +# Project related configuration options +#--------------------------------------------------------------------------- + +# The PROJECT_NAME tag is a single word (or a sequence of words surrounded +# by quotes) that should identify the project. + +PROJECT_NAME = proteaAudio + +# The PROJECT_NUMBER tag can be used to enter a project or revision number. +# This could be handy for archiving the generated documentation or +# if some version control system is used. + +PROJECT_NUMBER = v0.6.2 + +# The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute) +# base path where the generated documentation will be put. +# If a relative path is entered, it will be relative to the location +# where doxygen was started. If left blank the current directory will be used. + +OUTPUT_DIRECTORY = doc + +# If the CREATE_SUBDIRS tag is set to YES, then doxygen will create +# 4096 sub-directories (in 2 levels) under the output directory of each output +# format and will distribute the generated files over these directories. +# Enabling this option can be useful when feeding doxygen a huge amount of +# source files, where putting all generated files in the same directory would +# otherwise cause performance problems for the file system. + +CREATE_SUBDIRS = NO + +# The OUTPUT_LANGUAGE tag is used to specify the language in which all +# documentation generated by doxygen is written. Doxygen will use this +# information to generate all constant output in the proper language. +# The default language is English, other supported languages are: +# Brazilian, Catalan, Chinese, Chinese-Traditional, Croatian, Czech, Danish, +# Dutch, Finnish, French, German, Greek, Hungarian, Italian, Japanese, +# Japanese-en (Japanese with English messages), Korean, Korean-en, Norwegian, +# Polish, Portuguese, Romanian, Russian, Serbian, Slovak, Slovene, Spanish, +# Swedish, and Ukrainian. + +OUTPUT_LANGUAGE = English + +# If the BRIEF_MEMBER_DESC tag is set to YES (the default) Doxygen will +# include brief member descriptions after the members that are listed in +# the file and class documentation (similar to JavaDoc). +# Set to NO to disable this. + +BRIEF_MEMBER_DESC = NO + +# If the REPEAT_BRIEF tag is set to YES (the default) Doxygen will prepend +# the brief description of a member or function before the detailed description. +# Note: if both HIDE_UNDOC_MEMBERS and BRIEF_MEMBER_DESC are set to NO, the +# brief descriptions will be completely suppressed. + +REPEAT_BRIEF = YES + +# This tag implements a quasi-intelligent brief description abbreviator +# that is used to form the text in various listings. Each string +# in this list, if found as the leading text of the brief description, will be +# stripped from the text and the result after processing the whole list, is +# used as the annotated text. Otherwise, the brief description is used as-is. +# If left blank, the following values are used ("$name" is automatically +# replaced with the name of the entity): "The $name class" "The $name widget" +# "The $name file" "is" "provides" "specifies" "contains" +# "represents" "a" "an" "the" + +ABBREVIATE_BRIEF = + +# If the ALWAYS_DETAILED_SEC and REPEAT_BRIEF tags are both set to YES then +# Doxygen will generate a detailed section even if there is only a brief +# description. + +ALWAYS_DETAILED_SEC = NO + +# If the INLINE_INHERITED_MEMB tag is set to YES, doxygen will show all +# inherited members of a class in the documentation of that class as if those +# members were ordinary class members. Constructors, destructors and assignment +# operators of the base classes will not be shown. + +INLINE_INHERITED_MEMB = NO + +# If the FULL_PATH_NAMES tag is set to YES then Doxygen will prepend the full +# path before files name in the file list and in the header files. If set +# to NO the shortest path that makes the file name unique will be used. + +FULL_PATH_NAMES = YES + +# If the FULL_PATH_NAMES tag is set to YES then the STRIP_FROM_PATH tag +# can be used to strip a user-defined part of the path. Stripping is +# only done if one of the specified strings matches the left-hand part of +# the path. The tag can be used to show relative paths in the file list. +# If left blank the directory from which doxygen is run is used as the +# path to strip. + +STRIP_FROM_PATH = + +# The STRIP_FROM_INC_PATH tag can be used to strip a user-defined part of +# the path mentioned in the documentation of a class, which tells +# the reader which header file to include in order to use a class. +# If left blank only the name of the header file containing the class +# definition is used. Otherwise one should specify the include paths that +# are normally passed to the compiler using the -I flag. + +STRIP_FROM_INC_PATH = + +# If the SHORT_NAMES tag is set to YES, doxygen will generate much shorter +# (but less readable) file names. This can be useful is your file systems +# doesn't support long names like on DOS, Mac, or CD-ROM. + +SHORT_NAMES = NO + +# If the JAVADOC_AUTOBRIEF tag is set to YES then Doxygen +# will interpret the first line (until the first dot) of a JavaDoc-style +# comment as the brief description. If set to NO, the JavaDoc +# comments will behave just like the Qt-style comments (thus requiring an +# explicit @brief command for a brief description. + +JAVADOC_AUTOBRIEF = NO + +# The MULTILINE_CPP_IS_BRIEF tag can be set to YES to make Doxygen +# treat a multi-line C++ special comment block (i.e. a block of //! or /// +# comments) as a brief description. This used to be the default behaviour. +# The new default is to treat a multi-line C++ comment block as a detailed +# description. Set this tag to YES if you prefer the old behaviour instead. + +MULTILINE_CPP_IS_BRIEF = NO + +# If the DETAILS_AT_TOP tag is set to YES then Doxygen +# will output the detailed description near the top, like JavaDoc. +# If set to NO, the detailed description appears after the member +# documentation. + +DETAILS_AT_TOP = NO + +# If the INHERIT_DOCS tag is set to YES (the default) then an undocumented +# member inherits the documentation from any documented member that it +# re-implements. + +INHERIT_DOCS = YES + +# If member grouping is used in the documentation and the DISTRIBUTE_GROUP_DOC +# tag is set to YES, then doxygen will reuse the documentation of the first +# member in the group (if any) for the other members of the group. By default +# all members of a group must be documented explicitly. + +DISTRIBUTE_GROUP_DOC = NO + +# If the SEPARATE_MEMBER_PAGES tag is set to YES, then doxygen will produce +# a new page for each member. If set to NO, the documentation of a member will +# be part of the file/class/namespace that contains it. + +SEPARATE_MEMBER_PAGES = NO + +# The TAB_SIZE tag can be used to set the number of spaces in a tab. +# Doxygen uses this value to replace tabs by spaces in code fragments. + +TAB_SIZE = 4 + +# This tag can be used to specify a number of aliases that acts +# as commands in the documentation. An alias has the form "name=value". +# For example adding "sideeffect=\par Side Effects:\n" will allow you to +# put the command \sideeffect (or @sideeffect) in the documentation, which +# will result in a user-defined paragraph with heading "Side Effects:". +# You can put \n's in the value part of an alias to insert newlines. + +ALIASES = + +# Set the OPTIMIZE_OUTPUT_FOR_C tag to YES if your project consists of C +# sources only. Doxygen will then generate output that is more tailored for C. +# For instance, some of the names that are used will be different. The list +# of all members will be omitted, etc. + +OPTIMIZE_OUTPUT_FOR_C = NO + +# Set the OPTIMIZE_OUTPUT_JAVA tag to YES if your project consists of Java sources +# only. Doxygen will then generate output that is more tailored for Java. +# For instance, namespaces will be presented as packages, qualified scopes +# will look different, etc. + +OPTIMIZE_OUTPUT_JAVA = NO + +# Set the SUBGROUPING tag to YES (the default) to allow class member groups of +# the same type (for instance a group of public functions) to be put as a +# subgroup of that type (e.g. under the Public Functions section). Set it to +# NO to prevent subgrouping. Alternatively, this can be done per class using +# the \nosubgrouping command. + +SUBGROUPING = YES + +#--------------------------------------------------------------------------- +# Build related configuration options +#--------------------------------------------------------------------------- + +# If the EXTRACT_ALL tag is set to YES doxygen will assume all entities in +# documentation are documented, even if no documentation was available. +# Private class members and static file members will be hidden unless +# the EXTRACT_PRIVATE and EXTRACT_STATIC tags are set to YES + +EXTRACT_ALL = NO + +# If the EXTRACT_PRIVATE tag is set to YES all private members of a class +# will be included in the documentation. + +EXTRACT_PRIVATE = NO + +# If the EXTRACT_STATIC tag is set to YES all static members of a file +# will be included in the documentation. + +EXTRACT_STATIC = NO + +# If the EXTRACT_LOCAL_CLASSES tag is set to YES classes (and structs) +# defined locally in source files will be included in the documentation. +# If set to NO only classes defined in header files are included. + +EXTRACT_LOCAL_CLASSES = YES + +# This flag is only useful for Objective-C code. When set to YES local +# methods, which are defined in the implementation section but not in +# the interface are included in the documentation. +# If set to NO (the default) only methods in the interface are included. + +EXTRACT_LOCAL_METHODS = NO + +# If the HIDE_UNDOC_MEMBERS tag is set to YES, Doxygen will hide all +# undocumented members of documented classes, files or namespaces. +# If set to NO (the default) these members will be included in the +# various overviews, but no documentation section is generated. +# This option has no effect if EXTRACT_ALL is enabled. + +HIDE_UNDOC_MEMBERS = NO + +# If the HIDE_UNDOC_CLASSES tag is set to YES, Doxygen will hide all +# undocumented classes that are normally visible in the class hierarchy. +# If set to NO (the default) these classes will be included in the various +# overviews. This option has no effect if EXTRACT_ALL is enabled. + +HIDE_UNDOC_CLASSES = NO + +# If the HIDE_FRIEND_COMPOUNDS tag is set to YES, Doxygen will hide all +# friend (class|struct|union) declarations. +# If set to NO (the default) these declarations will be included in the +# documentation. + +HIDE_FRIEND_COMPOUNDS = NO + +# If the HIDE_IN_BODY_DOCS tag is set to YES, Doxygen will hide any +# documentation blocks found inside the body of a function. +# If set to NO (the default) these blocks will be appended to the +# function's detailed documentation block. + +HIDE_IN_BODY_DOCS = NO + +# The INTERNAL_DOCS tag determines if documentation +# that is typed after a \internal command is included. If the tag is set +# to NO (the default) then the documentation will be excluded. +# Set it to YES to include the internal documentation. + +INTERNAL_DOCS = NO + +# If the CASE_SENSE_NAMES tag is set to NO then Doxygen will only generate +# file names in lower-case letters. If set to YES upper-case letters are also +# allowed. This is useful if you have classes or files whose names only differ +# in case and if your file system supports case sensitive file names. Windows +# and Mac users are advised to set this option to NO. + +CASE_SENSE_NAMES = NO + +# If the HIDE_SCOPE_NAMES tag is set to NO (the default) then Doxygen +# will show members with their full class and namespace scopes in the +# documentation. If set to YES the scope will be hidden. + +HIDE_SCOPE_NAMES = NO + +# If the SHOW_INCLUDE_FILES tag is set to YES (the default) then Doxygen +# will put a list of the files that are included by a file in the documentation +# of that file. + +SHOW_INCLUDE_FILES = YES + +# If the INLINE_INFO tag is set to YES (the default) then a tag [inline] +# is inserted in the documentation for inline members. + +INLINE_INFO = YES + +# If the SORT_MEMBER_DOCS tag is set to YES (the default) then doxygen +# will sort the (detailed) documentation of file and class members +# alphabetically by member name. If set to NO the members will appear in +# declaration order. + +SORT_MEMBER_DOCS = YES + +# If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the +# brief documentation of file, namespace and class members alphabetically +# by member name. If set to NO (the default) the members will appear in +# declaration order. + +SORT_BRIEF_DOCS = NO + +# If the SORT_BY_SCOPE_NAME tag is set to YES, the class list will be +# sorted by fully-qualified names, including namespaces. If set to +# NO (the default), the class list will be sorted only by class name, +# not including the namespace part. +# Note: This option is not very useful if HIDE_SCOPE_NAMES is set to YES. +# Note: This option applies only to the class list, not to the +# alphabetical list. + +SORT_BY_SCOPE_NAME = NO + +# The GENERATE_TODOLIST tag can be used to enable (YES) or +# disable (NO) the todo list. This list is created by putting \todo +# commands in the documentation. + +GENERATE_TODOLIST = YES + +# The GENERATE_TESTLIST tag can be used to enable (YES) or +# disable (NO) the test list. This list is created by putting \test +# commands in the documentation. + +GENERATE_TESTLIST = YES + +# The GENERATE_BUGLIST tag can be used to enable (YES) or +# disable (NO) the bug list. This list is created by putting \bug +# commands in the documentation. + +GENERATE_BUGLIST = YES + +# The GENERATE_DEPRECATEDLIST tag can be used to enable (YES) or +# disable (NO) the deprecated list. This list is created by putting +# \deprecated commands in the documentation. + +GENERATE_DEPRECATEDLIST= YES + +# The ENABLED_SECTIONS tag can be used to enable conditional +# documentation sections, marked by \if sectionname ... \endif. + +ENABLED_SECTIONS = + +# The MAX_INITIALIZER_LINES tag determines the maximum number of lines +# the initial value of a variable or define consists of for it to appear in +# the documentation. If the initializer consists of more lines than specified +# here it will be hidden. Use a value of 0 to hide initializers completely. +# The appearance of the initializer of individual variables and defines in the +# documentation can be controlled using \showinitializer or \hideinitializer +# command in the documentation regardless of this setting. + +MAX_INITIALIZER_LINES = 30 + +# Set the SHOW_USED_FILES tag to NO to disable the list of files generated +# at the bottom of the documentation of classes and structs. If set to YES the +# list will mention the files that were used to generate the documentation. + +SHOW_USED_FILES = YES + +# If the sources in your project are distributed over multiple directories +# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy +# in the documentation. + +SHOW_DIRECTORIES = YES + +# The FILE_VERSION_FILTER tag can be used to specify a program or script that +# doxygen should invoke to get the current version for each file (typically from the +# version control system). Doxygen will invoke the program by executing (via +# popen()) the command , where is the value of +# the FILE_VERSION_FILTER tag, and is the name of an input file +# provided by doxygen. Whatever the progam writes to standard output +# is used as the file version. See the manual for examples. + +FILE_VERSION_FILTER = + +#--------------------------------------------------------------------------- +# configuration options related to warning and progress messages +#--------------------------------------------------------------------------- + +# The QUIET tag can be used to turn on/off the messages that are generated +# by doxygen. Possible values are YES and NO. If left blank NO is used. + +QUIET = NO + +# The WARNINGS tag can be used to turn on/off the warning messages that are +# generated by doxygen. Possible values are YES and NO. If left blank +# NO is used. + +WARNINGS = YES + +# If WARN_IF_UNDOCUMENTED is set to YES, then doxygen will generate warnings +# for undocumented members. If EXTRACT_ALL is set to YES then this flag will +# automatically be disabled. + +WARN_IF_UNDOCUMENTED = YES + +# If WARN_IF_DOC_ERROR is set to YES, doxygen will generate warnings for +# potential errors in the documentation, such as not documenting some +# parameters in a documented function, or documenting parameters that +# don't exist or using markup commands wrongly. + +WARN_IF_DOC_ERROR = YES + +# This WARN_NO_PARAMDOC option can be abled to get warnings for +# functions that are documented, but have no documentation for their parameters +# or return value. If set to NO (the default) doxygen will only warn about +# wrong or incomplete parameter documentation, but not about the absence of +# documentation. + +WARN_NO_PARAMDOC = NO + +# The WARN_FORMAT tag determines the format of the warning messages that +# doxygen can produce. The string should contain the $file, $line, and $text +# tags, which will be replaced by the file and line number from which the +# warning originated and the warning text. Optionally the format may contain +# $version, which will be replaced by the version of the file (if it could +# be obtained via FILE_VERSION_FILTER) + +WARN_FORMAT = "$file:$line: $text" + +# The WARN_LOGFILE tag can be used to specify a file to which warning +# and error messages should be written. If left blank the output is written +# to stderr. + +WARN_LOGFILE = + +#--------------------------------------------------------------------------- +# configuration options related to the input files +#--------------------------------------------------------------------------- + +# The INPUT tag can be used to specify the files and/or directories that contain +# documented source files. You may enter file names like "myfile.cpp" or +# directories like "/usr/src/myproject". Separate the files or directories +# with spaces. + +INPUT = . readme.txt lua.txt changelog.txt + +# If the value of the INPUT tag contains directories, you can use the +# FILE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp +# and *.h) to filter out the source-files in the directories. If left +# blank the following patterns are tested: +# *.c *.cc *.cxx *.cpp *.c++ *.java *.ii *.ixx *.ipp *.i++ *.inl *.h *.hh *.hxx +# *.hpp *.h++ *.idl *.odl *.cs *.php *.php3 *.inc *.m *.mm + +FILE_PATTERNS = *.h + +# The RECURSIVE tag can be used to turn specify whether or not subdirectories +# should be searched for input files as well. Possible values are YES and NO. +# If left blank NO is used. + +RECURSIVE = NO + +# The EXCLUDE tag can be used to specify files and/or directories that should +# excluded from the INPUT source files. This way you can easily exclude a +# subdirectory from a directory tree whose root is specified with the INPUT tag. + +EXCLUDE = + +# The EXCLUDE_SYMLINKS tag can be used select whether or not files or +# directories that are symbolic links (a Unix filesystem feature) are excluded +# from the input. + +EXCLUDE_SYMLINKS = NO + +# If the value of the INPUT tag contains directories, you can use the +# EXCLUDE_PATTERNS tag to specify one or more wildcard patterns to exclude +# certain files from those directories. + +EXCLUDE_PATTERNS = + +# The EXAMPLE_PATH tag can be used to specify one or more files or +# directories that contain example code fragments that are included (see +# the \include command). + +EXAMPLE_PATH = + +# If the value of the EXAMPLE_PATH tag contains directories, you can use the +# EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp +# and *.h) to filter out the source-files in the directories. If left +# blank all files are included. + +EXAMPLE_PATTERNS = + +# If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be +# searched for input files to be used with the \include or \dontinclude +# commands irrespective of the value of the RECURSIVE tag. +# Possible values are YES and NO. If left blank NO is used. + +EXAMPLE_RECURSIVE = NO + +# The IMAGE_PATH tag can be used to specify one or more files or +# directories that contain image that are included in the documentation (see +# the \image command). + +IMAGE_PATH = + +# The INPUT_FILTER tag can be used to specify a program that doxygen should +# invoke to filter for each input file. Doxygen will invoke the filter program +# by executing (via popen()) the command , where +# is the value of the INPUT_FILTER tag, and is the name of an +# input file. Doxygen will then use the output that the filter program writes +# to standard output. If FILTER_PATTERNS is specified, this tag will be +# ignored. + +INPUT_FILTER = + +# The FILTER_PATTERNS tag can be used to specify filters on a per file pattern +# basis. Doxygen will compare the file name with each pattern and apply the +# filter if there is a match. The filters are a list of the form: +# pattern=filter (like *.cpp=my_cpp_filter). See INPUT_FILTER for further +# info on how filters are used. If FILTER_PATTERNS is empty, INPUT_FILTER +# is applied to all files. + +FILTER_PATTERNS = + +# If the FILTER_SOURCE_FILES tag is set to YES, the input filter (if set using +# INPUT_FILTER) will be used to filter the input files when producing source +# files to browse (i.e. when SOURCE_BROWSER is set to YES). + +FILTER_SOURCE_FILES = NO + +#--------------------------------------------------------------------------- +# configuration options related to source browsing +#--------------------------------------------------------------------------- + +# If the SOURCE_BROWSER tag is set to YES then a list of source files will +# be generated. Documented entities will be cross-referenced with these sources. +# Note: To get rid of all source code in the generated output, make sure also +# VERBATIM_HEADERS is set to NO. + +SOURCE_BROWSER = NO + +# Setting the INLINE_SOURCES tag to YES will include the body +# of functions and classes directly in the documentation. + +INLINE_SOURCES = NO + +# Setting the STRIP_CODE_COMMENTS tag to YES (the default) will instruct +# doxygen to hide any special comment blocks from generated source code +# fragments. Normal C and C++ comments will always remain visible. + +STRIP_CODE_COMMENTS = YES + +# If the REFERENCED_BY_RELATION tag is set to YES (the default) +# then for each documented function all documented +# functions referencing it will be listed. + +REFERENCED_BY_RELATION = YES + +# If the REFERENCES_RELATION tag is set to YES (the default) +# then for each documented function all documented entities +# called/used by that function will be listed. + +REFERENCES_RELATION = YES + +# If the USE_HTAGS tag is set to YES then the references to source code +# will point to the HTML generated by the htags(1) tool instead of doxygen +# built-in source browser. The htags tool is part of GNU's global source +# tagging system (see http://www.gnu.org/software/global/global.html). You +# will need version 4.8.6 or higher. + +USE_HTAGS = NO + +# If the VERBATIM_HEADERS tag is set to YES (the default) then Doxygen +# will generate a verbatim copy of the header file for each class for +# which an include is specified. Set to NO to disable this. + +VERBATIM_HEADERS = YES + +#--------------------------------------------------------------------------- +# configuration options related to the alphabetical class index +#--------------------------------------------------------------------------- + +# If the ALPHABETICAL_INDEX tag is set to YES, an alphabetical index +# of all compounds will be generated. Enable this if the project +# contains a lot of classes, structs, unions or interfaces. + +ALPHABETICAL_INDEX = NO + +# If the alphabetical index is enabled (see ALPHABETICAL_INDEX) then +# the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns +# in which this list will be split (can be a number in the range [1..20]) + +COLS_IN_ALPHA_INDEX = 5 + +# In case all classes in a project start with a common prefix, all +# classes will be put under the same header in the alphabetical index. +# The IGNORE_PREFIX tag can be used to specify one or more prefixes that +# should be ignored while generating the index headers. + +IGNORE_PREFIX = + +#--------------------------------------------------------------------------- +# configuration options related to the HTML output +#--------------------------------------------------------------------------- + +# If the GENERATE_HTML tag is set to YES (the default) Doxygen will +# generate HTML output. + +GENERATE_HTML = YES + +# The HTML_OUTPUT tag is used to specify where the HTML docs will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `html' will be used as the default path. + +HTML_OUTPUT = . + +# The HTML_FILE_EXTENSION tag can be used to specify the file extension for +# each generated HTML page (for example: .htm,.php,.asp). If it is left blank +# doxygen will generate files with .html extension. + +HTML_FILE_EXTENSION = .html + +# The HTML_HEADER tag can be used to specify a personal HTML header for +# each generated HTML page. If it is left blank doxygen will generate a +# standard header. + +HTML_HEADER = header.html + +# The HTML_FOOTER tag can be used to specify a personal HTML footer for +# each generated HTML page. If it is left blank doxygen will generate a +# standard footer. + +HTML_FOOTER = footer.html + +# The HTML_STYLESHEET tag can be used to specify a user-defined cascading +# style sheet that is used by each HTML page. It can be used to +# fine-tune the look of the HTML output. If the tag is left blank doxygen +# will generate a default style sheet. Note that doxygen will try to copy +# the style sheet file to the HTML output directory, so don't put your own +# stylesheet in the HTML output directory as well, or it will be erased! + +HTML_STYLESHEET = protea.css + +# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes, +# files or namespaces will be aligned in HTML using tables. If set to +# NO a bullet list will be used. + +HTML_ALIGN_MEMBERS = YES + +# If the GENERATE_HTMLHELP tag is set to YES, additional index files +# will be generated that can be used as input for tools like the +# Microsoft HTML help workshop to generate a compressed HTML help file (.chm) +# of the generated HTML documentation. + +GENERATE_HTMLHELP = NO + +# If the GENERATE_HTMLHELP tag is set to YES, the CHM_FILE tag can +# be used to specify the file name of the resulting .chm file. You +# can add a path in front of the file if the result should not be +# written to the html output directory. + +CHM_FILE = proteaAudio.chm + +# If the GENERATE_HTMLHELP tag is set to YES, the HHC_LOCATION tag can +# be used to specify the location (absolute path including file name) of +# the HTML help compiler (hhc.exe). If non-empty doxygen will try to run +# the HTML help compiler on the generated index.hhp. + +HHC_LOCATION = + +# If the GENERATE_HTMLHELP tag is set to YES, the GENERATE_CHI flag +# controls if a separate .chi index file is generated (YES) or that +# it should be included in the master .chm file (NO). + +GENERATE_CHI = NO + +# If the GENERATE_HTMLHELP tag is set to YES, the BINARY_TOC flag +# controls whether a binary table of contents is generated (YES) or a +# normal table of contents (NO) in the .chm file. + +BINARY_TOC = NO + +# The TOC_EXPAND flag can be set to YES to add extra items for group members +# to the contents of the HTML help documentation and to the tree view. + +TOC_EXPAND = NO + +# The DISABLE_INDEX tag can be used to turn on/off the condensed index at +# top of each HTML page. The value NO (the default) enables the index and +# the value YES disables it. + +DISABLE_INDEX = NO + +# This tag can be used to set the number of enum values (range [1..20]) +# that doxygen will group on one line in the generated HTML documentation. + +ENUM_VALUES_PER_LINE = 4 + +# If the GENERATE_TREEVIEW tag is set to YES, a side panel will be +# generated containing a tree-like index structure (just like the one that +# is generated for HTML Help). For this to work a browser that supports +# JavaScript, DHTML, CSS and frames is required (for instance Mozilla 1.0+, +# Netscape 6.0+, Internet explorer 5.0+, or Konqueror). Windows users are +# probably better off using the HTML help feature. + +GENERATE_TREEVIEW = NO + +# If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be +# used to set the initial width (in pixels) of the frame in which the tree +# is shown. + +TREEVIEW_WIDTH = 250 + +#--------------------------------------------------------------------------- +# configuration options related to the LaTeX output +#--------------------------------------------------------------------------- + +# If the GENERATE_LATEX tag is set to YES (the default) Doxygen will +# generate Latex output. + +GENERATE_LATEX = NO + +# The LATEX_OUTPUT tag is used to specify where the LaTeX docs will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `latex' will be used as the default path. + +LATEX_OUTPUT = latex + +# The LATEX_CMD_NAME tag can be used to specify the LaTeX command name to be +# invoked. If left blank `latex' will be used as the default command name. + +LATEX_CMD_NAME = latex + +# The MAKEINDEX_CMD_NAME tag can be used to specify the command name to +# generate index for LaTeX. If left blank `makeindex' will be used as the +# default command name. + +MAKEINDEX_CMD_NAME = makeindex + +# If the COMPACT_LATEX tag is set to YES Doxygen generates more compact +# LaTeX documents. This may be useful for small projects and may help to +# save some trees in general. + +COMPACT_LATEX = NO + +# The PAPER_TYPE tag can be used to set the paper type that is used +# by the printer. Possible values are: a4, a4wide, letter, legal and +# executive. If left blank a4wide will be used. + +PAPER_TYPE = a4wide + +# The EXTRA_PACKAGES tag can be to specify one or more names of LaTeX +# packages that should be included in the LaTeX output. + +EXTRA_PACKAGES = + +# The LATEX_HEADER tag can be used to specify a personal LaTeX header for +# the generated latex document. The header should contain everything until +# the first chapter. If it is left blank doxygen will generate a +# standard header. Notice: only use this tag if you know what you are doing! + +LATEX_HEADER = + +# If the PDF_HYPERLINKS tag is set to YES, the LaTeX that is generated +# is prepared for conversion to pdf (using ps2pdf). The pdf file will +# contain links (just like the HTML output) instead of page references +# This makes the output suitable for online browsing using a pdf viewer. + +PDF_HYPERLINKS = NO + +# If the USE_PDFLATEX tag is set to YES, pdflatex will be used instead of +# plain latex in the generated Makefile. Set this option to YES to get a +# higher quality PDF documentation. + +USE_PDFLATEX = NO + +# If the LATEX_BATCHMODE tag is set to YES, doxygen will add the \\batchmode. +# command to the generated LaTeX files. This will instruct LaTeX to keep +# running if errors occur, instead of asking the user for help. +# This option is also used when generating formulas in HTML. + +LATEX_BATCHMODE = NO + +# If LATEX_HIDE_INDICES is set to YES then doxygen will not +# include the index chapters (such as File Index, Compound Index, etc.) +# in the output. + +LATEX_HIDE_INDICES = NO + +#--------------------------------------------------------------------------- +# configuration options related to the RTF output +#--------------------------------------------------------------------------- + +# If the GENERATE_RTF tag is set to YES Doxygen will generate RTF output +# The RTF output is optimized for Word 97 and may not look very pretty with +# other RTF readers or editors. + +GENERATE_RTF = NO + +# The RTF_OUTPUT tag is used to specify where the RTF docs will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `rtf' will be used as the default path. + +RTF_OUTPUT = rtf + +# If the COMPACT_RTF tag is set to YES Doxygen generates more compact +# RTF documents. This may be useful for small projects and may help to +# save some trees in general. + +COMPACT_RTF = NO + +# If the RTF_HYPERLINKS tag is set to YES, the RTF that is generated +# will contain hyperlink fields. The RTF file will +# contain links (just like the HTML output) instead of page references. +# This makes the output suitable for online browsing using WORD or other +# programs which support those fields. +# Note: wordpad (write) and others do not support links. + +RTF_HYPERLINKS = NO + +# Load stylesheet definitions from file. Syntax is similar to doxygen's +# config file, i.e. a series of assignments. You only have to provide +# replacements, missing definitions are set to their default value. + +RTF_STYLESHEET_FILE = + +# Set optional variables used in the generation of an rtf document. +# Syntax is similar to doxygen's config file. + +RTF_EXTENSIONS_FILE = + +#--------------------------------------------------------------------------- +# configuration options related to the man page output +#--------------------------------------------------------------------------- + +# If the GENERATE_MAN tag is set to YES (the default) Doxygen will +# generate man pages + +GENERATE_MAN = NO + +# The MAN_OUTPUT tag is used to specify where the man pages will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `man' will be used as the default path. + +MAN_OUTPUT = man + +# The MAN_EXTENSION tag determines the extension that is added to +# the generated man pages (default is the subroutine's section .3) + +MAN_EXTENSION = .3 + +# If the MAN_LINKS tag is set to YES and Doxygen generates man output, +# then it will generate one additional man file for each entity +# documented in the real man page(s). These additional files +# only source the real man page, but without them the man command +# would be unable to find the correct page. The default is NO. + +MAN_LINKS = NO + +#--------------------------------------------------------------------------- +# configuration options related to the XML output +#--------------------------------------------------------------------------- + +# If the GENERATE_XML tag is set to YES Doxygen will +# generate an XML file that captures the structure of +# the code including all documentation. + +GENERATE_XML = NO + +# The XML_OUTPUT tag is used to specify where the XML pages will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `xml' will be used as the default path. + +XML_OUTPUT = xml + +# The XML_SCHEMA tag can be used to specify an XML schema, +# which can be used by a validating XML parser to check the +# syntax of the XML files. + +XML_SCHEMA = + +# The XML_DTD tag can be used to specify an XML DTD, +# which can be used by a validating XML parser to check the +# syntax of the XML files. + +XML_DTD = + +# If the XML_PROGRAMLISTING tag is set to YES Doxygen will +# dump the program listings (including syntax highlighting +# and cross-referencing information) to the XML output. Note that +# enabling this will significantly increase the size of the XML output. + +XML_PROGRAMLISTING = YES + +#--------------------------------------------------------------------------- +# configuration options for the AutoGen Definitions output +#--------------------------------------------------------------------------- + +# If the GENERATE_AUTOGEN_DEF tag is set to YES Doxygen will +# generate an AutoGen Definitions (see autogen.sf.net) file +# that captures the structure of the code including all +# documentation. Note that this feature is still experimental +# and incomplete at the moment. + +GENERATE_AUTOGEN_DEF = NO + +#--------------------------------------------------------------------------- +# configuration options related to the Perl module output +#--------------------------------------------------------------------------- + +# If the GENERATE_PERLMOD tag is set to YES Doxygen will +# generate a Perl module file that captures the structure of +# the code including all documentation. Note that this +# feature is still experimental and incomplete at the +# moment. + +GENERATE_PERLMOD = NO + +# If the PERLMOD_LATEX tag is set to YES Doxygen will generate +# the necessary Makefile rules, Perl scripts and LaTeX code to be able +# to generate PDF and DVI output from the Perl module output. + +PERLMOD_LATEX = NO + +# If the PERLMOD_PRETTY tag is set to YES the Perl module output will be +# nicely formatted so it can be parsed by a human reader. This is useful +# if you want to understand what is going on. On the other hand, if this +# tag is set to NO the size of the Perl module output will be much smaller +# and Perl will parse it just the same. + +PERLMOD_PRETTY = YES + +# The names of the make variables in the generated doxyrules.make file +# are prefixed with the string contained in PERLMOD_MAKEVAR_PREFIX. +# This is useful so different doxyrules.make files included by the same +# Makefile don't overwrite each other's variables. + +PERLMOD_MAKEVAR_PREFIX = + +#--------------------------------------------------------------------------- +# Configuration options related to the preprocessor +#--------------------------------------------------------------------------- + +# If the ENABLE_PREPROCESSING tag is set to YES (the default) Doxygen will +# evaluate all C-preprocessor directives found in the sources and include +# files. + +ENABLE_PREPROCESSING = YES + +# If the MACRO_EXPANSION tag is set to YES Doxygen will expand all macro +# names in the source code. If set to NO (the default) only conditional +# compilation will be performed. Macro expansion can be done in a controlled +# way by setting EXPAND_ONLY_PREDEF to YES. + +MACRO_EXPANSION = NO + +# If the EXPAND_ONLY_PREDEF and MACRO_EXPANSION tags are both set to YES +# then the macro expansion is limited to the macros specified with the +# PREDEFINED and EXPAND_AS_PREDEFINED tags. + +EXPAND_ONLY_PREDEF = NO + +# If the SEARCH_INCLUDES tag is set to YES (the default) the includes files +# in the INCLUDE_PATH (see below) will be search if a #include is found. + +SEARCH_INCLUDES = YES + +# The INCLUDE_PATH tag can be used to specify one or more directories that +# contain include files that are not input files but should be processed by +# the preprocessor. + +INCLUDE_PATH = + +# You can use the INCLUDE_FILE_PATTERNS tag to specify one or more wildcard +# patterns (like *.h and *.hpp) to filter out the header-files in the +# directories. If left blank, the patterns specified with FILE_PATTERNS will +# be used. + +INCLUDE_FILE_PATTERNS = + +# The PREDEFINED tag can be used to specify one or more macro names that +# are defined before the preprocessor is started (similar to the -D option of +# gcc). The argument of the tag is a list of macros of the form: name +# or name=definition (no spaces). If the definition and the = are +# omitted =1 is assumed. To prevent a macro definition from being +# undefined via #undef or recursively expanded use the := operator +# instead of the = operator. + +PREDEFINED = + +# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then +# this tag can be used to specify a list of macro names that should be expanded. +# The macro definition that is found in the sources will be used. +# Use the PREDEFINED tag if you want to use a different macro definition. + +EXPAND_AS_DEFINED = + +# If the SKIP_FUNCTION_MACROS tag is set to YES (the default) then +# doxygen's preprocessor will remove all function-like macros that are alone +# on a line, have an all uppercase name, and do not end with a semicolon. Such +# function macros are typically used for boiler-plate code, and will confuse +# the parser if not removed. + +SKIP_FUNCTION_MACROS = YES + +#--------------------------------------------------------------------------- +# Configuration::additions related to external references +#--------------------------------------------------------------------------- + +# The TAGFILES option can be used to specify one or more tagfiles. +# Optionally an initial location of the external documentation +# can be added for each tagfile. The format of a tag file without +# this location is as follows: +# TAGFILES = file1 file2 ... +# Adding location for the tag files is done as follows: +# TAGFILES = file1=loc1 "file2 = loc2" ... +# where "loc1" and "loc2" can be relative or absolute paths or +# URLs. If a location is present for each tag, the installdox tool +# does not have to be run to correct the links. +# Note that each tag file must have a unique name +# (where the name does NOT include the path) +# If a tag file is not located in the directory in which doxygen +# is run, you must also specify the path to the tagfile here. + +TAGFILES = + +# When a file name is specified after GENERATE_TAGFILE, doxygen will create +# a tag file that is based on the input files it reads. + +GENERATE_TAGFILE = + +# If the ALLEXTERNALS tag is set to YES all external classes will be listed +# in the class index. If set to NO only the inherited external classes +# will be listed. + +ALLEXTERNALS = NO + +# If the EXTERNAL_GROUPS tag is set to YES all external groups will be listed +# in the modules index. If set to NO, only the current project's groups will +# be listed. + +EXTERNAL_GROUPS = YES + +# The PERL_PATH should be the absolute path and name of the perl script +# interpreter (i.e. the result of `which perl'). + +PERL_PATH = /usr/bin/perl + +#--------------------------------------------------------------------------- +# Configuration options related to the dot tool +#--------------------------------------------------------------------------- + +# If the CLASS_DIAGRAMS tag is set to YES (the default) Doxygen will +# generate a inheritance diagram (in HTML, RTF and LaTeX) for classes with base +# or super classes. Setting the tag to NO turns the diagrams off. Note that +# this option is superseded by the HAVE_DOT option below. This is only a +# fallback. It is recommended to install and use dot, since it yields more +# powerful graphs. + +CLASS_DIAGRAMS = YES + +# If set to YES, the inheritance and collaboration graphs will hide +# inheritance and usage relations if the target is undocumented +# or is not a class. + +HIDE_UNDOC_RELATIONS = YES + +# If you set the HAVE_DOT tag to YES then doxygen will assume the dot tool is +# available from the path. This tool is part of Graphviz, a graph visualization +# toolkit from AT&T and Lucent Bell Labs. The other options in this section +# have no effect if this option is set to NO (the default) + +HAVE_DOT = NO + +# If the CLASS_GRAPH and HAVE_DOT tags are set to YES then doxygen +# will generate a graph for each documented class showing the direct and +# indirect inheritance relations. Setting this tag to YES will force the +# the CLASS_DIAGRAMS tag to NO. + +CLASS_GRAPH = YES + +# If the COLLABORATION_GRAPH and HAVE_DOT tags are set to YES then doxygen +# will generate a graph for each documented class showing the direct and +# indirect implementation dependencies (inheritance, containment, and +# class references variables) of the class with other documented classes. + +COLLABORATION_GRAPH = YES + +# If the GROUP_GRAPHS and HAVE_DOT tags are set to YES then doxygen +# will generate a graph for groups, showing the direct groups dependencies + +GROUP_GRAPHS = YES + +# If the UML_LOOK tag is set to YES doxygen will generate inheritance and +# collaboration diagrams in a style similar to the OMG's Unified Modeling +# Language. + +UML_LOOK = NO + +# If set to YES, the inheritance and collaboration graphs will show the +# relations between templates and their instances. + +TEMPLATE_RELATIONS = NO + +# If the ENABLE_PREPROCESSING, SEARCH_INCLUDES, INCLUDE_GRAPH, and HAVE_DOT +# tags are set to YES then doxygen will generate a graph for each documented +# file showing the direct and indirect include dependencies of the file with +# other documented files. + +INCLUDE_GRAPH = YES + +# If the ENABLE_PREPROCESSING, SEARCH_INCLUDES, INCLUDED_BY_GRAPH, and +# HAVE_DOT tags are set to YES then doxygen will generate a graph for each +# documented header file showing the documented files that directly or +# indirectly include this file. + +INCLUDED_BY_GRAPH = YES + +# If the CALL_GRAPH and HAVE_DOT tags are set to YES then doxygen will +# generate a call dependency graph for every global function or class method. +# Note that enabling this option will significantly increase the time of a run. +# So in most cases it will be better to enable call graphs for selected +# functions only using the \callgraph command. + +CALL_GRAPH = NO + +# If the GRAPHICAL_HIERARCHY and HAVE_DOT tags are set to YES then doxygen +# will graphical hierarchy of all classes instead of a textual one. + +GRAPHICAL_HIERARCHY = YES + +# If the DIRECTORY_GRAPH, SHOW_DIRECTORIES and HAVE_DOT tags are set to YES +# then doxygen will show the dependencies a directory has on other directories +# in a graphical way. The dependency relations are determined by the #include +# relations between the files in the directories. + +DIRECTORY_GRAPH = YES + +# The DOT_IMAGE_FORMAT tag can be used to set the image format of the images +# generated by dot. Possible values are png, jpg, or gif +# If left blank png will be used. + +DOT_IMAGE_FORMAT = png + +# The tag DOT_PATH can be used to specify the path where the dot tool can be +# found. If left blank, it is assumed the dot tool can be found in the path. + +DOT_PATH = + +# The DOTFILE_DIRS tag can be used to specify one or more directories that +# contain dot files that are included in the documentation (see the +# \dotfile command). + +DOTFILE_DIRS = + +# The MAX_DOT_GRAPH_DEPTH tag can be used to set the maximum depth of the +# graphs generated by dot. A depth value of 3 means that only nodes reachable +# from the root by following a path via at most 3 edges will be shown. Nodes +# that lay further from the root node will be omitted. Note that setting this +# option to 1 or 2 may greatly reduce the computation time needed for large +# code bases. Also note that a graph may be further truncated if the graph's +# image dimensions are not sufficient to fit the graph (see MAX_DOT_GRAPH_WIDTH +# and MAX_DOT_GRAPH_HEIGHT). If 0 is used for the depth value (the default), +# the graph is not depth-constrained. + +MAX_DOT_GRAPH_DEPTH = 0 + +# Set the DOT_TRANSPARENT tag to YES to generate images with a transparent +# background. This is disabled by default, which results in a white background. +# Warning: Depending on the platform used, enabling this option may lead to +# badly anti-aliased labels on the edges of a graph (i.e. they become hard to +# read). + +DOT_TRANSPARENT = NO + +# Set the DOT_MULTI_TARGETS tag to YES allow dot to generate multiple output +# files in one run (i.e. multiple -o and -T options on the command line). This +# makes dot run faster, but since only newer versions of dot (>1.8.10) +# support this, this feature is disabled by default. + +DOT_MULTI_TARGETS = NO + +# If the GENERATE_LEGEND tag is set to YES (the default) Doxygen will +# generate a legend page explaining the meaning of the various boxes and +# arrows in the dot generated graphs. + +GENERATE_LEGEND = YES + +# If the DOT_CLEANUP tag is set to YES (the default) Doxygen will +# remove the intermediate dot files that are used to generate +# the various graphs. + +DOT_CLEANUP = YES + +#--------------------------------------------------------------------------- +# Configuration::additions related to the search engine +#--------------------------------------------------------------------------- + +# The SEARCHENGINE tag specifies whether or not a search engine should be +# used. If set to NO the values of all tags below this one will be ignored. + +SEARCHENGINE = NO diff --git a/Makefile b/Makefile new file mode 100644 index 0000000..224c97b --- /dev/null +++ b/Makefile @@ -0,0 +1,103 @@ +#--- generic settings ---------------------------------------------- +# settings for C++ compiler: +C = gcc +CC = g++ +CFLAGS = -O2 -Wall # -D_DEBUG -g +INCDIR = -Irtaudio -Irtaudio/include -I../lua/src + +# linker settings: +LCC = ar +LFLAGS = -rcs +LNAME = libproAudio.a +LIB = $(LNAME) +LIBDIR = + +# settings for optional libSDL backend: +INCDIR += -I../archive/baseCode/include +SDLLIB = -lSDLmain -lSDL +SDLDIR = -L/usr/lib -L../archive/baseCode/lib + +#--- platform specific settings ------------------------------------ +ARCH = $(shell uname -s) +ifeq ($(ARCH),Linux) + LIBS = $(LIBDIR) $(LIB) -lpthread -lasound + LUALIB = -llua -ldl + CFLAGS += -DHAVE_GETTIMEOFDAY -D__LINUX_ALSA__ #-D__LINUX_OSS__ + DLLFLAGS = -fPIC -shared + DLLSUFFIX = .so + EXESUFFIX = + +else + ifeq ($(ARCH),Darwin) # MacOSX + LIBS = $(LIBDIR) $(LIB) + LUALIB = -L/usr/local/lib -llua + CFLAGS += -DHAVE_GETTIMEOFDAY -D__MACOSX_CORE__ + DLLFLAGS = -bundle + DLLSUFFIX = .so + EXESUFFIX = .app + + else # windows, MinGW + LIBS = $(LIBDIR) $(LIB) -lole32 -ldsound -lwinmm -mconsole -s + LUALIB = -L../lua/src -llua51 + SDLLIB = -lmingw32 -lSDLmain -lSDL -mconsole -s + CFLAGS += -D__WINDOWS_DS__ + DLLFLAGS = -shared + DLLSUFFIX = .dll + EXESUFFIX = .exe + endif +endif + +#--- make targets and rules ---------------------------------------- + +# by default, proteaAudio makes use of the included rtAudio backend +rtaudio: $(LNAME) example$(EXESUFFIX) playAudioRt$(EXESUFFIX) + +# the make all target additionally builds the Lua frontend and SDL backend, and therefore has additional dependencies +ALL = $(LNAME) example$(EXESUFFIX) playAudioRt$(EXESUFFIX) proAudioRt$(DLLSUFFIX) playAudioSdl$(EXESUFFIX) +all: $(ALL) + +# static library +OBJ = proAudio.o proAudioRt.o stb_vorbis.o rtaudio/RtAudio.o +$(LNAME) : $(OBJ) + $(LCC) $(LFLAGS) $@ $^ + +# minimal example +example$(EXESUFFIX) : example.o + $(CC) $^ $(LIBS) -o $@ + +# flexible example +playAudioRt$(EXESUFFIX) : playAudioRt.o + $(CC) $^ $(LIBS) -o $@ + +# optional Lua frontend +lua: proAudioRt$(DLLSUFFIX) + +proAudioRt$(DLLSUFFIX): proAudioRt_lua.o + $(CC) -o $@ $(DLLFLAGS) $^ $(LIBS) $(LUALIB) + +# example for optional libSDL backend +sdl: playAudioSdl$(EXESUFFIX) + +playAudioSdl$(EXESUFFIX): playAudioSdl.o proAudio.o proAudioSdl.o stb_vorbis.o + $(CC) $(CFLAGS) $^ $(SDLDIR) $(SDLLIB) -o $@ + +# generic rules +.c.o: + $(C) $(CFLAGS) $(INCDIR) -c $< -o $@ +.cpp.o: + $(CC) $(CFLAGS) $(INCDIR) -c $< -o $@ +clean: + rm -f *.o *~ $(OBJ) $(ALL) + +#--- project specific dependencies --------------------------------- +HDR = proAudio.h proAudioRt.h +playAudioRt.o: playAudioRt.cpp $(HDR) +serverAudioRt.o: serverAudioRt.cpp $(HDR) +proAudioRt_lua.o: proAudioRt_lua.cpp $(HDR) +proAudio.o: proAudio.cpp proAudio.h +proAudioRt.o: proAudioRt.cpp $(HDR) +stb_vorbis.o: stb_vorbis.c +rtaudio/RtAudio.o: rtaudio/RtAudio.cpp rtaudio/RtAudio.h rtaudio/RtError.h +example.o: example.cpp $(HDR) +playAudioSdl.o: playAudioSdl.cpp proAudioSdl.h proAudio.h +proAudioSdl.o: proAudioSdl.cpp proAudioSdl.h proAudio.h diff --git a/build/proAudio.vcproj b/build/proAudio.vcproj new file mode 100644 index 0000000..49c69ff --- /dev/null +++ b/build/proAudio.vcproj @@ -0,0 +1,228 @@ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/build/proteaAudio.sln b/build/proteaAudio.sln new file mode 100644 index 0000000..c388f4f --- /dev/null +++ b/build/proteaAudio.sln @@ -0,0 +1,20 @@ + +Microsoft Visual Studio Solution File, Format Version 9.00 +# Visual C++ Express 2005 +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "proAudio", "proAudio.vcproj", "{8DC73909-327A-45CD-93CA-7BA09993357E}" +EndProject +Global + GlobalSection(SolutionConfigurationPlatforms) = preSolution + Debug|Win32 = Debug|Win32 + Release|Win32 = Release|Win32 + EndGlobalSection + GlobalSection(ProjectConfigurationPlatforms) = postSolution + {8DC73909-327A-45CD-93CA-7BA09993357E}.Debug|Win32.ActiveCfg = Debug|Win32 + {8DC73909-327A-45CD-93CA-7BA09993357E}.Debug|Win32.Build.0 = Debug|Win32 + {8DC73909-327A-45CD-93CA-7BA09993357E}.Release|Win32.ActiveCfg = Release|Win32 + {8DC73909-327A-45CD-93CA-7BA09993357E}.Release|Win32.Build.0 = Release|Win32 + EndGlobalSection + GlobalSection(SolutionProperties) = preSolution + HideSolutionNode = FALSE + EndGlobalSection +EndGlobal diff --git a/changelog.txt b/changelog.txt new file mode 100644 index 0000000..789166b --- /dev/null +++ b/changelog.txt @@ -0,0 +1,7 @@ +/** \page Changelog Changelog + +\section v062 Version 0.6.2 (2009-02-04) +- switch to new rtAudio version 4.0.5 +- additional includes to support also newer versions of gcc/g++ +- proteaAudio zip archives expand to single folders +*/ \ No newline at end of file diff --git a/doc/annotated.html b/doc/annotated.html new file mode 100644 index 0000000..e3cb458 --- /dev/null +++ b/doc/annotated.html @@ -0,0 +1,27 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proteaAudio Class List

Here are the classes, structs, unions and interfaces with brief descriptions: + + + + +
AudioSampleClass representing an audio sample
DeviceAudioAbstract base class for stereo audio mixer/playback devices
DeviceAudioRtRtAudio based stereo audio mixer/playback device
DeviceAudioSdlSDL based stereo audio mixer/playback device
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/changelog.html b/doc/changelog.html new file mode 100644 index 0000000..1e1990f --- /dev/null +++ b/doc/changelog.html @@ -0,0 +1,25 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

Changelog

+Version 0.6.2 (2009-02-04)

+
    +
  • switch to new rtAudio version 4.0.5
  • additional includes to support also newer versions of gcc/g++
  • proteaAudio zip archives expand to single folders
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_audio_sample-members.html b/doc/class_audio_sample-members.html new file mode 100644 index 0000000..7547b01 --- /dev/null +++ b/doc/class_audio_sample-members.html @@ -0,0 +1,43 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

AudioSample Member List

This is the complete list of members for AudioSample, including all inherited members.

+ + + + + + + + + + + + + + + + + + + + + +
AudioSample(unsigned char *data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample)AudioSample [inline]
AudioSample(const AudioSample &source)AudioSample
bitsPerSample() const AudioSample [inline]
bitsPerSample(unsigned short bits)AudioSample
bytesPerSample() const AudioSample [inline]
channels() const AudioSample [inline]
data()AudioSample [inline]
data() const AudioSample [inline]
frames() const AudioSample [inline]
loadWav(const std::string &fname)AudioSample [static]
m_bitsPerSampleAudioSample [protected]
m_channelsAudioSample [protected]
m_dataAudioSample [protected]
m_sampleRateAudioSample [protected]
m_sizeAudioSample [protected]
readWav(FILE *stream, size_t(*readFunc)(void *, size_t, size_t, FILE *))AudioSample [static]
sampleRate() const AudioSample [inline]
size() const AudioSample [inline]
sizeFrame() const AudioSample [inline]
volume(float f)AudioSample
~AudioSample()AudioSample [inline]

+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_audio_sample.html b/doc/class_audio_sample.html new file mode 100644 index 0000000..61facff --- /dev/null +++ b/doc/class_audio_sample.html @@ -0,0 +1,713 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

AudioSample Class Reference

class representing an audio sample +More... +

+#include <proAudio.h> +

+List of all members. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +

Public Member Functions

 AudioSample (unsigned char *data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample)
 AudioSample (const AudioSample &source)
 ~AudioSample ()
unsigned char * data ()
const unsigned char * data () const
unsigned int size () const
unsigned int frames () const
unsigned int sizeFrame () const
unsigned short channels () const
unsigned int sampleRate () const
unsigned short bitsPerSample () const
bool bitsPerSample (unsigned short bits)
unsigned short bytesPerSample () const
void volume (float f)

Static Public Member Functions

static AudioSample * loadWav (const std::string &fname)
static AudioSample * readWav (FILE *stream, size_t(*readFunc)(void *, size_t, size_t, FILE *))

Protected Attributes

unsigned char * m_data
unsigned int m_size
unsigned short m_channels
unsigned int m_sampleRate
unsigned short m_bitsPerSample
+

Detailed Description

+class representing an audio sample +

+

Constructor & Destructor Documentation

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
AudioSample::AudioSample ( unsigned char *  data,
unsigned int  size,
unsigned short  channels,
unsigned int  sampleRate,
unsigned short  bitsPerSample
)  [inline]
+
+ + + + + +
+   + + +

+constructor from memory data +

+

+

+ + + + +
+ + + + + + + + + +
AudioSample::AudioSample ( const AudioSample &  source  ) 
+
+ + + + + +
+   + + +

+copy constructor +

+

+

+ + + + +
+ + + + + + + + +
AudioSample::~AudioSample (   )  [inline]
+
+ + + + + +
+   + + +

+destructor +

+

+

Member Function Documentation

+

+ + + + +
+ + + + + + + + + +
bool AudioSample::bitsPerSample ( unsigned short  bits  ) 
+
+ + + + + +
+   + + +

+converts to a different bit rate, e.g., 8, 16 +

+

+

+ + + + +
+ + + + + + + + +
unsigned short AudioSample::bitsPerSample (   )  const [inline]
+
+ + + + + +
+   + + +

+returns number of bits per mono sample, e.g., 8, 16 +

+

+

+ + + + +
+ + + + + + + + +
unsigned short AudioSample::bytesPerSample (   )  const [inline]
+
+ + + + + +
+   + + +

+returns number of bytes per sample, e.g., 1, 2 +

+

+

+ + + + +
+ + + + + + + + +
unsigned short AudioSample::channels (   )  const [inline]
+
+ + + + + +
+   + + +

+returns number of parallel channels, 1 mono, 2 stereo +

+

+

+ + + + +
+ + + + + + + + +
const unsigned char* AudioSample::data (   )  const [inline]
+
+ + + + + +
+   + + +

+allows reading sample data +

+

+

+ + + + +
+ + + + + + + + +
unsigned char* AudioSample::data (   )  [inline]
+
+ + + + + +
+   + + +

+allows accessing sample data +

+

+

+ + + + +
+ + + + + + + + +
unsigned int AudioSample::frames (   )  const [inline]
+
+ + + + + +
+   + + +

+returns sample size in number of frames +

+

+

+ + + + +
+ + + + + + + + + +
static AudioSample* AudioSample::loadWav ( const std::string &  fname  )  [static]
+
+ + + + + +
+   + + +

+loads a WAV file +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
static AudioSample* AudioSample::readWav ( FILE *  stream,
size_t(*)(void *, size_t, size_t, FILE *)  readFunc
)  [static]
+
+ + + + + +
+   + + +

+reads WAV data from a stream via a function compatible to std::fread +

+

+

+ + + + +
+ + + + + + + + +
unsigned int AudioSample::sampleRate (   )  const [inline]
+
+ + + + + +
+   + + +

+returns number of frames per second, e.g., 44100, 22050 +

+

+

+ + + + +
+ + + + + + + + +
unsigned int AudioSample::size (   )  const [inline]
+
+ + + + + +
+   + + +

+returns sample size in bytes +

+

+

+ + + + +
+ + + + + + + + +
unsigned int AudioSample::sizeFrame (   )  const [inline]
+
+ + + + + +
+   + + +

+returns size of a single frame in bytes +

+

+

+ + + + +
+ + + + + + + + + +
void AudioSample::volume ( float  f  ) 
+
+ + + + + +
+   + + +

+changes volume by given factor +

+

+

Member Data Documentation

+

+ + + + +
+ + + + +
unsigned short AudioSample::m_bitsPerSample [protected]
+
+ + + + + +
+   + + +

+number of bits per sample, e.g., 8, 16 +

+

+

+ + + + +
+ + + + +
unsigned short AudioSample::m_channels [protected]
+
+ + + + + +
+   + + +

+number of parallel channels, 1 mono, 2 stereo +

+

+

+ + + + +
+ + + + +
unsigned char* AudioSample::m_data [protected]
+
+ + + + + +
+   + + +

+stores sample data +

+

+

+ + + + +
+ + + + +
unsigned int AudioSample::m_sampleRate [protected]
+
+ + + + + +
+   + + +

+number of samples per second, e.g., 44100, 22050 +

+

+

+ + + + +
+ + + + +
unsigned int AudioSample::m_size [protected]
+
+ + + + + +
+   + + +

+sample size in bytes +

+

+

The documentation for this class was generated from the following file: +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio-members.html b/doc/class_device_audio-members.html new file mode 100644 index 0000000..bb8bbe7 --- /dev/null +++ b/doc/class_device_audio-members.html @@ -0,0 +1,45 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

DeviceAudio Member List

This is the complete list of members for DeviceAudio, including all inherited members.

+ + + + + + + + + + + + + + + + + + + + + + + +
destroy()DeviceAudio [inline, static]
DeviceAudio()DeviceAudio [protected]
loaderAvailable(const std::string &suffix) const DeviceAudio
loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string &suffix)DeviceAudio
m_freqOutDeviceAudio [protected]
m_volLDeviceAudio [protected]
m_volRDeviceAudio [protected]
mm_loaderDeviceAudio [protected]
s_instanceDeviceAudio [protected, static]
sample(unsigned int handle) const DeviceAudio [inline, virtual]
sampleDestroy(unsigned int sample)=0DeviceAudio [pure virtual]
sampleFromFile(const std::string &filename, float volume=1.0f)DeviceAudio [virtual]
sampleFromMemory(const AudioSample &sample, float volume=1.0f)=0DeviceAudio [pure virtual]
singleton()DeviceAudio [inline, static]
soundActive() const =0DeviceAudio [pure virtual]
soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)=0DeviceAudio [pure virtual]
soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)=0DeviceAudio [pure virtual]
soundStop(unsigned int sound)=0DeviceAudio [pure virtual]
soundStop()=0DeviceAudio [pure virtual]
soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f)=0DeviceAudio [pure virtual]
volume(float left, float right)DeviceAudio [inline]
volume(float leftAndRight)DeviceAudio [inline]
~DeviceAudio()DeviceAudio [inline, protected, virtual]

+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio.html b/doc/class_device_audio.html new file mode 100644 index 0000000..6292198 --- /dev/null +++ b/doc/class_device_audio.html @@ -0,0 +1,919 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

DeviceAudio Class Reference

abstract base class for stereo audio mixer/playback devices +More... +

+#include <proAudio.h> +

+

Inheritance diagram for DeviceAudio: +

+ +DeviceAudioRt +DeviceAudioSdl + +List of all members. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +

Public Member Functions

void volume (float left, float right)
void volume (float leftAndRight)
bool loaderRegister (AudioSample *(*loadFunc)(const std::string &), const std::string &suffix)
bool loaderAvailable (const std::string &suffix) const
virtual unsigned int sampleFromFile (const std::string &filename, float volume=1.0f)
virtual unsigned int sampleFromMemory (const AudioSample &sample, float volume=1.0f)=0
virtual bool sampleDestroy (unsigned int sample)=0
virtual const AudioSample * sample (unsigned int handle) const
virtual unsigned int soundPlay (unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)=0
virtual unsigned int soundLoop (unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)=0
virtual bool soundUpdate (unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f)=0
virtual bool soundStop (unsigned int sound)=0
virtual void soundStop ()=0
virtual unsigned int soundActive () const =0

Static Public Member Functions

static DeviceAudio & singleton ()
static void destroy ()

Protected Member Functions

 DeviceAudio ()
virtual ~DeviceAudio ()

Protected Attributes

unsigned int m_freqOut
float m_volL
float m_volR
std::map< std::string, AudioSample
+*(*)(const std::string &)> 		mm_loader

Static Protected Attributes

static DeviceAudio * s_instance
+

Detailed Description

+abstract base class for stereo audio mixer/playback devices +

+

Constructor & Destructor Documentation

+

+ + + + +
+ + + + + + + + +
DeviceAudio::DeviceAudio (   )  [protected]
+
+ + + + + +
+   + + +

+constructor +

+

+

+ + + + +
+ + + + + + + + +
virtual DeviceAudio::~DeviceAudio (   )  [inline, protected, virtual]
+
+ + + + + +
+   + + +

+destructor +

+

+

Member Function Documentation

+

+ + + + +
+ + + + + + + + +
static void DeviceAudio::destroy (   )  [inline, static]
+
+ + + + + +
+   + + +

+calls the destructor of the singleton object +

+

+

+ + + + +
+ + + + + + + + + +
bool DeviceAudio::loaderAvailable ( const std::string &  suffix  )  const
+
+ + + + + +
+   + + +

+returns true in case a loader for this file type is available +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
bool DeviceAudio::loaderRegister ( AudioSample *(*)(const std::string &)  loadFunc,
const std::string &  suffix
) 
+
+ + + + + +
+   + + +

+registers an audio sample loader function handling a file type identified by suffix +

+The function has to be of type AudioSample * loadXYZ(const std::string & filename).

+

+ + + + +
+ + + + + + + + + +
virtual const AudioSample* DeviceAudio::sample ( unsigned int  handle  )  const [inline, virtual]
+
+ + + + + +
+   + + +

+allows read access to a sample identified by its handle +

+ +

+Reimplemented in DeviceAudioRt.

+

+ + + + +
+ + + + + + + + + +
virtual bool DeviceAudio::sampleDestroy ( unsigned int  sample  )  [pure virtual]
+
+ + + + + +
+   + + +

+deletes a previously created sound sample resource identified by its handle +

+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudio::sampleFromFile ( const std::string &  filename,
float  volume = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+loads a sound sample from file, optionally adjusts volume, returns handle +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudio::sampleFromMemory ( const AudioSample &  sample,
float  volume = 1.0f
)  [pure virtual]
+
+ + + + + +
+   + + +

+converts a sound sample to internal audio format, returns handle +

+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + +
static DeviceAudio& DeviceAudio::singleton (   )  [inline, static]
+
+ + + + + +
+   + + +

+returns singleton object +

+This call is only allowed after a successful precedent creation of an audio device

+

+ + + + +
+ + + + + + + + +
virtual unsigned int DeviceAudio::soundActive (   )  const [pure virtual]
+
+ + + + + +
+   + + +

+returns number of currently active sounds +

+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudio::soundLoop ( unsigned int  sample,
float  volumeL = 1.0f,
float  volumeR = 1.0f,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [pure virtual]
+
+ + + + + +
+   + + +

+plays a specified sound sample continuously and sets its parameters +

+

Parameters:
+ + + + + + +
sample handle of a previously loaded sample
volumeL (optional) left volume
volumeR (optional) right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
a handle to the currently played sound or -1 in case of error
+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudio::soundPlay ( unsigned int  sample,
float  volumeL = 1.0f,
float  volumeR = 1.0f,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [pure virtual]
+
+ + + + + +
+   + + +

+plays a specified sound sample once and sets its parameters +

+

Parameters:
+ + + + + + +
sample handle of a previously loaded sample
volumeL (optional) left volume
volumeR (optional) right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
a handle to the currently played sound or -1 in case of error
+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + +
virtual void DeviceAudio::soundStop (   )  [pure virtual]
+
+ + + + + +
+   + + +

+stops all sounds immediately +

+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + + +
virtual bool DeviceAudio::soundStop ( unsigned int  sound  )  [pure virtual]
+
+ + + + + +
+   + + +

+stops a specified sound immediately +

+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual bool DeviceAudio::soundUpdate ( unsigned int  sound,
float  volumeL,
float  volumeR,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [pure virtual]
+
+ + + + + +
+   + + +

+updates parameters of a specified sound +

+

Parameters:
+ + + + + + +
sound handle of a currently active sound
volumeL left volume
volumeR right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
true in case the parameters have been updated successfully
+ +

+Implemented in DeviceAudioRt, and DeviceAudioSdl.

+

+ + + + +
+ + + + + + + + + +
void DeviceAudio::volume ( float  leftAndRight  )  [inline]
+
+ + + + + +
+   + + +

+sets master volume +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
void DeviceAudio::volume ( float  left,
float  right
)  [inline]
+
+ + + + + +
+   + + +

+sets master volume +

+

+

Member Data Documentation

+

+ + + + +
+ + + + +
unsigned int DeviceAudio::m_freqOut [protected]
+
+ + + + + +
+   + + +

+stores output stream frequency +

+

+

+ + + + +
+ + + + +
float DeviceAudio::m_volL [protected]
+
+ + + + + +
+   + + +

+stores left master volume +

+

+

+ + + + +
+ + + + +
float DeviceAudio::m_volR [protected]
+
+ + + + + +
+   + + +

+stores right master volume +

+

+

+ + + + +
+ + + + +
std::map<std::string, AudioSample * (*)(const std::string &)> DeviceAudio::mm_loader [protected]
+
+ + + + + +
+   + + +

+map associating suffixes to loader functions +

+

+

+ + + + +
+ + + + +
DeviceAudio* DeviceAudio::s_instance [static, protected]
+
+ + + + + +
+   + + +

+pointer to singleton +

+

+

The documentation for this class was generated from the following file: +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio.png b/doc/class_device_audio.png new file mode 100644 index 0000000..9d71e01 Binary files /dev/null and b/doc/class_device_audio.png differ diff --git a/doc/class_device_audio_rt-members.html b/doc/class_device_audio_rt-members.html new file mode 100644 index 0000000..a05d5a0 --- /dev/null +++ b/doc/class_device_audio_rt-members.html @@ -0,0 +1,55 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

DeviceAudioRt Member List

This is the complete list of members for DeviceAudioRt, including all inherited members.

+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
cbMix(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *data)DeviceAudioRt [inline, protected, static]
create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024)DeviceAudioRt [static]
destroy()DeviceAudio [inline, static]
DeviceAudio()DeviceAudio [protected]
DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize)DeviceAudioRt [protected]
loaderAvailable(const std::string &suffix) const DeviceAudio
loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string &suffix)DeviceAudio
m_dacDeviceAudioRt [protected]
m_freqOutDeviceAudio [protected]
m_nSoundDeviceAudioRt [protected]
m_sampleCounterDeviceAudioRt [protected]
m_volLDeviceAudio [protected]
m_volRDeviceAudio [protected]
ma_soundDeviceAudioRt [protected]
mixOutputFloat(signed short *outputBuffer, unsigned int nFrames)DeviceAudioRt [protected]
mm_loaderDeviceAudio [protected]
mm_sampleDeviceAudioRt [protected]
s_instanceDeviceAudio [protected, static]
sample(unsigned int handle) const DeviceAudioRt [virtual]
sampleDestroy(unsigned int sample)DeviceAudioRt [virtual]
sampleFromFile(const std::string &filename, float volume=1.0f)DeviceAudio [virtual]
sampleFromMemory(const AudioSample &sample, float volume=1.0f)DeviceAudioRt [virtual]
singleton()DeviceAudio [inline, static]
soundActive() const DeviceAudioRt [virtual]
soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)DeviceAudioRt [virtual]
soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)DeviceAudioRt [virtual]
soundStop(unsigned int sound)DeviceAudioRt [virtual]
soundStop()DeviceAudioRt [virtual]
soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f)DeviceAudioRt [virtual]
volume(float left, float right)DeviceAudio [inline]
volume(float leftAndRight)DeviceAudio [inline]
~DeviceAudio()DeviceAudio [inline, protected, virtual]
~DeviceAudioRt()DeviceAudioRt [protected, virtual]

+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio_rt.html b/doc/class_device_audio_rt.html new file mode 100644 index 0000000..ad0cb66 --- /dev/null +++ b/doc/class_device_audio_rt.html @@ -0,0 +1,845 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

DeviceAudioRt Class Reference

an rtAudio based stereo audio mixer/playback device +More... +

+#include <proAudioRt.h> +

+

Inheritance diagram for DeviceAudioRt: +

+ +DeviceAudio + +List of all members. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +

Public Member Functions

virtual unsigned int sampleFromMemory (const AudioSample &sample, float volume=1.0f)
virtual bool sampleDestroy (unsigned int sample)
virtual const AudioSample * sample (unsigned int handle) const
virtual unsigned int soundPlay (unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)
virtual unsigned int soundLoop (unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)
virtual bool soundUpdate (unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f)
virtual bool soundStop (unsigned int sound)
virtual void soundStop ()
virtual unsigned soundActive () const

Static Public Member Functions

static DeviceAudio * create (unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024)

Protected Member Functions

 DeviceAudioRt (unsigned int nTracks, unsigned int frequency, unsigned int chunkSize)
virtual ~DeviceAudioRt ()
int mixOutputFloat (signed short *outputBuffer, unsigned int nFrames)

Static Protected Member Functions

static int cbMix (void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *data)

Protected Attributes

std::map< unsigned int, AudioSample * > mm_sample
unsigned int m_sampleCounter
_AudioTrack * ma_sound
unsigned int m_nSound
RtAudio m_dac
+

Detailed Description

+an rtAudio based stereo audio mixer/playback device +

+DeviceAudioRt offers some advanced features such as dynamic pitch, independent volume control for both channels, and user-defined time shifts between the channels. +

+

Constructor & Destructor Documentation

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + +
DeviceAudioRt::DeviceAudioRt ( unsigned int  nTracks,
unsigned int  frequency,
unsigned int  chunkSize
)  [protected]
+
+ + + + + +
+   + + +

+constructor. Use the create() method instead +

+

+

+ + + + +
+ + + + + + + + +
virtual DeviceAudioRt::~DeviceAudioRt (   )  [protected, virtual]
+
+ + + + + +
+   + + +

+destructor. Use the destroy() method instead +

+

+

Member Function Documentation

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
static int DeviceAudioRt::cbMix ( void *  outputBuffer,
void *  inputBuffer,
unsigned int  nFrames,
double  streamTime,
RtAudioStreamStatus  status,
void *  data
)  [inline, static, protected]
+
+ + + + + +
+   + + +

+mixer callback +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + +
static DeviceAudio* DeviceAudioRt::create ( unsigned int  nTracks = 8,
unsigned int  frequency = 22050,
unsigned int  chunkSize = 1024
)  [static]
+
+ + + + + +
+   + + +

+creates audio device +

+Use this method instead of a constructor.

Parameters:
+ + + + +
nTracks (optional) the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor.
frequency (optional) sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor.
chunkSize (optional) the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency.
+
+
Returns:
a pointer to an audio device object in case of success Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called.
+
+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
int DeviceAudioRt::mixOutputFloat ( signed short *  outputBuffer,
unsigned int  nFrames
)  [protected]
+
+ + + + + +
+   + + +

+mixes tracks to a single output stream +

+

+

+ + + + +
+ + + + + + + + + +
virtual const AudioSample* DeviceAudioRt::sample ( unsigned int  handle  )  const [virtual]
+
+ + + + + +
+   + + +

+allows read access to a sample identified by its handle +

+ +

+Reimplemented from DeviceAudio.

+

+ + + + +
+ + + + + + + + + +
virtual bool DeviceAudioRt::sampleDestroy ( unsigned int  sample  )  [virtual]
+
+ + + + + +
+   + + +

+deletes a previously created sound sample resource identified by its handle +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudioRt::sampleFromMemory ( const AudioSample &  sample,
float  volume = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+converts a sound sample to internal audio format, returns handle +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + +
virtual unsigned DeviceAudioRt::soundActive (   )  const [virtual]
+
+ + + + + +
+   + + +

+returns number of currently active sounds +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudioRt::soundLoop ( unsigned int  sample,
float  volumeL = 1.0f,
float  volumeR = 1.0f,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+plays a specified sample continuously and sets its parameters

Parameters:
+ + + + + + +
sample a sample handle returned by a previous load() call
volumeL (optional) left volume
volumeR (optional) right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
a handle to the currently played sound or 0 in case of error
+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudioRt::soundPlay ( unsigned int  sample,
float  volumeL = 1.0f,
float  volumeR = 1.0f,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+plays a specified sample once and sets its parameters +

+

Parameters:
+ + + + + + +
sample a sample handle returned by a previous load() call
volumeL (optional) left volume
volumeR (optional) right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
a handle to the currently played sound or 0 in case of error
+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + +
virtual void DeviceAudioRt::soundStop (   )  [virtual]
+
+ + + + + +
+   + + +

+stops all sounds immediately +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + +
virtual bool DeviceAudioRt::soundStop ( unsigned int  sound  )  [virtual]
+
+ + + + + +
+   + + +

+stops a specified sound immediately +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual bool DeviceAudioRt::soundUpdate ( unsigned int  sound,
float  volumeL,
float  volumeR,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+updates parameters of a specified sound +

+

Parameters:
+ + + + + + +
sound handle of a currently active sound
volumeL left volume
volumeR right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
true in case the parameters have been updated successfully
+ +

+Implements DeviceAudio.

+

Member Data Documentation

+

+ + + + +
+ + + + +
RtAudio DeviceAudioRt::m_dac [protected]
+
+ + + + + +
+   + + +

+audio manager +

+

+

+ + + + +
+ + + + +
unsigned int DeviceAudioRt::m_nSound [protected]
+
+ + + + + +
+   + + +

+stores number of parallel sounds +

+

+

+ + + + +
+ + + + +
unsigned int DeviceAudioRt::m_sampleCounter [protected]
+
+ + + + + +
+   + + +

+stores maximum sample id +

+

+

+ + + + +
+ + + + +
_AudioTrack* DeviceAudioRt::ma_sound [protected]
+
+ + + + + +
+   + + +

+stores sounds to be mixed +

+

+

+ + + + +
+ + + + +
std::map<unsigned int, AudioSample*> DeviceAudioRt::mm_sample [protected]
+
+ + + + + +
+   + + +

+stores loaded sound samples +

+

+

The documentation for this class was generated from the following file: +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio_rt.png b/doc/class_device_audio_rt.png new file mode 100644 index 0000000..6ce25b5 Binary files /dev/null and b/doc/class_device_audio_rt.png differ diff --git a/doc/class_device_audio_sdl-members.html b/doc/class_device_audio_sdl-members.html new file mode 100644 index 0000000..48aef73 --- /dev/null +++ b/doc/class_device_audio_sdl-members.html @@ -0,0 +1,57 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

DeviceAudioSdl Member List

This is the complete list of members for DeviceAudioSdl, including all inherited members.

+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
cbOutput(void *userData, Uint8 *stream, int len)DeviceAudioSdl [protected, static]
create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024)DeviceAudioSdl [static]
destroy()DeviceAudio [inline, static]
DeviceAudio()DeviceAudio [protected]
DeviceAudioSdl(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize)DeviceAudioSdl [protected]
loaderAvailable(const std::string &suffix) const DeviceAudio
loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string &suffix)DeviceAudio
m_freqOutDeviceAudio [protected]
m_isDesiredFormatDeviceAudioSdl [protected]
m_nSoundDeviceAudioSdl [protected]
m_sampleCounterDeviceAudioSdl [protected]
m_specDeviceAudioSdl [protected]
m_volLDeviceAudio [protected]
m_volRDeviceAudio [protected]
ma_soundDeviceAudioSdl [protected]
mixOutputFloat(signed short *outputBuffer, unsigned int nFrames)DeviceAudioSdl [protected]
mixOutputSInt(Uint8 *stream, int len)DeviceAudioSdl [protected]
mm_loaderDeviceAudio [protected]
mm_sampleDeviceAudioSdl [protected]
s_instanceDeviceAudio [protected, static]
sample(unsigned int handle) const DeviceAudio [inline, virtual]
sampleDestroy(unsigned int sample)DeviceAudioSdl [virtual]
sampleFromFile(const std::string &filename, float volume=1.0f)DeviceAudio [virtual]
sampleFromMemory(const AudioSample &sample, float volume=1.0f)DeviceAudioSdl [virtual]
singleton()DeviceAudio [inline, static]
soundActive() const DeviceAudioSdl [virtual]
soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)DeviceAudioSdl [virtual]
soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)DeviceAudioSdl [virtual]
soundStop(unsigned int sound)DeviceAudioSdl [virtual]
soundStop()DeviceAudioSdl [virtual]
soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f)DeviceAudioSdl [virtual]
volume(float left, float right)DeviceAudio [inline]
volume(float leftAndRight)DeviceAudio [inline]
~DeviceAudio()DeviceAudio [inline, protected, virtual]
~DeviceAudioSdl()DeviceAudioSdl [protected, virtual]

+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio_sdl.html b/doc/class_device_audio_sdl.html new file mode 100644 index 0000000..fff6273 --- /dev/null +++ b/doc/class_device_audio_sdl.html @@ -0,0 +1,859 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

DeviceAudioSdl Class Reference

SDL based stereo audio mixer/playback device. +More... +

+#include <proAudioSdl.h> +

+

Inheritance diagram for DeviceAudioSdl: +

+ +DeviceAudio + +List of all members. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +

Public Member Functions

virtual unsigned int sampleFromMemory (const AudioSample &sample, float volume=1.0f)
virtual bool sampleDestroy (unsigned int sample)
virtual unsigned int soundPlay (unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)
virtual unsigned int soundLoop (unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f)
virtual bool soundUpdate (unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f)
virtual bool soundStop (unsigned int sound)
virtual void soundStop ()
virtual unsigned int soundActive () const

Static Public Member Functions

static DeviceAudio * create (unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024)

Protected Member Functions

 DeviceAudioSdl (unsigned int nTracks, unsigned int frequency, unsigned int chunkSize)
virtual ~DeviceAudioSdl ()
void mixOutputFloat (signed short *outputBuffer, unsigned int nFrames)
void mixOutputSInt (Uint8 *stream, int len)

Static Protected Member Functions

static void cbOutput (void *userData, Uint8 *stream, int len)

Protected Attributes

SDL_AudioSpec m_spec
std::map< unsigned int, _AudioTrack > mm_sample
unsigned int m_sampleCounter
bool m_isDesiredFormat
_AudioTrack * ma_sound
unsigned int m_nSound
+

Detailed Description

+SDL based stereo audio mixer/playback device. +

+

Constructor & Destructor Documentation

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + +
DeviceAudioSdl::DeviceAudioSdl ( unsigned int  nTracks,
unsigned int  frequency,
unsigned int  chunkSize
)  [protected]
+
+ + + + + +
+   + + +

+constructor. Use the create() method instead +

+

+

+ + + + +
+ + + + + + + + +
virtual DeviceAudioSdl::~DeviceAudioSdl (   )  [protected, virtual]
+
+ + + + + +
+   + + +

+destructor. Use the destroy() method instead +

+

+

Member Function Documentation

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + +
static void DeviceAudioSdl::cbOutput ( void *  userData,
Uint8 *  stream,
int  len
)  [static, protected]
+
+ + + + + +
+   + + +

+output callback +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + +
static DeviceAudio* DeviceAudioSdl::create ( unsigned int  nTracks = 8,
unsigned int  frequency = 22050,
unsigned int  chunkSize = 1024
)  [static]
+
+ + + + + +
+   + + +

+creates audio device +

+Use this method instead of a constructor.

Parameters:
+ + + + +
nTracks (optional) the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor.
frequency (optional) sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor.
chunkSize (optional) the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency.
+
+
Returns:
a pointer to an audio device object in case of success Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called.
+
+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
void DeviceAudioSdl::mixOutputFloat ( signed short *  outputBuffer,
unsigned int  nFrames
)  [protected]
+
+ + + + + +
+   + + +

+advanced mixer method +

+

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
void DeviceAudioSdl::mixOutputSInt ( Uint8 *  stream,
int  len
)  [protected]
+
+ + + + + +
+   + + +

+fallback mixer method +

+

+

+ + + + +
+ + + + + + + + + +
virtual bool DeviceAudioSdl::sampleDestroy ( unsigned int  sample  )  [virtual]
+
+ + + + + +
+   + + +

+deletes a previously created sound sample resource identified by its handle +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudioSdl::sampleFromMemory ( const AudioSample &  sample,
float  volume = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+converts a sound sample to internal audio format, returns handle +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + +
virtual unsigned int DeviceAudioSdl::soundActive (   )  const [virtual]
+
+ + + + + +
+   + + +

+returns number of currently active sounds +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudioSdl::soundLoop ( unsigned int  sample,
float  volumeL = 1.0f,
float  volumeR = 1.0f,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+plays a specified sample continuously and sets its parameters

Parameters:
+ + + + + + +
sample a sample handle returned by a previous load() call
volumeL (optional) left volume
volumeR (optional) right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
a handle to the currently played sound or 0 in case of error
+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual unsigned int DeviceAudioSdl::soundPlay ( unsigned int  sample,
float  volumeL = 1.0f,
float  volumeR = 1.0f,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+plays a specified sample once and sets its parameters +

+

Parameters:
+ + + + + + +
sample a sample handle returned by a previous load() call
volumeL (optional) left volume
volumeR (optional) right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
a handle to the currently played sound or 0 in case of error
+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + +
virtual void DeviceAudioSdl::soundStop (   )  [virtual]
+
+ + + + + +
+   + + +

+stops all sounds immediately +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + +
virtual bool DeviceAudioSdl::soundStop ( unsigned int  sound  )  [virtual]
+
+ + + + + +
+   + + +

+stops a specified sound immediately +

+ +

+Implements DeviceAudio.

+

+ + + + +
+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
virtual bool DeviceAudioSdl::soundUpdate ( unsigned int  sound,
float  volumeL,
float  volumeR,
float  disparity = 0.0f,
float  pitch = 1.0f
)  [virtual]
+
+ + + + + +
+   + + +

+updates parameters of a specified sound +

+

Parameters:
+ + + + + + +
sound handle of a currently active sound
volumeL left volume
volumeR right volume
disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+
+
Returns:
true in case the parameters have been updated successfully
+ +

+Implements DeviceAudio.

+

Member Data Documentation

+

+ + + + +
+ + + + +
bool DeviceAudioSdl::m_isDesiredFormat [protected]
+
+ + + + + +
+   + + +

+stores whether obtained audio format corresponds to expectations +

+

+

+ + + + +
+ + + + +
unsigned int DeviceAudioSdl::m_nSound [protected]
+
+ + + + + +
+   + + +

+stores number of parallel tracks +

+

+

+ + + + +
+ + + + +
unsigned int DeviceAudioSdl::m_sampleCounter [protected]
+
+ + + + + +
+   + + +

+stores maximum sample id +

+

+

+ + + + +
+ + + + +
SDL_AudioSpec DeviceAudioSdl::m_spec [protected]
+
+ + + + + +
+   + + +

+stores audio specification +

+

+

+ + + + +
+ + + + +
_AudioTrack* DeviceAudioSdl::ma_sound [protected]
+
+ + + + + +
+   + + +

+stores sounds to be mixed +

+

+

+ + + + +
+ + + + +
std::map<unsigned int, _AudioTrack> DeviceAudioSdl::mm_sample [protected]
+
+ + + + + +
+   + + +

+stores loaded sound samples +

+

+

The documentation for this class was generated from the following file: +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/class_device_audio_sdl.png b/doc/class_device_audio_sdl.png new file mode 100644 index 0000000..7c5e126 Binary files /dev/null and b/doc/class_device_audio_sdl.png differ diff --git a/doc/files.html b/doc/files.html new file mode 100644 index 0000000..58310f2 --- /dev/null +++ b/doc/files.html @@ -0,0 +1,26 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proteaAudio File List

Here is a list of all documented files with brief descriptions: + + + +
proAudio.h [code]Public interface of proteaAudio
proAudioRt.h [code]RtAudio backend of proteaAudio
proAudioSdl.h [code]SDL backend of proteaAudio
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/functions.html b/doc/functions.html new file mode 100644 index 0000000..e4b3895 --- /dev/null +++ b/doc/functions.html @@ -0,0 +1,103 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+
All | Functions | Variables
+
a | b | c | d | f | l | m | r | s | v | ~
+ +

+Here is a list of all documented class members with links to the class documentation for each member: +

+

- a -

+

- b -

+

- c -

+

- d -

+

- f -

+

- l -

+

- m -

+

- r -

+

- s -

+

- v -

+

- ~ -

+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/functions_func.html b/doc/functions_func.html new file mode 100644 index 0000000..e055d85 --- /dev/null +++ b/doc/functions_func.html @@ -0,0 +1,86 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+
All | Functions | Variables
+
a | b | c | d | f | l | m | r | s | v | ~
+ +

+ +

+

- a -

+

- b -

+

- c -

+

- d -

+

- f -

+

- l -

+

- m -

+

- r -

+

- s -

+

- v -

+

- ~ -

+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/functions_vars.html b/doc/functions_vars.html new file mode 100644 index 0000000..b1c082c --- /dev/null +++ b/doc/functions_vars.html @@ -0,0 +1,43 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+
All | Functions | Variables
+ +

+

+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/hierarchy.html b/doc/hierarchy.html new file mode 100644 index 0000000..3bdfd0b --- /dev/null +++ b/doc/hierarchy.html @@ -0,0 +1,29 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proteaAudio Class Hierarchy

This inheritance list is sorted roughly, but not completely, alphabetically: +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/index.html b/doc/index.html new file mode 100644 index 0000000..0ef26bb --- /dev/null +++ b/doc/index.html @@ -0,0 +1,52 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proteaAudio Documentation

+

+

v0.6.2

+Overview

+proteaAudio is a free cross-platform 2D audio library for C++ and Lua.

+Due to its straightforward interface and minimal open-source code base, proteaAudio is both easy to maintain and use.

+By default proteaAudio internally makes use of the excellent RtAudio low-level realtime audio input/output API, and therefore has no external dependencies apart from standard system libraries. Together with its liberal open-source licensing conditions (zlib style), this makes the integration of proteaAudio into both free and closed-source commercial software very easy. Alternatively, proteaAudio contains optional bindings for the highly portable SDL multimedia library and is therefore also usable on a plentitude of further platforms (e.g., iPhone, BEOS, FreeBSD).

+Despite its minimalistic design, proteaAudio offers advanced features such as dynamic pitch, per-channel volume control, and user-definable time shifts between channels. proteaAudio is capable of handling normal .wav files as well as ogg/vorbis audio samples (via stb_vorbis). Additionally it offers a simple interface for integrating further custom audio format loaders (e.g., .mp3).

+Main features at a glance

+
    +
  • clean minimal C++ code base, portable, extendable class design
  • supports playback of an arbitrary number of parallel mono/stereo sounds
  • dynamic pitch shifts, panning between the stereo channels, user-definable disparity
  • low CPU consumption, runs in its own dedicated thread
  • regularly tested on MS Windows (MinGW/VisualStudio) and Linux (gcc)
  • loaders for standard wav and ogg/vorbis audio samples included
  • easily extensible to support further audio sample formats, e.g., mp3
  • C++ and Lua API
  • by default makes use of the RtAudio low-level cross-platform audio backend, supporting Windows (DirectSound, ASIO), Linux (ALSA, OSS, JACK), and MacOSX (CoreAudio, JACK)
  • makes optionally use of libSDL as cross-platform audio backend, supporting various further platforms
+

+Download

+ +

+Documentation

+ +

+Links

+proteaAudio internally makes use of the following excellent open-source components:

+

    +
  • RtAudio cross-platform low-level audio library (optional)
  • SDL cross-platform multimedia layer (optional)
  • stb_vorbis Ogg Vorbis audio decoder
+

+License notice (zlib license):

+(c) 2009 by Gerald Franz, www.viremo.de

+This software is provided 'as-is', without any express or implied warranty. In no event will the author be held liable for any damages arising from the use of this software.

+Permission is granted to anyone to use this software for any purpose, including commercial applications, and to alter it and redistribute it freely, subject to the following restrictions:

+

    +
  1. The origin of this software must not be misrepresented; you must not claim that you wrote the original software. If you use this software in a product, an acknowledgment in the product documentation would be appreciated but is not required.
  2. Altered source versions must be plainly marked as such, and must not be misrepresented as being the original software.
  3. This notice may not be removed or altered from any source distribution.
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pages.html b/doc/pages.html new file mode 100644 index 0000000..dde7785 --- /dev/null +++ b/doc/pages.html @@ -0,0 +1,27 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proteaAudio Related Pages

Here is a list of all related documentation pages: +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pro_audio_8h-source.html b/doc/pro_audio_8h-source.html new file mode 100644 index 0000000..65219a3 --- /dev/null +++ b/doc/pro_audio_8h-source.html @@ -0,0 +1,100 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proAudio.h

Go to the documentation of this file.
00001 #ifndef _PRO_AUDIO
+00002 #define _PRO_AUDIO
+00003 
+00004 #include <string>
+00005 #include <map>
+00006 
+00036 //--- class AudioSample --------------------------------------------
+00038 class AudioSample {
+00039 public:
+00041     AudioSample(unsigned char * data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample) :
+00042         m_data(data), m_size(size), m_channels(channels), m_sampleRate(sampleRate), m_bitsPerSample(bitsPerSample) { }
+00044     AudioSample(const AudioSample & source);
+00046     ~AudioSample() { delete[] m_data; }
+00047     
+00049     unsigned char * data() { return m_data; };
+00051     const unsigned char * data() const { return m_data; };
+00053     unsigned int size() const { return m_size; }
+00055     unsigned int frames() const { return m_size/m_channels/(m_bitsPerSample>>3); }
+00057     unsigned int sizeFrame() const { return m_channels*(m_bitsPerSample>>3); }
+00059     unsigned short channels() const { return m_channels; }
+00061     unsigned int sampleRate() const { return m_sampleRate; }
+00063     unsigned short bitsPerSample() const { return m_bitsPerSample; }
+00065     bool bitsPerSample(unsigned short bits);
+00067     unsigned short bytesPerSample() const { return m_bitsPerSample>>3; }
+00068 
+00070     void volume(float f);
+00071     
+00073     static AudioSample* loadWav(const std::string & fname);
+00075     static AudioSample* readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *));
+00076 protected:
+00078     unsigned char * m_data;
+00080     unsigned int m_size;
+00082     unsigned short m_channels;
+00084     unsigned int m_sampleRate;
+00086     unsigned short m_bitsPerSample;
+00087 };
+00088 
+00089 //--- class DeviceAudio --------------------------------------------
+00090 
+00092 class DeviceAudio {
+00093 public:
+00095 
+00096     static DeviceAudio& singleton() { return *s_instance; }
+00098     static void destroy() { if(s_instance) delete s_instance; s_instance=0; };
+00099 
+00101     void volume(float left, float right) { m_volL=left; m_volR=right; }
+00103     void volume(float leftAndRight) { m_volL=m_volR=leftAndRight; }
+00105 
+00106     bool loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix);
+00108     bool loaderAvailable(const std::string & suffix) const;
+00109 
+00111     virtual unsigned int sampleFromFile(const std::string & filename, float volume=1.0f);
+00113     virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f)=0;
+00115     virtual bool sampleDestroy(unsigned int sample)=0;
+00117     virtual const AudioSample* sample(unsigned int handle) const { return 0; }
+00118     
+00120 
+00126     virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;
+00128 
+00134     virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0;
+00136 
+00142     virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f )=0;
+00144     virtual bool soundStop(unsigned int sound)=0;
+00146     virtual void soundStop()=0;
+00148     virtual unsigned int soundActive() const=0;
+00149 
+00150 protected:
+00152     DeviceAudio();
+00154     virtual ~DeviceAudio() { s_instance = 0; }
+00155     
+00157     unsigned int m_freqOut;
+00159     float m_volL;
+00161     float m_volR;
+00163     std::map<std::string, AudioSample * (*)(const std::string &)> mm_loader;
+00164     
+00166     static DeviceAudio * s_instance;
+00167 };
+00168 
+00169 #endif // _PRO_AUDIO
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pro_audio_8h.html b/doc/pro_audio_8h.html new file mode 100644 index 0000000..196b72b --- /dev/null +++ b/doc/pro_audio_8h.html @@ -0,0 +1,47 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proAudio.h File Reference

Public interface of proteaAudio. More... +

+#include <string>
+#include <map>
+ +

+Go to the source code of this file. + + + + + + + + +

Classes

class  AudioSample
 class representing an audio sample More...
class  DeviceAudio
 abstract base class for stereo audio mixer/playback devices More...
+

Detailed Description

+Public interface of proteaAudio. +

+Contains the declaration of the audio sample class and the abstract base class for audio mixer/playback devices

+

Author:
Gerald Franz, www.viremo.de
+
Version:
0.6
+License notice (zlib license):

+(c) 2009 by Gerald Franz, www.viremo.de

+This software is provided 'as-is', without any express or implied warranty. In no event will the author be held liable for any damages arising from the use of this software.

+Permission is granted to anyone to use this software for any purpose, including commercial applications, and to alter it and redistribute it freely, subject to the following restrictions:

+1. The origin of this software must not be misrepresented; you must not claim that you wrote the original software. If you use this software in a product, an acknowledgment in the product documentation would be appreciated but is not required. 2. Altered source versions must be plainly marked as such, and must not be misrepresented as being the original software. 3. This notice may not be removed or altered from any source distribution.

+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pro_audio_rt_8h-source.html b/doc/pro_audio_rt_8h-source.html new file mode 100644 index 0000000..5a23e3b --- /dev/null +++ b/doc/pro_audio_rt_8h-source.html @@ -0,0 +1,60 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proAudioRt.h

Go to the documentation of this file.
00001 #include "proAudio.h"
+00002 #include <RtAudio.h>
+00003 #include <map>
+00004 
+00011 struct _AudioTrack;
+00012 
+00014 
+00016 class DeviceAudioRt : public DeviceAudio {
+00017 public:
+00019 
+00026     static DeviceAudio* create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024);
+00027 
+00029     virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f);
+00031     virtual bool sampleDestroy(unsigned int sample);
+00033     virtual const AudioSample* sample(unsigned int handle) const;
+00034 
+00036 
+00042     virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );
+00050     virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );
+00052 
+00058     virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f );
+00060     virtual bool soundStop(unsigned int sound);
+00062     virtual void soundStop();
+00064     virtual unsigned soundActive() const;
+00065 protected:
+00067     DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize);
+00069     virtual ~DeviceAudioRt();
+00071     int mixOutputFloat(signed short *outputBuffer, unsigned int nFrames);
+00072 
+00074     std::map<unsigned int, AudioSample*> mm_sample;
+00076     unsigned int m_sampleCounter;
+00077 
+00079     _AudioTrack * ma_sound;
+00081     unsigned int m_nSound;
+00083     RtAudio m_dac;
+00084 
+00086     static int cbMix(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *data) {
+00087         return static_cast<DeviceAudioRt*>(data)->mixOutputFloat((signed short*)outputBuffer, nFrames); }
+00088 };
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pro_audio_rt_8h.html b/doc/pro_audio_rt_8h.html new file mode 100644 index 0000000..537dab3 --- /dev/null +++ b/doc/pro_audio_rt_8h.html @@ -0,0 +1,40 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proAudioRt.h File Reference

RtAudio backend of proteaAudio. More... +

+#include "proAudio.h"
+#include <RtAudio.h>
+#include <map>
+ +

+Go to the source code of this file. + + + + + +

Classes

class  DeviceAudioRt
 an rtAudio based stereo audio mixer/playback device More...
+

Detailed Description

+RtAudio backend of proteaAudio. +

+

Author:
Gerald Franz, www.viremo.de
+
Version:
0.6
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pro_audio_sdl_8h-source.html b/doc/pro_audio_sdl_8h-source.html new file mode 100644 index 0000000..72da0ac --- /dev/null +++ b/doc/pro_audio_sdl_8h-source.html @@ -0,0 +1,67 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proAudioSdl.h

Go to the documentation of this file.
00001 #ifndef _AUDIO_SDL_H
+00002 #define _AUDIO_SDL_H
+00003 
+00004 extern "C" {
+00005 #include <SDL_audio.h>
+00006 };
+00007 #include "proAudio.h"
+00008 #include <map>
+00009 
+00016 //--- class DeviceAudioSdl -----------------------------------------
+00017 
+00019 class _AudioTrack;
+00020 
+00022 class DeviceAudioSdl : public DeviceAudio {
+00023 public:
+00025 
+00032     static DeviceAudio* create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024);
+00033 
+00035     virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f);
+00037     virtual bool sampleDestroy(unsigned int sample);
+00038 
+00040 
+00046     virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );
+00054     virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f );
+00056 
+00062     virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f );
+00064     virtual bool soundStop(unsigned int sound);
+00066     virtual void soundStop();
+00068     virtual unsigned int soundActive() const;
+00069 protected:
+00071     DeviceAudioSdl(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize);
+00073     virtual ~DeviceAudioSdl();
+00075     SDL_AudioSpec m_spec;
+00077     std::map<unsigned int, _AudioTrack> mm_sample;
+00079     unsigned int m_sampleCounter;
+00081     bool m_isDesiredFormat;
+00082 
+00084     _AudioTrack * ma_sound;
+00086     unsigned int m_nSound;
+00087 
+00089     static void cbOutput(void *userData, Uint8 *stream, int len);
+00091     void mixOutputFloat(signed short *outputBuffer, unsigned int nFrames);
+00093     void mixOutputSInt(Uint8 *stream, int len);
+00094 };
+00095 
+00096 #endif // _AUDIO_SDL_H
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/pro_audio_sdl_8h.html b/doc/pro_audio_sdl_8h.html new file mode 100644 index 0000000..0c64be7 --- /dev/null +++ b/doc/pro_audio_sdl_8h.html @@ -0,0 +1,40 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proAudioSdl.h File Reference

SDL backend of proteaAudio. More... +

+#include <SDL_audio.h>
+#include "proAudio.h"
+#include <map>
+ +

+Go to the source code of this file. + + + + + +

Classes

class  DeviceAudioSdl
 SDL based stereo audio mixer/playback device. More...
+

Detailed Description

+SDL backend of proteaAudio. +

+

Author:
Gerald Franz, www.viremo.de
+
Version:
2.0
+
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/doc/protea.css b/doc/protea.css new file mode 100644 index 0000000..e4213df --- /dev/null +++ b/doc/protea.css @@ -0,0 +1,116 @@ + body { + font-family: Helvetica,sans-serif; + color: black; + background-color: white; + margin: 30px 50px 50px 50px; + text-align: left; + width: 640px; + font-size: 90%; +} + +h1, h2, h3, h4, h5, h6 { + color: maroon; +} + + h1 { + font-size: 150%; + } + + h2 { + margin: 25px 0px 10px 0px; + font-size: 110%; + } + + a:link, a:visited, a:link:active { + text-decoration: none; + color: maroon; + } + + a:link:hover, a:visited:hover { + text-decoration: underline; + color: red; + } + + p, ul { + font-size: 90%; + } + + pre { + background-color: #DDD; + width: 100%; + font-weight: bold; + } + + + +CAPTION { font-weight: bold } +DIV.qindex { + font-size: 90%; + width: 100%; + border-bottom: 1px solid #b0b0b0; + margin: 2px; + padding: 2px; + line-height: 140%; +} + +TD { + font-size: 90%; +} +TABLE.mdtable { + background-color: #DDD; + width: 100%; +} +TD.md { + font-weight: bold; +} +TD.mdPrefix { + color: #606060; +} +TD.mdname { + font-weight: bold; + color: #602020; +} + +DIV.groupHeader { + margin-left: 16px; + margin-top: 12px; + margin-bottom: 6px; + font-weight: bold; +} +DIV.groupText { margin-left: 16px; font-style: italic; font-size: 90% } +TD.indexkey { + font-weight: bold; + padding-right : 10px; + padding-top : 2px; + padding-left : 10px; + padding-bottom : 2px; + margin-left : 0px; + margin-right : 0px; + margin-top : 2px; + margin-bottom : 2px; + border-top: 1px solid #CCCCCC; + border-bottom: 1px solid #CCCCCC; +} +TD.indexvalue { + font-style: italic; + padding-right : 10px; + padding-top : 2px; + padding-left : 10px; + padding-bottom : 2px; + margin-left : 0px; + margin-right : 0px; + margin-top : 2px; + margin-bottom : 2px; + border-top: 1px solid #CCCCCC; + border-bottom: 1px solid #CCCCCC; +} +TR.memlist { + background-color: #DDD; +} +SPAN.keyword { color: #008000 } +SPAN.keywordtype { color: #604020 } +SPAN.keywordflow { color: #e08000 } +SPAN.comment { color: #800000 } +SPAN.preprocessor { color: #806020 } +SPAN.stringliteral { color: #002080 } +SPAN.charliteral { color: #008080 } diff --git a/doc/proteaAudio.png b/doc/proteaAudio.png new file mode 100644 index 0000000..c7f3a55 Binary files /dev/null and b/doc/proteaAudio.png differ diff --git a/doc/proteaaudiolua.html b/doc/proteaaudiolua.html new file mode 100644 index 0000000..2bcc532 --- /dev/null +++ b/doc/proteaaudiolua.html @@ -0,0 +1,112 @@ + + +proteaAudio + + +
+ +
Main Page | Class Hierarchy | Class List | File List | Class Members | Related Pages
+

proteaAudio Lua API

+Overview

+Lua is a lightweight yet powerful dynamic language. proteaAudio contains almost complete Lua bindings that widely correspond to the C++ API. By default, the proteaAudio Lua bindings are compiled to a dynamic library based on the RtAudio backend which is imported via the following statement:

+

require("proAudioRt")

+All API calls are collected in the global table proAudio. Therefore, proteaAudio is initialized (using the default parameters) by the following call:

+

proAudio.create()

+After that the individual methods and functions may be called analogous to the C++ interface of class DeviceAudio.

+Functions

+
proAudio.create( tracks = 8, frequency = 22050, chunkSize = 1024 )
initializes audio playback device

+Parameters:

    +
  • tracks (optional) number of sounds that can be played in parallel
  • frequency (optional) playback frequency
  • chunkSize (optional) size of the internal buffer in bytes. Note that a small chunkSize results in low playback latency, but may cause computational overhead and hick-ups under higher system load
+

+Returns: true in case the device initialization was successful

+

proAudio.destroy( )
closes audio device and terminates playback

+Returns: true in case the device was successfully closed

+

proAudio.loaderAvailable ( suffix )
returns true in case a loader for this file type is available

+

proAudio.volume 	(  left, [ right ] )
sets master volume, either for both channels uniformly, or individually

+

proAudio.sleep( seconds )
Suspends the execution of the current thread for a definable number of seconds. Note that audio mixing and playback runs in its own background thread and is therefore not affected by this auxiliary call.

+

proAudio.sampleFromFile (  filename, volume = 1.0 )
loads a sound sample from file, optionally adjusts volume, returns handle

+

proAudio.sampleFromMemory ( data, sampleRate )
converts an array of numeric data into a sound sample having the defined sample rate, returns handle

+

proAudio.sampleDestroy ( sample )
deletes a previously created sound sample resource identified by its handle

+

duration, channels, sampleRate, bitsPerSample = proAudio.sampleProperties ( sample )
returns properties of a sample identified by its handle

+

proAudio.soundActive ( )
returns number of currently active sounds

+

proAudio.soundLoop ( sample, volumeL = 1.0, volumeR = 1.0, disparity = 0.0, pitch = 1.0 )
plays a specified sound sample continuously and sets its parameters

+Parameters:

    +
  • sample handle of a previously loaded sample
  • volumeL (optional) left volume
  • volumeR (optional) right volume
  • disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
  • pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+

+Returns: a handle to the currently played sound or -1 in case of error

+

proAudio.soundPlay ( sample, volumeL = 1.0, volumeR = 1.0, disparity = 0.0, pitch = 1.0 )
plays a specified sound sample once and sets its parameters

+Parameters:

    +
  • sample handle of a previously loaded sample
  • volumeL (optional) left volume
  • volumeR (optional) right volume
  • disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
  • pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+

+Returns: a handle to the currently played sound or -1 in case of error

+

proAudio.soundStop 	(  [ sound ] )
stops a specified sound immediately, if a sound handle is passed, or stops all sounds

+

proAudio.soundUpdate ( sound, volumeL, volumeR, disparity = 0.0, pitch = 1.0 )
updates parameters of a specified sound

+Parameters:

    +
  • sound handle of a currently active sound
  • volumeL left volume
  • volumeR right volume
  • disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right.
  • pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample.
+

+Returns: true in case the parameters have been updated successfully

+Example 1

+
+-- create an audio device using default parameters or exit in case of errors
+require("proAudioRt")
+if not proAudio.create() then os.exit(1) end

+

-- load and play a sample:
+sample = proAudio.sampleFromFile("sample.ogg")
+if sample then proAudio.soundPlay(sample) end

+

-- wait until the sound has finished:
+while proAudio.soundActive()>0 do
+  proAudio.sleep(0.05)
+end

+

-- close audio device
+proAudio.destroy()
+

+Dynamic creation of audio samples

+There is no equivalent to the C++ AudioSample class in the proteaAudio Lua API. However, mono audio samples may be dynamically created by the following call:

+

proAudio.sampleFromMemory(data, sampleRate)

+The data parameter has to be a table reference containing an array of numeric PCM data ranging from -1.0 to +1.0. The sampleRate parameter defines the number of samples per second. Typical sample rates are 22050 or 44100. Note that for obtaining good qualities when doing dynamic pitch shifts high sample rates (up to 88200) are recommended.

+Example 2

+
+-- function creating a sine wave sample:
+function sampleSine(freq, duration, sampleRate)
+	local data = { }
+	for i = 1,duration*sampleRate do
+		data[i] = math.sin( (i*freq/sampleRate)*math.pi*2)
+	end
+	return proAudio.sampleFromMemory(data, sampleRate)
+end

+

-- plays a sample shifted by a number of halftones for a definable period of time
+function playNote(sample, pitch, duration, volumeL, volumeR, disparity)
+	local scale = 2^(pitch/12)
+	local sound = proAudio.soundLoop(sample, volumeL, volumeR, disparity, scale)
+	proAudio.sleep(duration)
+	proAudio.soundStop(sound)
+end

+

-- create an audio device using default parameters and exit in case of errors
+require("proAudioRt")
+if not proAudio.create() then os.exit(1) end

+

-- generate a sample:
+local sample = sampleSine(440, 0.5, 88200)

+

-- play scale (a major):
+local duration = 0.5
+for i,note in ipairs({ 0, 2, 4, 5, 7, 9, 11, 12 }) do
+	playNote(sample, note, duration)
+end

+

-- cleanup
+proAudio.destroy()
+

+Limitations

+ +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/example.cpp b/example.cpp new file mode 100644 index 0000000..b0a4157 --- /dev/null +++ b/example.cpp @@ -0,0 +1,18 @@ +#include "proAudioRt.h" + +int main( int argc, char **argv ) { + // create an audio device using the RtAudio backend and default parameters: + DeviceAudio* audio=DeviceAudioRt::create(); + if(!audio) return 1; // exit in case of errors + + // load and play a sample: + unsigned int sample = audio->sampleFromFile("sample.ogg"); + if(sample) audio->soundPlay(sample); + + // wait until the sound has finished: + while(audio->soundActive()); // for the sake of simplicity busy waiting instead of a preferable sleep() call + + // cleanup and exit: + audio->destroy(); + return 0; +} diff --git a/example.lua b/example.lua new file mode 100644 index 0000000..6f68406 --- /dev/null +++ b/example.lua @@ -0,0 +1,15 @@ +-- create an audio device using default parameters and exit in case of errors +require("proAudioRt") +if not proAudio.create() then os.exit(1) end + +-- load and play a sample: +sample = proAudio.sampleFromFile("sample.ogg") +if sample then proAudio.soundPlay(sample) end + +-- wait until the sound has finished: +while proAudio.soundActive()>0 do + proAudio.sleep(0.05) +end + +-- close audio device +proAudio.destroy() diff --git a/footer.html b/footer.html new file mode 100644 index 0000000..ca3b19e --- /dev/null +++ b/footer.html @@ -0,0 +1,13 @@ +
+ + + + +
+ © 2009-02-04 by Gerald Franz, www.viremo.de + + impressum +
+
+ + diff --git a/header.html b/header.html new file mode 100644 index 0000000..76b0bc3 --- /dev/null +++ b/header.html @@ -0,0 +1,6 @@ + + +proteaAudio + + +
diff --git a/lua.txt b/lua.txt new file mode 100644 index 0000000..a59062d --- /dev/null +++ b/lua.txt @@ -0,0 +1,176 @@ +/** \page proteaAudioLua proteaAudio Lua API + + + +\section intro Overview +Lua is a lightweight yet powerful dynamic language. proteaAudio contains almost complete Lua bindings that widely correspond to the C++ API. By default, the proteaAudio Lua bindings are compiled to a dynamic library based on the RtAudio backend which is imported via the following statement: + +
require("proAudioRt")
+ +All API calls are collected in the global table proAudio. Therefore, proteaAudio is initialized (using the default parameters) by the following call: + +
proAudio.create()
+ +After that the individual methods and functions may be called analogous to the C++ interface of class DeviceAudio. + + + +\section functions Functions + +
proAudio.create( tracks = 8, frequency = 22050, chunkSize = 1024 )
+initializes audio playback device + +Parameters: +- tracks (optional) number of sounds that can be played in parallel +- frequency (optional) playback frequency +- chunkSize (optional) size of the internal buffer in bytes. Note that a small chunkSize results in low playback latency, but may cause computational overhead and hick-ups under higher system load + +Returns: + true in case the device initialization was successful + +
proAudio.destroy( )
+closes audio device and terminates playback + +Returns: + true in case the device was successfully closed + +
proAudio.loaderAvailable ( suffix )
+returns true in case a loader for this file type is available + +
proAudio.volume 	(  left, [ right ] )
+sets master volume, either for both channels uniformly, or individually + +
proAudio.sleep( seconds )
+Suspends the execution of the current thread for a definable number of seconds. Note that audio mixing and playback runs in its own background thread and is therefore not affected by this auxiliary call. + +
proAudio.sampleFromFile (  filename, volume = 1.0 )
+loads a sound sample from file, optionally adjusts volume, returns handle + +
proAudio.sampleFromMemory ( data, sampleRate )
+converts an array of numeric data into a sound sample having the defined sample rate, returns handle + +
proAudio.sampleDestroy ( sample )
+deletes a previously created sound sample resource identified by its handle + +
duration, channels, sampleRate, bitsPerSample = proAudio.sampleProperties ( sample )
+returns properties of a sample identified by its handle + +
proAudio.soundActive ( )
+returns number of currently active sounds + +
proAudio.soundLoop ( sample, volumeL = 1.0, volumeR = 1.0, disparity = 0.0, pitch = 1.0 )
+plays a specified sound sample continuously and sets its parameters + +Parameters: +- sample handle of a previously loaded sample +- volumeL (optional) left volume +- volumeR (optional) right volume +- disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. +- pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + +Returns: + a handle to the currently played sound or -1 in case of error + +
proAudio.soundPlay ( sample, volumeL = 1.0, volumeR = 1.0, disparity = 0.0, pitch = 1.0 )
+plays a specified sound sample once and sets its parameters + +Parameters: +- sample handle of a previously loaded sample +- volumeL (optional) left volume +- volumeR (optional) right volume +- disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. +- pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + +Returns: + a handle to the currently played sound or -1 in case of error + +
proAudio.soundStop 	(  [ sound ] )
+stops a specified sound immediately, if a sound handle is passed, or stops all sounds + +
proAudio.soundUpdate ( sound, volumeL, volumeR, disparity = 0.0, pitch = 1.0 )
+updates parameters of a specified sound + +Parameters: +- sound handle of a currently active sound +- volumeL left volume +- volumeR right volume +- disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. +- pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + +Returns: + true in case the parameters have been updated successfully + + + +\section example1 Example 1 + +
+-- create an audio device using default parameters or exit in case of errors
+require("proAudioRt")
+if not proAudio.create() then os.exit(1) end
+
+-- load and play a sample:
+sample = proAudio.sampleFromFile("sample.ogg")
+if sample then proAudio.soundPlay(sample) end
+
+-- wait until the sound has finished:
+while proAudio.soundActive()>0 do
+  proAudio.sleep(0.05)
+end
+
+-- close audio device
+proAudio.destroy()
+
+ +\section dynSamples Dynamic creation of audio samples + +There is no equivalent to the C++ AudioSample class in the proteaAudio Lua API. However, mono audio samples may be dynamically created by the following call: + +
proAudio.sampleFromMemory(data, sampleRate)
+ +The data parameter has to be a table reference containing an array of numeric PCM data ranging from -1.0 to +1.0. The sampleRate parameter defines the number of samples per second. Typical sample rates are 22050 or 44100. Note that for obtaining good qualities when doing dynamic pitch shifts high sample rates (up to 88200) are recommended. + +\section example2 Example 2 + +
+-- function creating a sine wave sample:
+function sampleSine(freq, duration, sampleRate)
+	local data = { }
+	for i = 1,duration*sampleRate do
+		data[i] = math.sin( (i*freq/sampleRate)*math.pi*2)
+	end
+	return proAudio.sampleFromMemory(data, sampleRate)
+end
+
+-- plays a sample shifted by a number of halftones for a definable period of time
+function playNote(sample, pitch, duration, volumeL, volumeR, disparity)
+	local scale = 2^(pitch/12)
+	local sound = proAudio.soundLoop(sample, volumeL, volumeR, disparity, scale)
+	proAudio.sleep(duration)
+	proAudio.soundStop(sound)
+end
+
+
+-- create an audio device using default parameters and exit in case of errors
+require("proAudioRt")
+if not proAudio.create() then os.exit(1) end
+
+-- generate a sample:
+local sample = sampleSine(440, 0.5, 88200)
+
+-- play scale (a major):
+local duration = 0.5
+for i,note in ipairs({ 0, 2, 4, 5, 7, 9, 11, 12 }) do
+	playNote(sample, note, duration)
+end
+
+-- cleanup
+proAudio.destroy()
+
+ +\section limitations Limitations + +- The proteaAudio Lua API currently has no equivalent to the C++ API's DeviceAudio::loaderRegister() method. +- no bindings to the SDL backend yet + +*/ \ No newline at end of file diff --git a/playAudioRt.cpp b/playAudioRt.cpp new file mode 100644 index 0000000..d7e2fff --- /dev/null +++ b/playAudioRt.cpp @@ -0,0 +1,38 @@ +#include "proAudioRt.h" +#include +#include + +using namespace std; + +// Platform-dependent sleep routines. +#if defined( __WINDOWS_ASIO__ ) || defined( __WINDOWS_DS__ ) + #include + #define SLEEP( milliseconds ) Sleep( (DWORD) milliseconds ) +#else // Unix variants + #include + #define SLEEP( milliseconds ) usleep( (unsigned long) (milliseconds * 1000.0) ) +#endif + + +int main( int argc, char **argv ) { + if(argc<2) { + fprintf(stderr, "usage: %s audiofile [volumeFactor] [pitchFactor]\n", argv[0]); + return 1; + } + DeviceAudio* audio=DeviceAudioRt::create(); + if(!audio) { + fprintf(stderr, "ERROR opening audio device. Abort.\n"); + return 1; + } + + float volume = (argc>2) ? atof(argv[2]) : 1.0f; + float pitch = (argc>3) ? atof(argv[3]) : 1.0f; + unsigned int sample1=audio->sampleFromFile(argv[1], volume); + if(sample1) audio->soundPlay(sample1, 1.0f,1.0f,0.0f, pitch); + + // main loop: + while(audio->soundActive()) + SLEEP(100); + + return 0; +} diff --git a/playAudioSdl.cpp b/playAudioSdl.cpp new file mode 100644 index 0000000..843026f --- /dev/null +++ b/playAudioSdl.cpp @@ -0,0 +1,32 @@ +#include +#include +#include + +#include +#include "proAudioSdl.h" + +using namespace std; + +//--- main --------------------------------------------------------- + +int main(int argc, char **argv) { + if(argc<2) { + fprintf(stderr, "usage: %s audiofile [volumeFactor] [pitchFactor]\n", argv[0]); + return 1; + } + DeviceAudio* audio=DeviceAudioSdl::create(); + if(!audio) { + fprintf(stderr, "ERROR opening audio device. Abort.\n"); + return 1; + } + + float volume = (argc>2) ? atof(argv[2]) : 1.0f; + float pitch = (argc>3) ? atof(argv[3]) : 1.0f; + unsigned int sample1=audio->sampleFromFile(argv[1], volume); + if(sample1) audio->soundPlay(sample1, 1.0f,1.0f,0.0f, pitch); + + // main loop: + while(audio->soundActive()) + SDL_Delay(10); + return 0; +} diff --git a/proAudio.cpp b/proAudio.cpp new file mode 100644 index 0000000..a4cf9ee --- /dev/null +++ b/proAudio.cpp @@ -0,0 +1,167 @@ +#include "proAudio.h" +#include +#include +#include +#include + +using namespace std; + +//--- class AudioSample -------------------------------------------- +AudioSample::AudioSample(const AudioSample & source) : + m_size(source.m_size), m_channels(source.m_channels), m_sampleRate(source.m_sampleRate), m_bitsPerSample(source.m_bitsPerSample) { + m_data = new unsigned char [m_size]; memcpy(m_data,source.m_data, m_size); +} + +bool AudioSample::bitsPerSample(unsigned short bits) { + if(bits==16) { + if(m_bitsPerSample==8) { + unsigned char* data = new unsigned char[2*m_size]; + for(unsigned int i=0; iCHAR_MAX) *ptr =CHAR_MAX; + else if(valueSHRT_MAX) *ptr =SHRT_MAX; + else if(value1.0f) *ptr=1.0f; + else if(*ptr<-1.0f) *ptr=-1.0f; + } + else fprintf(stderr,"AudioSample::changeVolume ERROR: %i bits per sample not supported.\n",m_bitsPerSample); +} + +AudioSample* AudioSample::readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *)) { + char id[4]; //four unsigned chars to hold chunk IDs + readFunc(id,sizeof(unsigned char),4,stream); + if (strncmp(id,"RIFF",4)!=0) return 0; + + unsigned int size; + readFunc(&size,sizeof(unsigned int),1,stream); + + readFunc(id,sizeof(unsigned char),4,stream); + if (strncmp(id,"WAVE",4)!=0) return 0; + + unsigned short encoding, block_align, channels, bitsPerSample; + unsigned int chunk_length, byte_rate, sampleRate; + + readFunc(id, sizeof(unsigned char), 4, stream); //read ID 'fmt '; + readFunc(&chunk_length, sizeof(unsigned int),1,stream); // header length, 16 expected + readFunc(&encoding, sizeof(short), 1, stream); // should be "1" for simple PCM data + if(encoding!=1) return 0; + + readFunc(&channels, sizeof(short),1,stream); + readFunc(&sampleRate, sizeof(unsigned int), 1, stream); + readFunc(&byte_rate, sizeof(unsigned int), 1, stream); + readFunc(&block_align, sizeof(short), 1, stream); + readFunc(&bitsPerSample, sizeof(short), 1, stream); + + readFunc(id, sizeof(unsigned char), 4, stream); // read ID 'data' + readFunc(&size, sizeof(unsigned int), 1, stream); + unsigned char *data = new unsigned char[size]; + readFunc(data, sizeof(unsigned char), size, stream); + + return new AudioSample(data,size, channels, sampleRate, bitsPerSample); +} + +AudioSample* AudioSample::loadWav(const std::string & fname) { +#ifdef _MSC_VER + FILE *fp = 0; + fopen_s(&fp, fname.c_str(), "rb"); +#else + FILE *fp = fopen(fname.c_str(), "rb"); +#endif + if (!fp) return 0; + AudioSample * pSample = readWav(fp, fread); + fclose(fp); + return pSample; +} + +//--- class DeviceAudio -------------------------------------------- + +DeviceAudio* DeviceAudio::s_instance=0; + +extern "C" { +extern int stb_vorbis_decode_filename(char *filename, int *channels, int* sample_rate, short **output); +}; + +static AudioSample* loadOgg(const std::string & fname) { + int channels, sampleRate; + short *decoded; + int len = stb_vorbis_decode_filename(const_cast(fname.c_str()), &channels, &sampleRate, &decoded); + if(len<0) return 0; + // convert to AudioSample: + unsigned int size = len*channels*sizeof(short); + unsigned char * data = new unsigned char[size]; + if(!data) return 0; + memcpy(data,decoded, size); + free(decoded); + return new AudioSample(data, size, channels, sampleRate, 16); +} + +static string toLower(const string & s) { + string retStr(s); + for(size_t i=0; i(tolower(retStr[i])); + return retStr; +} + +DeviceAudio::DeviceAudio() : m_freqOut(0), m_volL(1.0f), m_volR(1.0f) { + loaderRegister(AudioSample::loadWav,"wav"); + loaderRegister(loadOgg,"ogg"); +} + +unsigned int DeviceAudio::sampleFromFile(const std::string & filename, float volume) { + if(filename.rfind('.')>filename.size()) return 0; + string suffix=toLower(filename.substr(filename.rfind('.')+1)); + map::iterator it = mm_loader.find(suffix); + if(it==mm_loader.end()) return 0; + AudioSample* pSample = (*(it->second))(filename); + if(!pSample) return 0; + unsigned int ret = sampleFromMemory(*pSample, volume); + delete pSample; + return ret; +} + +bool DeviceAudio::loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix) { + return mm_loader.insert(std::make_pair(toLower(suffix),loadFunc)).second; +} + +bool DeviceAudio::loaderAvailable(const std::string & suffix) const { + return mm_loader.find(toLower(suffix))!=mm_loader.end(); +} diff --git a/proAudio.h b/proAudio.h new file mode 100644 index 0000000..5ef18a7 --- /dev/null +++ b/proAudio.h @@ -0,0 +1,169 @@ +#ifndef _PRO_AUDIO +#define _PRO_AUDIO + +#include +#include + +/** @file proAudio.h + \brief Public interface of proteaAudio + + Contains the declaration of the audio sample class and the abstract base class for audio mixer/playback devices + + \author Gerald Franz, www.viremo.de + \version 0.6 + + License notice (zlib license): + + (c) 2009 by Gerald Franz, www.viremo.de + + This software is provided 'as-is', without any express or implied + warranty. In no event will the author be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +//--- class AudioSample -------------------------------------------- +/// class representing an audio sample +class AudioSample { +public: + /// constructor from memory data + AudioSample(unsigned char * data, unsigned int size, unsigned short channels, unsigned int sampleRate, unsigned short bitsPerSample) : + m_data(data), m_size(size), m_channels(channels), m_sampleRate(sampleRate), m_bitsPerSample(bitsPerSample) { } + /// copy constructor + AudioSample(const AudioSample & source); + /// destructor + ~AudioSample() { delete[] m_data; } + + /// allows accessing sample data + unsigned char * data() { return m_data; }; + /// allows reading sample data + const unsigned char * data() const { return m_data; }; + /// returns sample size in bytes + unsigned int size() const { return m_size; } + /// returns sample size in number of frames + unsigned int frames() const { return m_size/m_channels/(m_bitsPerSample>>3); } + /// returns size of a single frame in bytes + unsigned int sizeFrame() const { return m_channels*(m_bitsPerSample>>3); } + /// returns number of parallel channels, 1 mono, 2 stereo + unsigned short channels() const { return m_channels; } + /// returns number of frames per second, e.g., 44100, 22050 + unsigned int sampleRate() const { return m_sampleRate; } + /// returns number of bits per mono sample, e.g., 8, 16 + unsigned short bitsPerSample() const { return m_bitsPerSample; } + /// converts to a different bit rate, e.g., 8, 16 + bool bitsPerSample(unsigned short bits); + /// returns number of bytes per sample, e.g., 1, 2 + unsigned short bytesPerSample() const { return m_bitsPerSample>>3; } + + /// changes volume by given factor + void volume(float f); + + /// loads a WAV file + static AudioSample* loadWav(const std::string & fname); + /// reads WAV data from a stream via a function compatible to std::fread + static AudioSample* readWav(FILE* stream, size_t (*readFunc)( void *, size_t, size_t, FILE *)); +protected: + /// stores sample data + unsigned char * m_data; + /// sample size in bytes + unsigned int m_size; + /// number of parallel channels, 1 mono, 2 stereo + unsigned short m_channels; + /// number of samples per second, e.g., 44100, 22050 + unsigned int m_sampleRate; + /// number of bits per sample, e.g., 8, 16 + unsigned short m_bitsPerSample; +}; + +//--- class DeviceAudio -------------------------------------------- + +/// abstract base class for stereo audio mixer/playback devices +class DeviceAudio { +public: + /// returns singleton object + /** This call is only allowed after a successful precedent creation of an audio device */ + static DeviceAudio& singleton() { return *s_instance; } + /// calls the destructor of the singleton object + static void destroy() { if(s_instance) delete s_instance; s_instance=0; }; + + /// sets master volume + void volume(float left, float right) { m_volL=left; m_volR=right; } + /// sets master volume + void volume(float leftAndRight) { m_volL=m_volR=leftAndRight; } + /// registers an audio sample loader function handling a file type identified by suffix + /** The function has to be of type AudioSample * loadXYZ(const std::string & filename).*/ + bool loaderRegister(AudioSample *(*loadFunc)(const std::string &), const std::string & suffix); + /// returns true in case a loader for this file type is available + bool loaderAvailable(const std::string & suffix) const; + + /// loads a sound sample from file, optionally adjusts volume, returns handle + virtual unsigned int sampleFromFile(const std::string & filename, float volume=1.0f); + /// converts a sound sample to internal audio format, returns handle + virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f)=0; + /// deletes a previously created sound sample resource identified by its handle + virtual bool sampleDestroy(unsigned int sample)=0; + /// allows read access to a sample identified by its handle + virtual const AudioSample* sample(unsigned int handle) const { return 0; } + + /// plays a specified sound sample once and sets its parameters + /** \param sample handle of a previously loaded sample + \param volumeL (optional) left volume + \param volumeR (optional) right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return a handle to the currently played sound or -1 in case of error */ + virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0; + /// plays a specified sound sample continuously and sets its parameters + /** \param sample handle of a previously loaded sample + \param volumeL (optional) left volume + \param volumeR (optional) right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return a handle to the currently played sound or -1 in case of error */ + virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f )=0; + /// updates parameters of a specified sound + /** \param sound handle of a currently active sound + \param volumeL left volume + \param volumeR right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return true in case the parameters have been updated successfully */ + virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f )=0; + /// stops a specified sound immediately + virtual bool soundStop(unsigned int sound)=0; + /// stops all sounds immediately + virtual void soundStop()=0; + /// returns number of currently active sounds + virtual unsigned int soundActive() const=0; + +protected: + /// constructor + DeviceAudio(); + /// destructor + virtual ~DeviceAudio() { s_instance = 0; } + + /// stores output stream frequency + unsigned int m_freqOut; + /// stores left master volume + float m_volL; + /// stores right master volume + float m_volR; + /// map associating suffixes to loader functions + std::map mm_loader; + + /// pointer to singleton + static DeviceAudio * s_instance; +}; + +#endif // _PRO_AUDIO diff --git a/proAudioRt.cpp b/proAudioRt.cpp new file mode 100644 index 0000000..c575709 --- /dev/null +++ b/proAudioRt.cpp @@ -0,0 +1,242 @@ +#include "proAudioRt.h" +#include +#include +#include +#include +#include + +using namespace std; + +struct _AudioTrack { + /// sample + AudioSample * sample; + + /// position in sample in frames + unsigned int dpos; + /// length of sample in frames + unsigned int dlen; + /// disparity in seconds between left and right, normally 0.0f + float disparity; + /// left volume + float volL; + /// right volume + float volR; + /// pitch factor, normally 1.0f + float pitch; + /// stores whether sample has to be looped + bool isLoop; + /// stores whether sample is currently playing + bool isPlaying; +}; + +DeviceAudio* DeviceAudioRt::create(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) { + if(!s_instance) { + DeviceAudioRt* pAudio = new DeviceAudioRt(nTracks,frequency,chunkSize); + if(!pAudio->m_freqOut) delete pAudio; + else s_instance = pAudio; + } + return s_instance; +} + +DeviceAudioRt::DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) : DeviceAudio() { + if ( m_dac.getDeviceCount() < 1 ) { + fprintf(stderr,"DeviceAudioRt ERROR: No audio devices found!\n"); + return; + } + // Set our stream parameters for output only. + RtAudio::StreamParameters oParams; + oParams.deviceId = m_dac.getDefaultOutputDevice(); // default device + oParams.nChannels = 2; // stereo + oParams.firstChannel = 0; + + try { + m_dac.openStream( &oParams, NULL, RTAUDIO_SINT16, frequency, &chunkSize, &cbMix, (void *)this ); + m_dac.startStream(); + } + catch ( RtError& e ) { + fprintf(stderr,"%s\n", e.getMessage().c_str()); + if(m_dac.isStreamOpen()) m_dac.closeStream(); + return; + } + + // initialize tracks: + m_nSound=nTracks; + ma_sound=new _AudioTrack[m_nSound]; + memset(ma_sound,0,m_nSound*sizeof(_AudioTrack)); + m_freqOut = frequency; +} + +DeviceAudioRt::~DeviceAudioRt() { + if(m_dac.isStreamOpen()) m_dac.closeStream(); + delete [] ma_sound; + for( map::iterator it=mm_sample.begin(); it!=mm_sample.end(); ++it) + delete it->second; + mm_sample.clear(); +} + +unsigned int DeviceAudioRt::sampleFromMemory(const AudioSample & sample, float volume) { + AudioSample * pSample = new AudioSample(sample); + if(volume!=1.0f) pSample->volume(volume); + pSample->bitsPerSample(16); + mm_sample.insert(make_pair(++m_sampleCounter,pSample)); + return m_sampleCounter; +} + +bool DeviceAudioRt::sampleDestroy(unsigned int sample) { + // look for sample: + map::iterator iter=mm_sample.find(sample); + if( iter == mm_sample.end() ) return false; + // stop currently playing sounds referring to this sample: + for (unsigned int i=0; isecond) + ma_sound[i].isPlaying=false; + // cleanup: + delete iter->second; + if(iter->first==m_sampleCounter) --m_sampleCounter; + mm_sample.erase(iter); + return true; +} + +const AudioSample* DeviceAudioRt::sample(unsigned int handle) const { + map::const_iterator it=mm_sample.find(handle); + if( it == mm_sample.end() ) return 0; + return it->second; +} + + +unsigned int DeviceAudioRt::soundPlay(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch ) { + // look for sample: + map::iterator iter=mm_sample.find(sample); + if( iter == mm_sample.end() ) return 0; // no sample found + // look for an empty (or finished) sound track + unsigned int i; + for ( i=0; isecond->sampleRate(); + if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut; + + // put the sample data in the slot and play it + ma_sound[i].sample = iter->second; + ma_sound[i].dlen = iter->second->frames(); + ma_sound[i].dpos = 0; + ma_sound[i].volL=volumeL; + ma_sound[i].volR=volumeR; + ma_sound[i].disparity=disparity; + ma_sound[i].pitch=fabs(pitch); + ma_sound[i].isLoop=false; + ma_sound[i].isPlaying=true; + return i+1; +} + +unsigned int DeviceAudioRt::soundLoop(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch ) { + unsigned int ret=soundPlay(sample,volumeL,volumeR,disparity, pitch); + if(ret) ma_sound[ret-1].isLoop=true; + return ret; +} + +bool DeviceAudioRt::soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity, float pitch ) { + if(!sound || (sound>m_nSound) || !ma_sound[sound-1].isPlaying) return false; + ma_sound[--sound].volL=volumeL; + ma_sound[sound].volR=volumeR; + ma_sound[sound].disparity=disparity; + unsigned int sampleRate = ma_sound[sound].sample->sampleRate(); + if(sampleRate!=m_freqOut) pitch*=(float)sampleRate/(float)m_freqOut; + ma_sound[sound].pitch=fabs(pitch); + return true; +} + +bool DeviceAudioRt::soundStop(unsigned int sound) { + if(!sound||(sound>m_nSound)||!ma_sound[sound-1].isPlaying) return false; + ma_sound[sound-1].isPlaying=false; + return true; +} + +void DeviceAudioRt::soundStop() { + for (unsigned int i=0; i(&m_dac)->isStreamRunning() ) return 0; + unsigned int ret = 0, i; + for ( i=0; ichannels(); + if((ma_sound[i].pitch==1.0f)&&!ma_sound[i].disparity) { // use optimized default mixing: + unsigned int currPos=ma_sound[i].dpos+j; + if(ma_sound[i].isLoop) currPos%=ma_sound[i].dlen; + else if(currPos >= ma_sound[i].dlen) continue; + currPos*=ma_sound[i].sample->sizeFrame(); + float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos]))); + left += dataL * m_volL*ma_sound[i].volL; + float dataR = (nChannels>1) ? (float)(*((signed short *)(&ma_sound[i].sample->data()[currPos+2]))) : dataL; + right+= dataR * m_volR*ma_sound[i].volR; + } + else { // use nearest sample and disparity: + double fract=ma_sound[i].dpos+j*ma_sound[i].pitch; + unsigned int currPos=(unsigned int)fract; + fract = fmod(fract,1.0); + int currPosL= (ma_sound[i].disparity<0.0f) ? currPos+int(m_freqOut*ma_sound[i].disparity) : currPos; + int currPosR= (ma_sound[i].disparity>0.0f) ? currPos-int(m_freqOut*ma_sound[i].disparity) : currPos; + if(nChannels>1) currPosR+=sizeof(signed short); // use second channel + if(ma_sound[i].isLoop) { + currPosL+=ma_sound[i].dlen; + currPosL%=ma_sound[i].dlen; + currPosR+=ma_sound[i].dlen; + currPosR%=ma_sound[i].dlen; + } + if(currPosL<0) { + // do nothing + } + else if((unsigned int)currPosL+1 < ma_sound[i].dlen) { + currPosL*=ma_sound[i].sample->sizeFrame(); + float dataL = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL]))) + + (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL+ma_sound[i].sample->sizeFrame()]))); + left += dataL * m_volL*ma_sound[i].volL; + } + else if((unsigned int)currPosL+1 == ma_sound[i].dlen) { + currPosL*=ma_sound[i].sample->sizeFrame(); + float dataL = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosL]))); + left += dataL * m_volL*ma_sound[i].volL; + } + + if(currPosR<0) { + // do nothing + } + else if((unsigned int)currPosR+1 < ma_sound[i].dlen) { + currPosR*=ma_sound[i].sample->sizeFrame(); + float dataR = (1.0f-(float)fract)*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR]))) + + (float)fract*(float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR+ma_sound[i].sample->sizeFrame()]))); + right += dataR * m_volR*ma_sound[i].volR; + } + else if((unsigned int)currPosR+1 == ma_sound[i].dlen) { + currPosR*=ma_sound[i].sample->sizeFrame(); + float dataR = (float)(*((signed short *)(&ma_sound[i].sample->data()[currPosR]))); + right += dataR * m_volR*ma_sound[i].volR; + } + } + } + // clamp and set output: + outputBuffer[2*j] = left>SHRT_MAX ? SHRT_MAX : leftSHRT_MAX ? SHRT_MAX : rightma_sound[i].dlen+2*abs(int(m_freqOut*-ma_sound[i].disparity))) + ma_sound[i].isPlaying=false; + } + return 0; +} diff --git a/proAudioRt.h b/proAudioRt.h new file mode 100644 index 0000000..7a28c57 --- /dev/null +++ b/proAudioRt.h @@ -0,0 +1,88 @@ +#include "proAudio.h" +#include +#include + +/** @file proAudioRt.h + \brief RtAudio backend of proteaAudio + \author Gerald Franz, www.viremo.de + \version 0.6 +*/ + +struct _AudioTrack; + +/// an rtAudio based stereo audio mixer/playback device +/** DeviceAudioRt offers some advanced features such as dynamic pitch, + independent volume control for both channels, and user-defined time shifts between the channels. */ +class DeviceAudioRt : public DeviceAudio { +public: + ///creates audio device + /** Use this method instead of a constructor. + \param nTracks (optional) the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor. + \param frequency (optional) sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor. + \param chunkSize (optional) the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency. + \return a pointer to an audio device object in case of success + Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called. + */ + static DeviceAudio* create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024); + + /// converts a sound sample to internal audio format, returns handle + virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f); + /// deletes a previously created sound sample resource identified by its handle + virtual bool sampleDestroy(unsigned int sample); + /// allows read access to a sample identified by its handle + virtual const AudioSample* sample(unsigned int handle) const; + + /// plays a specified sample once and sets its parameters + /** \param sample a sample handle returned by a previous load() call + \param volumeL (optional) left volume + \param volumeR (optional) right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return a handle to the currently played sound or 0 in case of error */ + virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f ); + /** plays a specified sample continuously and sets its parameters + \param sample a sample handle returned by a previous load() call + \param volumeL (optional) left volume + \param volumeR (optional) right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return a handle to the currently played sound or 0 in case of error */ + virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f ); + /// updates parameters of a specified sound + /** \param sound handle of a currently active sound + \param volumeL left volume + \param volumeR right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return true in case the parameters have been updated successfully */ + virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f ); + /// stops a specified sound immediately + virtual bool soundStop(unsigned int sound); + /// stops all sounds immediately + virtual void soundStop(); + /// returns number of currently active sounds + virtual unsigned soundActive() const; +protected: + /// constructor. Use the create() method instead + DeviceAudioRt(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize); + /// destructor. Use the destroy() method instead + virtual ~DeviceAudioRt(); + /// mixes tracks to a single output stream + int mixOutputFloat(signed short *outputBuffer, unsigned int nFrames); + + /// stores loaded sound samples + std::map mm_sample; + /// stores maximum sample id + unsigned int m_sampleCounter; + + /// stores sounds to be mixed + _AudioTrack * ma_sound; + /// stores number of parallel sounds + unsigned int m_nSound; + /// audio manager + RtAudio m_dac; + + /// mixer callback + static int cbMix(void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *data) { + return static_cast(data)->mixOutputFloat((signed short*)outputBuffer, nFrames); } +}; diff --git a/proAudioRt_lua.cpp b/proAudioRt_lua.cpp new file mode 100644 index 0000000..95ee38c --- /dev/null +++ b/proAudioRt_lua.cpp @@ -0,0 +1,219 @@ +#if defined __WIN32__ || defined WIN32 +# define WINDOWS_LEAN_AND_MEAN +# include +# define _EXPORT __declspec(dllexport) +#else +# include +# define _EXPORT +#endif + +extern "C" { +#include +#include +#include +}; + +#include "proAudioRt.h" +#include +#include + +using namespace std; + +static int create(lua_State *L) { + unsigned int nTracks = lua_isnumber(L,1) ? lua_tointeger(L,1) : 8; + unsigned int frequency = lua_isnumber(L,2) ? lua_tointeger(L,2) : 22050; + unsigned int chunkSize = lua_isnumber(L,3) ? lua_tointeger(L,3) : 1024; + DeviceAudio* pAudio = DeviceAudioRt::create(nTracks, frequency, chunkSize); + lua_pushboolean(L, pAudio ? 1 : 0); + return 1; +} + +static int destroy(lua_State *L) { + DeviceAudio::destroy(); + return 0; +} + +static int loaderAvailable(lua_State *L) { + const char * fname = luaL_checkstring(L,1); + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + lua_pushboolean(L, audio.loaderAvailable(fname)); + return 1; +} + +static int volume(lua_State *L) { + int argc = lua_gettop(L); + if(!argc) return 0; + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + if(argc==1) audio.volume(luaL_checknumber(L,1)); + else audio.volume(luaL_checknumber(L,1),luaL_checknumber(L,2)); + return 0; +} + +static int sleep(lua_State*L) { + double secs = luaL_checknumber(L,1); +#if defined __WIN32__ || defined WIN32 + Sleep( (DWORD)(secs*1000.0) ) ; +#else // Unix variants + usleep( (unsigned long) (secs * 1000000.0) ); +#endif + return 0; +} + +static int sampleFromFile(lua_State *L) { + const char * fname = luaL_checkstring(L,1); + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + int argc = lua_gettop(L); + float volume = argc>1 ? luaL_checknumber(L,2) : 1.0f; + int ret = (int)audio.sampleFromFile(fname, volume); + if(ret) { + lua_pushinteger(L, ret); + return 1; + } + return 0; +} + +static int sampleFromMemory(lua_State *L) { + int argc = lua_gettop(L); + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio||argc<2||!lua_istable(L,1)) return 0; + + size_t len = lua_objlen(L,1); + if(!len) return 0; + int sampleRate = luaL_checkinteger(L,2); + signed short *data = new signed short[len]; + for(size_t i = 0; i=1.0f) data[i] = SHRT_MAX; + else data[i] = (signed short)(value*SHRT_MAX); + } + lua_pop(L,1); + } + AudioSample sample((unsigned char*)data, len*sizeof(signed short), 1, sampleRate, 16); + int ret = (int)audio.sampleFromMemory(sample); + if(ret) { + lua_pushinteger(L, ret); + return 1; + } + return 0; +} + +static int sampleDestroy(lua_State *L) { + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + lua_pushboolean(L, audio.sampleDestroy(luaL_checkinteger(L,1))); + return 1; +} + +static int sampleProperties(lua_State *L) { + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + const AudioSample *pSample = audio.sample(luaL_checkinteger(L,1)); + if(!pSample) return 0; + lua_pushnumber(L, (double)pSample->frames() / (double)pSample->sampleRate()); + lua_pushinteger(L, pSample->channels()); + lua_pushinteger(L, pSample->sampleRate()); + lua_pushinteger(L, pSample->bitsPerSample()); + return 4; +} + +static int soundPlay(lua_State *L) { + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + unsigned int sample = luaL_checkinteger(L,1); + float volL= lua_isnumber(L,2) ? lua_tonumber(L,2) : 1.0f; + float volR= lua_isnumber(L,3) ? lua_tonumber(L,3) : 1.0f; + float disparity= lua_isnumber(L,4) ? lua_tonumber(L,4) : 0.0f; + float pitch = lua_isnumber(L,5) ? lua_tonumber(L,5) : 1.0f; + int ret = (int)audio.soundPlay(sample, volL,volR,disparity,pitch); + if(ret) { + lua_pushinteger(L, ret); + return 1; + } + return 0; +} + +static int soundLoop(lua_State *L) { + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + unsigned int sample = luaL_checkinteger(L,1); + float volL= lua_isnumber(L,2) ? lua_tonumber(L,2) : 1.0f; + float volR= lua_isnumber(L,3) ? lua_tonumber(L,3) : 1.0f; + float disparity= lua_isnumber(L,4) ? lua_tonumber(L,4) : 0.0f; + float pitch = lua_isnumber(L,5) ? lua_tonumber(L,5) : 1.0f; + int ret = (int)audio.soundLoop(sample, volL,volR,disparity,pitch); + if(ret) { + lua_pushinteger(L, ret); + return 1; + } + return 0; +} + +static int soundUpdate(lua_State *L) { + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + unsigned int sound = luaL_checkinteger(L,1); + float volL= lua_isnumber(L,2) ? lua_tonumber(L,2) : 1.0f; + float volR= lua_isnumber(L,3) ? lua_tonumber(L,3) : 1.0f; + float disparity= lua_isnumber(L,4) ? lua_tonumber(L,4) : 0.0f; + float pitch = lua_isnumber(L,5) ? lua_tonumber(L,5) : 1.0f; + int ret = (int)audio.soundUpdate(sound, volL,volR,disparity,pitch); + if(ret) { + lua_pushinteger(L, ret); + return 1; + } + return 0; +} + +static int soundStop(lua_State *L) { + int argc = lua_gettop(L); + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + if(!argc) { + audio.soundStop(); + return 0; + } + unsigned int sound = luaL_checkinteger(L,1); + int ret = (int)audio.soundStop(sound); + if(ret) { + lua_pushinteger(L, ret); + return 1; + } + return 0; +} + +static int soundActive(lua_State *L) { + DeviceAudio & audio = DeviceAudio::singleton(); + if(!&audio) return 0; + lua_pushinteger(L, (int)audio.soundActive()); + return 1; +} + +extern "C" int _EXPORT luaopen_proAudioRt (lua_State* L) { + static const struct luaL_Reg funcs[] = { + {"create", create }, + {"destroy", destroy }, + {"loaderAvailable", loaderAvailable }, + {"volume", volume }, + {"sleep", sleep }, + + {"sampleFromFile", sampleFromFile }, + {"sampleFromMemory", sampleFromMemory }, + {"sampleDestroy", sampleDestroy }, + {"sampleProperties", sampleProperties }, + + {"soundPlay", soundPlay }, + {"soundLoop", soundLoop }, + {"soundUpdate", soundUpdate }, + {"soundStop", soundStop }, + {"soundActive", soundActive }, + {NULL, NULL} + }; + luaL_register(L, "proAudio", funcs); + return 1; +} diff --git a/proAudioSdl.cpp b/proAudioSdl.cpp new file mode 100644 index 0000000..804de7a --- /dev/null +++ b/proAudioSdl.cpp @@ -0,0 +1,289 @@ +#include "proAudioSdl.h" +extern "C" { +#include +}; + +#include +#include +#include + +using namespace std; + +//--- class DeviceAudioSdl ------------------------------------------ + +/// internal class to store sample data +class _AudioTrack { +public: + /// default constructor + _AudioTrack(Uint8 * pData=0, unsigned int length=0) : data(pData), dlen(length) { + dpos=0; disparity=0.0f; volL=1.0f; volR=1.0f; pitch=1.0f; isLoop=false; isPlaying=false; }; + /// pointer to raw sample data + Uint8 *data; + /// position in playback + Uint32 dpos; + /// length of sample in bytes + Uint32 dlen; + /// disparity in seconds between left and right, normally 0.0f + float disparity; + /// left volume + float volL; + /// right volume + float volR; + /// pitch factor, normally 1.0f + float pitch; + /// stores whether sample has to be looped + bool isLoop; + /// stores whether sample is currently playing + bool isPlaying; +}; + +DeviceAudio* DeviceAudioSdl::create(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) { + if(!s_instance) { + DeviceAudioSdl* pAudio = new DeviceAudioSdl(nTracks,frequency,chunkSize); + if(!pAudio->m_freqOut) delete pAudio; + else s_instance = pAudio; + } + return s_instance; +} + +DeviceAudioSdl::DeviceAudioSdl(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize) : DeviceAudio(), m_sampleCounter(0) { + // initialize SDL sound system: + if(!SDL_WasInit(SDL_INIT_AUDIO)&&(SDL_InitSubSystem(SDL_INIT_AUDIO)<0)) { + fprintf(stderr, "DeviceAudioSdl ERROR: cannot initialize SDL audio subsystem.\n"); + return; + } + + // set the audio format: + SDL_AudioSpec desired; + desired.freq = frequency; + desired.format = AUDIO_S16; + desired.channels = 2; // 1 = mono, 2 = stereo + desired.samples = chunkSize; // good low-latency value for callback + desired.callback = cbOutput; + desired.userdata = NULL; + // open the audio device + if ( SDL_OpenAudio(&desired, &m_spec) < 0 ) { + fprintf(stderr, "DeviceAudioSdl ERROR: Couldn't open audio.\n"); + return; + } + if((m_spec.format!=AUDIO_S16)||(m_spec.channels!=2)) { + fprintf(stderr, "DeviceAudioSdl WARNING: Could not get desired signed 16 bit stereo. Expect low quality sound.\n"); + m_isDesiredFormat=false; + } + else { + m_isDesiredFormat=true; + } + + // initialize tracks: + m_nSound=nTracks; + ma_sound=new _AudioTrack[m_nSound]; + + SDL_PauseAudio(0); // start sound + m_freqOut = frequency; +} + +DeviceAudioSdl::~DeviceAudioSdl() { + SDL_PauseAudio(1); + SDL_CloseAudio(); + delete [] ma_sound; +} + +void DeviceAudioSdl::cbOutput(void * userData, Uint8 *stream, int len) { + if(dynamic_cast(s_instance)->m_isDesiredFormat) + dynamic_cast(s_instance)->mixOutputFloat((signed short *)stream, len/2); + else dynamic_cast(s_instance)->mixOutputSInt(stream, len); +} + +void DeviceAudioSdl::mixOutputFloat(signed short *outputBuffer, unsigned int nFrames) { + for(unsigned int j=0; j= ma_sound[i].dlen) continue; + left +=float(*((Sint16 *)(&ma_sound[i].data[currPos]))) + *m_volL*ma_sound[i].volL; + right+=float(*((Sint16 *)(&ma_sound[i].data[currPos]))) + *m_volR*ma_sound[i].volR; + } + else { // use linear interpolation and disparity: + double fract=ma_sound[i].dpos+j*ma_sound[i].pitch; + int currPos=int(fract*0.5f)*2; + fract=(fract-currPos)*0.5f; + + int currPosL= (ma_sound[i].disparity<0.0f) + ? currPos-2*int(m_spec.freq*-ma_sound[i].disparity) + : currPos; + int currPosR= (ma_sound[i].disparity>0.0f) + ? currPos-2*int(m_spec.freq*ma_sound[i].disparity) + : currPos; + + if(ma_sound[i].isLoop) { + currPosL=(currPosL+10*ma_sound[i].dlen-2)%(ma_sound[i].dlen-2); + currPosR=(currPosR+10*ma_sound[i].dlen-2)%(ma_sound[i].dlen-2); + } + if((currPosL < int(ma_sound[i].dlen)-2)&&(currPosL>=0)) { + float currWavL=(1.0f-fract)*float(*((Sint16 *)(&ma_sound[i].data[currPosL]))) + +fract*float(*((Sint16 *)(&ma_sound[i].data[currPosL+2]))); + left +=currWavL*m_volL*ma_sound[i].volL; + } + if((currPosR < int(ma_sound[i].dlen)-2)&&(currPosR>=0)) { + float currWavR=(1.0f-fract)*float(*((Sint16 *)(&ma_sound[i].data[currPosR]))) + +fract*float(*((Sint16 *)(&ma_sound[i].data[currPosR+2]))); + right+=currWavR*m_volR*ma_sound[i].volR; + } + } + } + // clamp: + if(left>(float)SHRT_MAX) outputBuffer[j]=SHRT_MAX; + else if(left<(float)SHRT_MIN) outputBuffer[j]=SHRT_MIN; + else outputBuffer[j]=(Sint16)left; + if(right>(float)SHRT_MAX) outputBuffer[j+1]=SHRT_MAX; + else if(right<(float)SHRT_MIN) outputBuffer[j+1]=SHRT_MIN; + else outputBuffer[j+1]=(Sint16)right; + } + for (unsigned int i=0; ima_sound[i].dlen+2*abs(int(m_spec.freq*-ma_sound[i].disparity))) + ma_sound[i].isPlaying=false; + } +} + +void DeviceAudioSdl::mixOutputSInt(Uint8 *stream, int len) { + for (unsigned int i=0; i (unsigned int)len) amount = (unsigned int)len; + SDL_MixAudio(stream, &ma_sound[i].data[ma_sound[i].dpos], amount, SDL_MIX_MAXVOLUME); + ma_sound[i].dpos += amount; + } +} + +static void adjustVolume(signed short* data, size_t len, float volume) { + for(size_t i=0; i(float)SHRT_MAX) value=(float)SHRT_MAX; // clamp + else if(value<(float)SHRT_MIN) value=(float)SHRT_MIN; + shortValue=(signed short)value; + } +} + +unsigned int DeviceAudioSdl::sampleFromMemory(const AudioSample & sample, float volume) { + Uint8 destChannels= m_isDesiredFormat ? 1 : m_spec.channels; // convert to mono + Uint16 format = sample.bytesPerSample()==2 ? AUDIO_S16 : sample.bytesPerSample()==1 ? AUDIO_S8 : 0; + if(!format) { + fprintf(stderr, "DeviceAudioSdl WARNING: %i bit samples not supported.\n", sample.bitsPerSample()); + return 0; + } + SDL_AudioCVT cvt; + SDL_BuildAudioCVT(&cvt, format, sample.channels(), sample.sampleRate(), + m_spec.format, destChannels, m_spec.freq); + cvt.buf = new Uint8[sample.size()*cvt.len_mult]; + memcpy(cvt.buf, sample.data(), sample.size()); + cvt.len = sample.size(); + SDL_ConvertAudio(&cvt); + + _AudioTrack track; + track.data = cvt.buf; + track.dlen = cvt.len_cvt; + track.dpos = 0; + if (!track.dlen) { + fprintf(stderr, "DeviceAudioSdl WARNING: Sample has zero length.\n"); + return 0; + } + // adjust volume: + if((volume!=1.0f)&&m_isDesiredFormat) + adjustVolume((signed short *)track.data, track.dlen/2, volume); + + mm_sample.insert(make_pair(++m_sampleCounter,track)); + return m_sampleCounter; +} + +bool DeviceAudioSdl::sampleDestroy(unsigned int sample) { + // look for sample: + map::iterator iter=mm_sample.find(sample); + if( iter == mm_sample.end() ) return false; + // stop currently playing sounds referring to this sample: + SDL_LockAudio(); + for (unsigned int i=0; isecond.data) + ma_sound[i].isPlaying=false; + SDL_UnlockAudio(); + // cleanup: + delete iter->second.data; + if(iter->first==m_sampleCounter) --m_sampleCounter; + mm_sample.erase(iter); + return true; +} + +unsigned int DeviceAudioSdl::soundPlay(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch) { + // look for sample: + map::iterator iter=mm_sample.find(sample); + if(iter==mm_sample.end()) return 0; // no sample found + // look for an empty (or finished) sound track + unsigned int i; + for ( i=0; isecond.data; + ma_sound[i].dlen = iter->second.dlen; + ma_sound[i].dpos = 0; + ma_sound[i].volL=volumeL; + ma_sound[i].volR=volumeR; + ma_sound[i].disparity=disparity; + ma_sound[i].pitch=fabs(pitch); + ma_sound[i].isLoop=false; + ma_sound[i].isPlaying=true; + SDL_UnlockAudio(); + return i+1; +} + +unsigned int DeviceAudioSdl::soundLoop(unsigned int sample, float volumeL, float volumeR, float disparity, float pitch) { + unsigned int ret=soundPlay(sample,volumeL,volumeR,disparity, pitch); + if(ret) { + SDL_LockAudio(); + ma_sound[ret-1].isLoop=true; + SDL_UnlockAudio(); + } + return ret; +} + +bool DeviceAudioSdl::soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity, float pitch ) { + if(!sound || (sound>m_nSound) || !ma_sound[sound-1].isPlaying) return false; + SDL_LockAudio(); + ma_sound[--sound].volL=volumeL; + ma_sound[sound].volR=volumeR; + ma_sound[sound].disparity=disparity; + ma_sound[sound].pitch=fabs(pitch); + SDL_UnlockAudio(); + return true; +} + +bool DeviceAudioSdl::soundStop(unsigned int sound) { + if(!sound||(sound>m_nSound)||!ma_sound[sound-1].isPlaying) return false; + SDL_LockAudio(); + ma_sound[sound-1].isPlaying=false; + SDL_UnlockAudio(); + return true; +} + +void DeviceAudioSdl::soundStop() { + SDL_LockAudio(); + for (unsigned int i=0; i +}; +#include "proAudio.h" +#include + +/** @file proAudioSdl.h + \brief SDL backend of proteaAudio + \author Gerald Franz, www.viremo.de + \version 2.0 +*/ + +//--- class DeviceAudioSdl ----------------------------------------- + +/// internal class to manage sample data +class _AudioTrack; + +/// SDL based stereo audio mixer/playback device +class DeviceAudioSdl : public DeviceAudio { +public: + ///creates audio device + /** Use this method instead of a constructor. + \param nTracks (optional) the maximum number of sounds that are played parallely. Computation time is linearly correlated to this factor. + \param frequency (optional) sample frequency of the playback in Hz. 22050 corresponds to FM radio 44100 is CD quality. Computation time is linearly correlated to this factor. + \param chunkSize (optional) the number of bytes that are sent to the sound card at once. Low numbers lead to smaller latencies but need more computation time (thread switches). If a too small number is chosen, the sounds might not be played continuously. The default value 512 guarantees a good latency below 40 ms at 22050 Hz sample frequency. + \return a pointer to an audio device object in case of success + Note that the parameters are only handled when calling for the first time. Afterwards always the same object is returned until an explicit destroy() is called. + */ + static DeviceAudio* create(unsigned int nTracks=8, unsigned int frequency=22050, unsigned int chunkSize=1024); + + /// converts a sound sample to internal audio format, returns handle + virtual unsigned int sampleFromMemory(const AudioSample & sample, float volume=1.0f); + /// deletes a previously created sound sample resource identified by its handle + virtual bool sampleDestroy(unsigned int sample); + + /// plays a specified sample once and sets its parameters + /** \param sample a sample handle returned by a previous load() call + \param volumeL (optional) left volume + \param volumeR (optional) right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return a handle to the currently played sound or 0 in case of error */ + virtual unsigned int soundPlay(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f ); + /** plays a specified sample continuously and sets its parameters + \param sample a sample handle returned by a previous load() call + \param volumeL (optional) left volume + \param volumeR (optional) right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return a handle to the currently played sound or 0 in case of error */ + virtual unsigned int soundLoop(unsigned int sample, float volumeL=1.0f, float volumeR=1.0f, float disparity=0.0f, float pitch=1.0f ); + /// updates parameters of a specified sound + /** \param sound handle of a currently active sound + \param volumeL left volume + \param volumeR right volume + \param disparity (optional) time difference between left and right channel in seconds. Use negative values to specify a delay for the left channel, positive for the right. + \param pitch (optional) pitch factor for playback. 0.5 corresponds to one octave below, 2.0 to one above the original sample. + \return true in case the parameters have been updated successfully */ + virtual bool soundUpdate(unsigned int sound, float volumeL, float volumeR, float disparity=0.0f, float pitch=1.0f ); + /// stops a specified sound immediately + virtual bool soundStop(unsigned int sound); + /// stops all sounds immediately + virtual void soundStop(); + /// returns number of currently active sounds + virtual unsigned int soundActive() const; +protected: + /// constructor. Use the create() method instead + DeviceAudioSdl(unsigned int nTracks, unsigned int frequency, unsigned int chunkSize); + /// destructor. Use the destroy() method instead + virtual ~DeviceAudioSdl(); + /// stores audio specification + SDL_AudioSpec m_spec; + /// stores loaded sound samples + std::map mm_sample; + /// stores maximum sample id + unsigned int m_sampleCounter; + /// stores whether obtained audio format corresponds to expectations + bool m_isDesiredFormat; + + /// stores sounds to be mixed + _AudioTrack * ma_sound; + /// stores number of parallel tracks + unsigned int m_nSound; + + /// output callback + static void cbOutput(void *userData, Uint8 *stream, int len); + /// advanced mixer method + void mixOutputFloat(signed short *outputBuffer, unsigned int nFrames); + /// fallback mixer method + void mixOutputSInt(Uint8 *stream, int len); +}; + +#endif // _AUDIO_SDL_H diff --git a/protea.css b/protea.css new file mode 100644 index 0000000..e4213df --- /dev/null +++ b/protea.css @@ -0,0 +1,116 @@ + body { + font-family: Helvetica,sans-serif; + color: black; + background-color: white; + margin: 30px 50px 50px 50px; + text-align: left; + width: 640px; + font-size: 90%; +} + +h1, h2, h3, h4, h5, h6 { + color: maroon; +} + + h1 { + font-size: 150%; + } + + h2 { + margin: 25px 0px 10px 0px; + font-size: 110%; + } + + a:link, a:visited, a:link:active { + text-decoration: none; + color: maroon; + } + + a:link:hover, a:visited:hover { + text-decoration: underline; + color: red; + } + + p, ul { + font-size: 90%; + } + + pre { + background-color: #DDD; + width: 100%; + font-weight: bold; + } + + + +CAPTION { font-weight: bold } +DIV.qindex { + font-size: 90%; + width: 100%; + border-bottom: 1px solid #b0b0b0; + margin: 2px; + padding: 2px; + line-height: 140%; +} + +TD { + font-size: 90%; +} +TABLE.mdtable { + background-color: #DDD; + width: 100%; +} +TD.md { + font-weight: bold; +} +TD.mdPrefix { + color: #606060; +} +TD.mdname { + font-weight: bold; + color: #602020; +} + +DIV.groupHeader { + margin-left: 16px; + margin-top: 12px; + margin-bottom: 6px; + font-weight: bold; +} +DIV.groupText { margin-left: 16px; font-style: italic; font-size: 90% } +TD.indexkey { + font-weight: bold; + padding-right : 10px; + padding-top : 2px; + padding-left : 10px; + padding-bottom : 2px; + margin-left : 0px; + margin-right : 0px; + margin-top : 2px; + margin-bottom : 2px; + border-top: 1px solid #CCCCCC; + border-bottom: 1px solid #CCCCCC; +} +TD.indexvalue { + font-style: italic; + padding-right : 10px; + padding-top : 2px; + padding-left : 10px; + padding-bottom : 2px; + margin-left : 0px; + margin-right : 0px; + margin-top : 2px; + margin-bottom : 2px; + border-top: 1px solid #CCCCCC; + border-bottom: 1px solid #CCCCCC; +} +TR.memlist { + background-color: #DDD; +} +SPAN.keyword { color: #008000 } +SPAN.keywordtype { color: #604020 } +SPAN.keywordflow { color: #e08000 } +SPAN.comment { color: #800000 } +SPAN.preprocessor { color: #806020 } +SPAN.stringliteral { color: #002080 } +SPAN.charliteral { color: #008080 } diff --git a/proteaAudio.png b/proteaAudio.png new file mode 100644 index 0000000..c7f3a55 Binary files /dev/null and b/proteaAudio.png differ diff --git a/readme.txt b/readme.txt new file mode 100644 index 0000000..9fc6a70 --- /dev/null +++ b/readme.txt @@ -0,0 +1,63 @@ +/**\mainpage + +\section intro Overview +proteaAudio is a free cross-platform 2D audio library for C++ and Lua. + +Due to its straightforward interface and minimal open-source code base, proteaAudio is both easy to maintain and use. + +By default proteaAudio internally makes use of the excellent RtAudio low-level realtime audio input/output API, and therefore has no external dependencies apart from standard system libraries. Together with its liberal open-source licensing conditions (zlib style), this makes the integration of proteaAudio into both free and closed-source commercial software very easy. Alternatively, proteaAudio contains optional bindings for the highly portable SDL multimedia library and is therefore also usable on a plentitude of further platforms (e.g., iPhone, BEOS, FreeBSD). + +Despite its minimalistic design, proteaAudio offers advanced features such as dynamic pitch, per-channel volume control, and user-definable time shifts between channels. proteaAudio is capable of handling normal .wav files as well as ogg/vorbis audio samples (via stb_vorbis). Additionally it offers a simple interface for integrating further custom audio format loaders (e.g., .mp3). + +\section features Main features at a glance +- clean minimal C++ code base, portable, extendable class design +- supports playback of an arbitrary number of parallel mono/stereo sounds +- dynamic pitch shifts, panning between the stereo channels, user-definable disparity +- low CPU consumption, runs in its own dedicated thread +- regularly tested on MS Windows (MinGW/VisualStudio) and Linux (gcc) +- loaders for standard wav and ogg/vorbis audio samples included +- easily extensible to support further audio sample formats, e.g., mp3 +- C++ and Lua API +- by default makes use of the RtAudio low-level cross-platform audio backend, supporting Windows (DirectSound, ASIO), Linux (ALSA, OSS, JACK), and MacOSX (CoreAudio, JACK) +- makes optionally use of libSDL as cross-platform audio backend, supporting various further platforms + +\section download Download +- proteaAudio source code release 2009-02-04 +- proteaAudio Lua Windows/Linux binary release 2009-02-04 + +\section documentation Documentation +- C++ API +- Lua API +- Changelog + +\section links Links + +proteaAudio internally makes use of the following excellent open-source components: + +- RtAudio cross-platform low-level audio library (optional) +- SDL cross-platform multimedia layer (optional) +- stb_vorbis Ogg Vorbis audio decoder + + +\section license License notice (zlib license): + + (c) 2009 by Gerald Franz, www.viremo.de + + This software is provided 'as-is', without any express or implied + warranty. In no event will the author be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + -# The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + -# Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + -# This notice may not be removed or altered from any source distribution. + +*/ + \ No newline at end of file diff --git a/rtaudio/RtAudio.cpp b/rtaudio/RtAudio.cpp new file mode 100644 index 0000000..c456d33 --- /dev/null +++ b/rtaudio/RtAudio.cpp @@ -0,0 +1,7889 @@ +/************************************************************************/ +/*! \class RtAudio + \brief Realtime audio i/o C++ classes. + + RtAudio provides a common API (Application Programming Interface) + for realtime audio input/output across Linux (native ALSA, Jack, + and OSS), Macintosh OS X (CoreAudio and Jack), and Windows + (DirectSound and ASIO) operating systems. + + RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ + + RtAudio: realtime audio i/o C++ classes + Copyright (c) 2001-2009 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + +// RtAudio: Version 4.0.5 + +#include "RtAudio.h" +#include +#include +#include +#include + +// Static variable definitions. +const unsigned int RtApi::MAX_SAMPLE_RATES = 14; +const unsigned int RtApi::SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) + #define MUTEX_DESTROY(A) DeleteCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) +#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#else + #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions + #define MUTEX_DESTROY(A) abs(*A) // dummy definitions +#endif + +// *************************************************** // +// +// RtAudio definitions. +// +// *************************************************** // + +void RtAudio :: getCompiledApi( std::vector &apis ) throw() +{ + apis.clear(); + + // The order here will control the order of RtAudio's API search in + // the constructor. +#if defined(__UNIX_JACK__) + apis.push_back( UNIX_JACK ); +#endif +#if defined(__LINUX_ALSA__) + apis.push_back( LINUX_ALSA ); +#endif +#if defined(__LINUX_OSS__) + apis.push_back( LINUX_OSS ); +#endif +#if defined(__WINDOWS_ASIO__) + apis.push_back( WINDOWS_ASIO ); +#endif +#if defined(__WINDOWS_DS__) + apis.push_back( WINDOWS_DS ); +#endif +#if defined(__MACOSX_CORE__) + apis.push_back( MACOSX_CORE ); +#endif +#if defined(__RTAUDIO_DUMMY__) + apis.push_back( RTAUDIO_DUMMY ); +#endif +} + +void RtAudio :: openRtApi( RtAudio::Api api ) +{ +#if defined(__UNIX_JACK__) + if ( api == UNIX_JACK ) + rtapi_ = new RtApiJack(); +#endif +#if defined(__LINUX_ALSA__) + if ( api == LINUX_ALSA ) + rtapi_ = new RtApiAlsa(); +#endif +#if defined(__LINUX_OSS__) + if ( api == LINUX_OSS ) + rtapi_ = new RtApiOss(); +#endif +#if defined(__WINDOWS_ASIO__) + if ( api == WINDOWS_ASIO ) + rtapi_ = new RtApiAsio(); +#endif +#if defined(__WINDOWS_DS__) + if ( api == WINDOWS_DS ) + rtapi_ = new RtApiDs(); +#endif +#if defined(__MACOSX_CORE__) + if ( api == MACOSX_CORE ) + rtapi_ = new RtApiCore(); +#endif +#if defined(__RTAUDIO_DUMMY__) + if ( api == RTAUDIO_DUMMY ) + rtapi_ = new RtApiDummy(); +#endif +} + +RtAudio :: RtAudio( RtAudio::Api api ) throw() +{ + rtapi_ = 0; + + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; + + // No compiled support for specified API value. Issue a debug + // warning and continue as if no API was specified. + std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; + } + + // Iterate through the compiled APIs and return as soon as we find + // one with at least one device or we reach the end of the list. + std::vector< RtAudio::Api > apis; + getCompiledApi( apis ); + for ( unsigned int i=0; igetDeviceCount() ) break; + } + + if ( rtapi_ ) return; + + // It should not be possible to get here because the preprocessor + // definition __RTAUDIO_DUMMY__ is automatically defined if no + // API-specific definitions are passed to the compiler. But just in + // case something weird happens, we'll print out an error message. + std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n"; +} + +RtAudio :: ~RtAudio() throw() +{ + delete rtapi_; +} + +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options ) +{ + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // + +RtApi :: RtApi() +{ + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; + stream_.apiHandle = 0; + stream_.userBuffer[0] = 0; + stream_.userBuffer[1] = 0; + MUTEX_INITIALIZE( &stream_.mutex ); + showWarnings_ = true; +} + +RtApi :: ~RtApi() +{ + MUTEX_DESTROY( &stream_.mutex ); +} + +void RtApi :: openStream( RtAudio::StreamParameters *oParams, + RtAudio::StreamParameters *iParams, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options ) +{ + if ( stream_.state != STREAM_CLOSED ) { + errorText_ = "RtApi::openStream: a stream is already open!"; + error( RtError::INVALID_USE ); + } + + if ( oParams && oParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; + error( RtError::INVALID_USE ); + } + + if ( iParams && iParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; + error( RtError::INVALID_USE ); + } + + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtError::INVALID_USE ); + } + + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtError::INVALID_USE ); + } + + unsigned int nDevices = getDeviceCount(); + unsigned int oChannels = 0; + if ( oParams ) { + oChannels = oParams->nChannels; + if ( oParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: output device parameter value is invalid."; + error( RtError::INVALID_USE ); + } + } + + unsigned int iChannels = 0; + if ( iParams ) { + iChannels = iParams->nChannels; + if ( iParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: input device parameter value is invalid."; + error( RtError::INVALID_USE ); + } + } + + clearStreamInfo(); + bool result; + + if ( oChannels > 0 ) { + + result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) error( RtError::SYSTEM_ERROR ); + } + + if ( iChannels > 0 ) { + + result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + if ( oChannels > 0 ) closeStream(); + error( RtError::SYSTEM_ERROR ); + } + } + + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; + + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; +} + +unsigned int RtApi :: getDefaultInputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +unsigned int RtApi :: getDefaultOutputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! + return; +} + +bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // MUST be implemented in subclasses! + return FAILURE; +} + +void RtApi :: tickStreamTime( void ) +{ + // Subclasses that do not provide their own implementation of + // getStreamTime should call this function once per buffer I/O to + // provide basic stream time support. + + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +long RtApi :: getStreamLatency( void ) +{ + verifyStream(); + + long totalLatency = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + totalLatency = stream_.latency[0]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + totalLatency += stream_.latency[1]; + + return totalLatency; +} + +double RtApi :: getStreamTime( void ) +{ + verifyStream(); + +#if defined( HAVE_GETTIMEOFDAY ) + // Return a very accurate estimate of the stream time by + // adding in the elapsed time since the last tick. + struct timeval then; + struct timeval now; + + if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + return stream_.streamTime; + + gettimeofday( &now, NULL ); + then = stream_.lastTickTimestamp; + return stream_.streamTime + + ((now.tv_sec + 0.000001 * now.tv_usec) - + (then.tv_sec + 0.000001 * then.tv_usec)); +#else + return stream_.streamTime; +#endif +} + +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. +// +// A property listener is installed for over/underrun information. +// However, no functionality is currently provided to allow property +// listeners to trigger user handlers because it is unclear what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. +// +// The mechanism for querying and setting system parameters was +// updated (and perhaps simplified) in OS-X version 10.4. However, +// since 10.4 support is not necessarily available to all users, I've +// decided not to update the respective code at this time. Perhaps +// this will happen when Apple makes 10.4 free for everyone. :-) + +// A structure to hold various information related to the CoreAudio API +// implementation. +struct CoreHandle { + AudioDeviceID id[2]; // device ids + AudioDeviceIOProcID procId[2]; + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use + bool xrun[2]; + char *deviceBuffer; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + CoreHandle() + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiCore :: RtApiCore() +{ + // Nothing to do here. +} + +RtApiCore :: ~RtApiCore() +{ + // The subclass destructor gets called before the base class + // destructor, so close an existing stream before deallocating + // apiDeviceId memory. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiCore :: getDeviceCount( void ) +{ + // Find out how many audio devices there are, if any. + UInt32 dataSize; + OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; + error( RtError::WARNING ); + return 0; + } + + return dataSize / sizeof( AudioDeviceID ); +} + +unsigned int RtApiCore :: getDefaultInputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice, + &dataSize, &id ); + + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; + error( RtError::WARNING ); + return 0; + } + + dataSize *= nDevices; + AudioDeviceID deviceList[ nDevices ]; + result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; + error( RtError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; + error( RtError::WARNING ); + return info; + } + + AudioDeviceID id = deviceList[ device ]; + + // Get the device name. + info.name.erase(); + char name[256]; + dataSize = 256; + result = AudioDeviceGetProperty( id, 0, false, + kAudioDevicePropertyDeviceManufacturer, + &dataSize, name ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + info.name.append( (const char *)name, strlen(name) ); + info.name.append( ": " ); + + dataSize = 256; + result = AudioDeviceGetProperty( id, 0, false, + kAudioDevicePropertyDeviceName, + &dataSize, name ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + info.name.append( (const char *)name, strlen(name) ); + + // Get the output stream "configuration". + AudioBufferList *bufferList = nil; + result = AudioDeviceGetPropertyInfo( id, 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (result != noErr || dataSize == 0) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; + error( RtError::WARNING ); + return info; + } + + result = AudioDeviceGetProperty( id, 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if ( result != noErr ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + result = AudioDeviceGetPropertyInfo( id, 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (result != noErr || dataSize == 0) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; + error( RtError::WARNING ); + return info; + } + + result = AudioDeviceGetProperty( id, 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if ( result != noErr ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get input channel information. + nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Probe the device sample rates. + bool isInput = false; + if ( info.outputChannels == 0 ) isInput = true; + + // Determine the supported sample rates. + result = AudioDeviceGetPropertyInfo( id, 0, isInput, + kAudioDevicePropertyAvailableNominalSampleRates, + &dataSize, NULL ); + + if ( result != kAudioHardwareNoError || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyAvailableNominalSampleRates, + &dataSize, &rangeList ); + + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + Float64 minimumRate = 100000000.0, maximumRate = 0.0; + for ( UInt32 i=0; i maximumRate ) maximumRate = rangeList[i].mMaximum; + } + + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); + } + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // CoreAudio always uses 32-bit floating point data for PCM streams. + // Thus, any other "physical" formats supported by the device are of + // no interest to the client. + info.nativeFormats = RTAUDIO_FLOAT32; + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; + + info.probed = true; + return info; +} + +OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* inNow, + const AudioBufferList* inInputData, + const AudioTimeStamp* inInputTime, + AudioBufferList* outOutputData, + const AudioTimeStamp* inOutputTime, + void* infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) + return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +OSStatus deviceListener( AudioDeviceID inDevice, + UInt32 channel, + Boolean isInput, + AudioDevicePropertyID propertyID, + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + if ( propertyID == kAudioDeviceProcessorOverload ) { + if ( isInput ) + handle->xrun[1] = true; + else + handle->xrun[0] = true; + } + + return kAudioHardwareNoError; +} + +static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property ) +{ + OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL ); + return result == 0; +} + +bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; + return FAILURE; + } + + AudioDeviceID id = deviceList[ device ]; + + // Setup for stream mode. + bool isInput = false; + if ( mode == INPUT ) isInput = true; + + // Set or disable "hog" mode. + dataSize = sizeof( UInt32 ); + UInt32 doHog = 0; + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1; + result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the stream "configuration". + AudioBufferList *bufferList; + result = AudioDeviceGetPropertyInfo( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (result != noErr || dataSize == 0) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; + return FAILURE; + } + + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if ( result != noErr ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for one or more streams that contain the desired number of + // channels. CoreAudio devices can have an arbitrary number of + // streams and each stream can have an arbitrary number of channels. + // For each stream, a single buffer of interleaved samples is + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; + bool foundStream = false; + + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + firstStream = iStream; + channelOffset = offsetCounter; + foundStream = true; + break; + } + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. + if ( foundStream == false ) { + monoMode = true; + offsetCounter = firstChannel; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } + } + + free( bufferList ); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof( AudioValueRange ); + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyBufferFrameSizeRange, + &dataSize, &bufferRange ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; + else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; + + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + result = AudioDeviceSetProperty( id, NULL, 0, isInput, + kAudioDevicePropertyBufferFrameSize, + dataSize, &theSize ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = theSize; + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // Get the stream ID(s) so we can set the stream format. We'll have + // to do this for each stream. + AudioStreamID streamIDs[ nStreams ]; + dataSize = nStreams * sizeof( AudioStreamID ); + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreams, + &dataSize, &streamIDs ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Now set the stream format. Also, check the physical format of the + // device and change that if necessary. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + + bool updateFormat; + for ( UInt32 i=0; i 1.0 ) { + description.mSampleRate = (double) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, + kAudioStreamPropertyVirtualFormat, + dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now check the physical format. + result = AudioStreamGetProperty( streamIDs[firstStream+i], 0, + kAudioStreamPropertyPhysicalFormat, + &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) { + description.mFormatID = kAudioFormatLinearPCM; + AudioStreamBasicDescription testDescription = description; + unsigned long formatFlags; + + // We'll try higher bit rates first and then work our way down. + testDescription.mBitsPerChannel = 32; + formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 32; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 24; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 16; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 8; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + } + + // Get the stream latency. There can be latency in both the device + // and the stream. First, attempt to get the device latency on the + // master channel or the first open channel. Errors that might + // occur here are not deemed critical. + + // ***** CHECK THIS ***** // + UInt32 latency, channel = 0; + dataSize = sizeof( UInt32 ); + AudioDevicePropertyID property = kAudioDevicePropertyLatency; + if ( hasProperty( id, channel, isInput, property ) == true ) { + result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency ); + if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency; + else { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + } + + // Now try to get the stream latency. For multiple streams, I assume the + // latency is equal for each. + result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency ); + if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency; + else { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + + // Byte-swapping: According to AudioHardware.h, the stream data will + // always be presented in native-endian format, so we should never + // need to byte swap. + stream_.doByteSwap[mode] = false; + + // From the CoreAudio documentation, PCM data must be supplied as + // 32-bit floats. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + + if ( streamCount == 1 ) + stream_.nDeviceChannels[mode] = description.mChannelsPerFrame; + else // multiple streams + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our CoreHandle structure for the stream. + CoreHandle *handle = 0; + if ( stream_.apiHandle == 0 ) { + try { + handle = new CoreHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } + else + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; + handle->id[mode] = id; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) { + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) + // Only one callback procedure per device. + stream_.mode = DUPLEX; + else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() + result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; + errorText_ = errorStream_.str(); + goto error; + } + if ( stream_.mode == OUTPUT && mode == INPUT ) + stream_.mode = DUPLEX; + else + stream_.mode = mode; + } + + // Setup the device property listener for over/underload. + result = AudioDeviceAddPropertyListener( id, 0, isInput, + kAudioDeviceProcessorOverload, + deviceListener, (void *) handle ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiCore :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[0], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[1], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + // Destroy pthread condition variable. + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiCore :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiCore::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + result = AudioDeviceStart( handle->id[0], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + + result = AudioDeviceStart( handle->id[1], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == noErr ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiCore :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + + result = AudioDeviceStop( handle->id[0], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + + result = AudioDeviceStop( handle->id[1], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + stream_.state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == noErr ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiCore :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + handle->drainCounter = 1; + + stopStream(); +} + +bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ) +{ + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + pthread_cond_signal( &handle->condition ); + else + stopStream(); + return SUCCESS; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } + + AudioDeviceID outputDevice = handle->id[0]; + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream or duplex mode AND the input/output devices are + // different AND this function is called for the input device. + if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + if ( handle->nStreams[0] == 1 ) { + memset( outBufferList->mBuffers[handle->iStream[0]].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { // fill multiple streams with zeros + for ( unsigned int i=0; inStreams[0]; i++ ) { + memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); + } + } + } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); + } + else { // copy from user buffer + memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode + UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + bool inInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 inChannels = stream_.nUserChannels[0]; + if ( stream_.doConvertBuffer[0] ) { + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; inStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; idrainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { + + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; + + if ( stream_.deviceInterleaved[1] == false ) { // mono mode + UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); + } + } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + bool outInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 outChannels = stream_.nUserChannels[1]; + if ( stream_.doConvertBuffer[1] ) { + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + outChannels = stream_.nDeviceChannels[1]; + } + + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; inStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } + + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; i +#include + +// A structure to hold various information related to the Jack API +// implementation. +struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + JackHandle() + :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +void jackSilentError( const char * ) {}; + +RtApiJack :: RtApiJack() +{ + // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif +} + +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; + } + } + } while ( ports[++nChannels] ); + free( ports ); + } + + jack_client_close( client ); + return nDevices; +} + +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; + error( RtError::WARNING ); + return info; + } + + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + // Get the current jack server sample rate. + info.sampleRates.clear(); + info.sampleRates.push_back( jack_get_sample_rate( client ) ); + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } + + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { + jack_client_close(client); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtError::WARNING ); + return info; + } + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // Jack doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + jack_client_close(client); + info.probed = true; + return info; +} + +int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + + return 0; +} + +void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; + + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; + + object->closeStream(); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} + +int jackXrun( void *infoPointer ) +{ + JackHandle *handle = (JackHandle *) infoPointer; + + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; + + return 0; +} + +bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtError::WARNING ); + return FAILURE; + } + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } + + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; + + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports[ firstChannel ] ) + stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + free( ports ); + + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; + + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; + + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; + } + + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); + if ( bufferBytes < bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + goto error; + } + + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); + } + + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); + } + } + else { + for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + } + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); + + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); + + jack_client_close( handle->client ); + } + + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiJack :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK(&stream_.mutex); + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; + goto unlock; + } + + const char **ports; + + // Get the list of available ports. + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + goto unlock; + } + + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + goto unlock; + } + } + free(ports); + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + goto unlock; + } + + // Now make the port connections. See note above. + for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + goto unlock; + } + } + free(ports); + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK(&stream_.mutex); + + if ( result == 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiJack :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } + + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; + + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 1; + + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the jack_deactivate() +// function will return. +extern "C" void *jackStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->stopStream(); + + pthread_exit( NULL ); +} + +bool RtApiJack :: callbackEvent( unsigned long nframes ) +{ + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == true ) { + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + } + else + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } + + // Invoke user callback first, to get fresh output data. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter > 0 ) { // write zeros to the output stream + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } + + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); + } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); + } + } + } + + unlock: + MUTEX_UNLOCK(&stream_.mutex); + + RtApi::tickStreamTime(); + return SUCCESS; +} + //******************** End of __UNIX_JACK__ *********************// +#endif + +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS-X CoreAudio and Linux +// Jack. The primary constraint with ASIO is that it only allows +// access to a single driver at a time. Thus, it is not possible to +// have more than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. +// +// On unix systems, we make use of a pthread condition variable. +// Since there is no equivalent in Windows, I hacked something based +// on information found in +// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. + +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include + +AsioDrivers drivers; +ASIOCallbacks asioCallbacks; +ASIODriverInfo driverInfo; +CallbackInfo *asioCallbackInfo; +bool asioXRun; + +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; + + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; + +// Function declarations (definitions at end of section) +static const char* getAsioErrorString( ASIOError result ); +void sampleRateChanged( ASIOSampleRate sRate ); +long asioMessages( long selector, long value, void* message, double* opt ); + +RtApiAsio :: RtApiAsio() +{ + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} + +unsigned int RtApiAsio :: getDeviceCount( void ) +{ + return (unsigned int) drivers.asioGetNumDev(); +} + +RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + // If a stream is already open, we cannot probe other devices. Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; + error( RtError::WARNING ); + return info; + } + return devices_[ device ]; + } + + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; iobject; + object->callbackEvent( index ); +} + +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; isaveDeviceInfo(); + + // Only load the driver once for duplex stream. + if ( mode != INPUT || stream_.mode != OUTPUT ) { + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + else channelInfo.isInput = true; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Assuming WINDOWS host is always little-endian. + stream_.doByteSwap[mode] = false; + stream_.userFormat = format; + stream_.deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; + } + + if ( stream_.deviceFormat[mode] == 0 ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + long minSize, maxSize, preferSize, granularity; + result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } + + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } + + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Allocate, if necessary, our AsioHandle structure for the stream. + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + //if ( handle == NULL ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + return FAILURE; + } + handle->bufferInfos = 0; + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + long inputLatency, outputLatency; + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + bool buffersAllocated = false; + unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; iisInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; + } + + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + + // Determine device latencies + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtError::WARNING); // warn but don't fail + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + return SUCCESS; + + error: + if ( buffersAllocated ) + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; + errorText_ = errorStream_.str(); + goto unlock; + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + asioXRun = false; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAsio :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); + } + } + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completely + // disposed. So now, calling abort is the same as calling stop. + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 1; + stopStream(); +} + +bool RtApiAsio :: callbackEvent( long bufferIndex ) +{ + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return SUCCESS; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } + + } + else { + + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } + + } + + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + + if (stream_.doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + } + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } + } + + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); + + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +void sampleRateChanged( ASIOSampleRate sRate ) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } + + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; +} + +long asioMessages( long selector, long value, void* message, double* opt ) +{ + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} + +static const char* getAsioErrorString( ASIOError result ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; + + static Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; + + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; + + return "Unknown error."; +} +//******************** End of __WINDOWS_ASIO__ *********************// +#endif + + +#if defined(__WINDOWS_DS__) // Windows DirectSound API + +// Modified by Robin Davies, October 2005 +// - Improvements to DirectX pointer chasing. +// - Backdoor RtDsStatistics hook provides DirectX performance information. +// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. +// - Auto-call CoInitialize for DSOUND and ASIO platforms. +// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 + +#include +#include + +#if defined(__MINGW32__) + // missing from latest mingw winapi +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( laterPointer > earlierPointer ) + return laterPointer - earlierPointer; + else + return laterPointer - earlierPointer + bufferSize; +} + +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} + +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +/* +RtApiDs::RtDsStatistics RtApiDs::statistics; + +// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. +RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() +{ + RtDsStatistics s = statistics; + + // update the calculated fields. + if ( s.inputFrameSize != 0 ) + s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate; + + if ( s.outputFrameSize != 0 ) + s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate; + + return s; +} +*/ + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static char* getErrorString( int code ); + +extern "C" unsigned __stdcall callbackHandler( void *ptr ); + +struct EnumInfo { + bool isInput; + bool getDefault; + bool findIndex; + unsigned int counter; + unsigned int index; + LPGUID id; + std::string name; + + EnumInfo() + : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} +}; + +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} + +RtApiDs :: ~RtApiDs() +{ + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + // Count output devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } + + // Now enumerate input devices until we find the id = NULL. + info.isInput = true; + info.getDefault = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } + + if ( info.counter > 0 ) return info.counter - 1; + return 0; +} + +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + // Enumerate output devices until we find the id = NULL. + EnumInfo info; + info.getDefault = true; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } + + if ( info.counter > 0 ) return info.counter - 1; + return 0; +} + +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Count DirectSound devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + + // Count DirectSoundCapture devices. + info.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + + return info.counter; +} + +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + // Because DirectSound always enumerates input and output devices + // separately (and because we don't attempt to combine devices + // internally), none of our "devices" will ever be duplex. + + RtAudio::DeviceInfo info; + info.probed = false; + + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + + if ( dsinfo.name.empty() ) goto probeInput; + + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); + } + + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + + // Copy name and return. + info.name = dsinfo.name; + + info.probed = true; + return info; + + probeInput: + + dsinfo.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + + if ( dsinfo.name.empty() ) return info; + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // Get sample rate and format information. + if ( inCaps.dwChannels == 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); + } + } + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error + + input->Release(); + + if ( info.inputChannels == 0 ) return info; + + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; + + // Copy name and return. + info.name = dsinfo.name; + info.probed = true; + return info; +} + +bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } + + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( mode == OUTPUT ) { + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT + dsinfo.isInput = true; + HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. However, for console + // applications, no sound was produced when using GetDesktopWindow(). + HWND hWnd = GetForegroundWindow(); + + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; + + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the device buffer size. By default, 32k, but we will + // grow it to make allowances for very large software buffer sizes. + DWORD dsBufferSize = 0; + DWORD dsPointerLeadTime = 0; + long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this + + void *ohandle = 0, *bhandle = 0; + if ( mode == OUTPUT ) { + + LPDIRECTSOUND output; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) + bufferBytes *= 2; + + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Setup the secondary DS buffer description. + dsBufferSize = (DWORD) bufferBytes; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = bufferBytes; + bufferDescription.lpwfxFormat = &waveFormat; + + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + bufferBytes = dsbcaps.dwBufferBytes; + + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = bufferBytes; + ohandle = (void *) output; + bhandle = (void *) buffer; + } + + if ( mode == INPUT ) { + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } + + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) + bufferBytes *= 2; + + // Setup the secondary DS buffer description. + dsBufferSize = bufferBytes; + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = bufferBytes; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the buffer size ... might be different from what we specified. + DSCBCAPS dscbcaps; + dscbcaps.dwSize = sizeof( DSCBCAPS ); + result = buffer->GetCaps( &dscbcaps ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + bufferBytes = dscbcaps.dwBufferBytes; + + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = bufferBytes; + ohandle = (void *) input; + bhandle = (void *) buffer; + } + + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup the callback thread. + unsigned threadId; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } + + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + return SUCCESS; + + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. + + MUTEX_LOCK( &stream_.mutex ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + timeBeginPeriod( 1 ); + + /* + memset( &statistics, 0, sizeof( statistics ) ); + statistics.sampleRate = stream_.sampleRate; + statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; + */ + + buffersRolling = false; + duplexPrerollBytes = 0; + + if ( stream_.mode == DUPLEX ) { + // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. + duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); + } + + HRESULT result = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; + + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; + + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + result = buffer->Start( DSCBSTART_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + HRESULT result = 0; + LPVOID audioPtr; + DWORD dataLen; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); + } + + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } + + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +} + +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 1; + + stopStream(); +} + +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED ) { + Sleep(50); // sleep 50 milliseconds + return; + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + HRESULT result; + DWORD currentWritePos, safeWritePos; + DWORD currentReadPos, safeReadPos; + DWORD leadPos; + UINT nextWritePos; + +#ifdef GENERATE_DEBUG_LOG + DWORD writeTime, readTime; +#endif + + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + + char *buffer; + long bufferBytes; + + if ( stream_.mode == DUPLEX && !buffersRolling ) { + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. + + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. + + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + + DWORD initialWritePos, initialSafeWritePos; + DWORD initialReadPos, initialSafeReadPos; + + result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; + Sleep( 1 ); + } + + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + buffersRolling = true; + handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); + handle->bufferPointer[1] = safeReadPos; + } + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + // No byte swapping necessary in DirectSound implementation. + + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; idsBufferSize[0]; + nextWritePos = handle->bufferPointer[0]; + + DWORD endWrite; + while ( true ) { + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + leadPos = safeWritePos + handle->dsPointerLeadTime[0]; + if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; + if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset + endWrite = nextWritePos + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPos >= endWrite ) break; + + // If we are here, then we must wait until the play pointer gets + // beyond the write region. The approach here is to use the + // Sleep() function to suspend operation until safePos catches + // up. Calculate number of milliseconds to wait as: + // time = distance * (milliseconds/second) * fudgefactor / + // ((bytes/sample) * (samples/second)) + // A "fudgefactor" less than 1 is used because it was found + // that sleeping too long was MUCH worse than sleeping for + // several shorter periods. + double millis = ( endWrite - leadPos ) * 900.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + if ( millis > 50.0 ) { + static int nOverruns = 0; + ++nOverruns; + } + Sleep( (DWORD) millis ); + } + + //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) { + // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); + //} + + if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + //++statistics.numberOfWriteUnderruns; + handle->xrun[0] = true; + nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; + while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; + handle->bufferPointer[0] = nextWritePos; + endWrite = nextWritePos + bufferBytes; + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePos; + + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPos = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPos + bufferBytes; + + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. + + if ( stream_.mode == DUPLEX ) { + if ( safeReadPos < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPos; + + handle->xrun[1] = true; + //++statistics.numberOfReadOverruns; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPos = safeReadPos-2*bufferBytes; + else + nextReadPos = safeReadPos-bufferBytes-adjustment; + + //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; + //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; + if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPos = safeReadPos-bufferBytes; + while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + } + endRead = nextReadPos + bufferBytes; + } + } + else { // mode == INPUT + while ( safeReadPos < endRead ) { + // See comments for playback. + double millis = (endRead - safeReadPos) * 900.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + } + } + + //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) + // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); + + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; + } + + // Update our buffer offset and unlock sound buffer + nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + handle->bufferPointer[1] = nextReadPos; + + // No byte swapping necessary in DirectSound implementation. + + // If necessary, convert 8-bit data from unsigned to signed. + if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) + for ( int j=0; jobject; + bool* isRunning = &info->isRunning; + + while ( *isRunning == true ) { + object->callbackEvent(); + } + + _endthreadex( 0 ); + return 0; +} + +#include "tchar.h" + +std::string convertTChar( LPCTSTR name ) +{ + std::string s; + +#if defined( UNICODE ) || defined( _UNICODE ) + // Yes, this conversion doesn't make sense for two-byte characters + // but RtAudio is currently written to return an std::string of + // one-byte chars for the device name. + for ( unsigned int i=0; iisInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + info->counter++; + } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + info->counter++; + } + object->Release(); + } + + if ( info->getDefault && lpguid == NULL ) return FALSE; + + if ( info->findIndex && info->counter > info->index ) { + info->id = lpguid; + info->name = convertTChar( description ); + return FALSE; + } + + return TRUE; +} + +static char* getErrorString( int code ) +{ + switch ( code ) { + + case DSERR_ALLOCATED: + return "Already allocated"; + + case DSERR_CONTROLUNAVAIL: + return "Control unavailable"; + + case DSERR_INVALIDPARAM: + return "Invalid parameter"; + + case DSERR_INVALIDCALL: + return "Invalid call"; + + case DSERR_GENERIC: + return "Generic error"; + + case DSERR_PRIOLEVELNEEDED: + return "Priority level needed"; + + case DSERR_OUTOFMEMORY: + return "Out of memory"; + + case DSERR_BADFORMAT: + return "The sample rate or the channel format is not supported"; + + case DSERR_UNSUPPORTED: + return "Not supported"; + + case DSERR_NODRIVER: + return "No driver"; + + case DSERR_ALREADYINITIALIZED: + return "Already initialized"; + + case DSERR_NOAGGREGATION: + return "No aggregation"; + + case DSERR_BUFFERLOST: + return "Buffer lost"; + + case DSERR_OTHERAPPHASPRIO: + return "Another application already has priority"; + + case DSERR_UNINITIALIZED: + return "Uninitialized"; + + default: + return "DirectSound unknown error"; + } +} +//******************** End of __WINDOWS_DS__ *********************// +#endif + + +#if defined(__LINUX_ALSA__) + +#include +#include + + // A structure to hold various information related to the ALSA API + // implementation. +struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable; + + AlsaHandle() + :synchronized(false) { xrun[0] = false; xrun[1] = false; } +}; + +extern "C" void *alsaCallbackHandler( void * ptr ); + +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} + +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiAlsa :: getDeviceCount( void ) +{ + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &handle, name, 0 ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( handle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + return nDevices; +} + +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *chandle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + break; + } + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; + } + nDevices++; + } + nextcard: + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + foundDevice: + + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtError::WARNING ); + return info; + } + return devices_[ device ]; + } + + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); + + // First try for playback + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + snd_pcm_info_set_stream( pcminfo, stream ); + + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } + + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + // Now try for capture + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + probeParameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if ( info.outputChannels >= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i= 0 ) + sprintf( name, "hw:%s,%d", cardname, subdevice ); + info.name = name; + + // That's all ... close the device and return + snd_pcm_close( phandle ); + info.probed = true; + return info; +} + +void RtApiAlsa :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; i= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) break; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + foundDevice: + + // The getDeviceInfo() function will not work for a device that is + // already open. Thus, we'll probe the system before opening a + // stream and save the results for use by getDeviceInfo(). + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once + this->saveDeviceInfo(); + + snd_pcm_stream_t stream; + if ( mode == OUTPUT ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + + snd_pcm_t *phandle; + int openMode = SND_PCM_ASYNC; + result = snd_pcm_open( &phandle, name, stream, openMode ); + if ( result < 0 ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; + else + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca( &hw_params ); + result = snd_pcm_hw_params_any( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set access ... check user preference. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { + stream_.userInterleaved = false; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + stream_.deviceInterleaved[mode] = true; + } + else + stream_.deviceInterleaved[mode] = false; + } + else { + stream_.userInterleaved = true; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + stream_.deviceInterleaved[mode] = false; + } + else + stream_.deviceInterleaved[mode] = true; + } + + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + + if ( format == RTAUDIO_SINT8 ) + deviceFormat = SND_PCM_FORMAT_S8; + else if ( format == RTAUDIO_SINT16 ) + deviceFormat = SND_PCM_FORMAT_S16; + else if ( format == RTAUDIO_SINT24 ) + deviceFormat = SND_PCM_FORMAT_S24; + else if ( format == RTAUDIO_SINT32 ) + deviceFormat = SND_PCM_FORMAT_S32; + else if ( format == RTAUDIO_FLOAT32 ) + deviceFormat = SND_PCM_FORMAT_FLOAT; + else if ( format == RTAUDIO_FLOAT64 ) + deviceFormat = SND_PCM_FORMAT_FLOAT64; + + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { + stream_.deviceFormat[mode] = format; + goto setFormat; + } + + // The user requested format is not natively supported by the device. + deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + goto setFormat; + } + + // If we get here, no supported format was found. + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + + setFormat: + result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine whether byte-swaping is necessary. + stream_.doByteSwap[mode] = false; + if ( deviceFormat != SND_PCM_FORMAT_S8 ) { + result = snd_pcm_format_cpu_endian( deviceFormat ); + if ( result == 0 ) + stream_.doByteSwap[mode] = true; + else if (result < 0) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Set the sample rate. + result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream_.nUserChannels[mode] = channels; + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); + unsigned int deviceChannels = value; + if ( result < 0 || deviceChannels < channels + firstChannel ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + deviceChannels = value; + if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; + stream_.nDeviceChannels[mode] = deviceChannels; + + // Set the device channels. + result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the buffer number, which in ALSA is referred to as the "period". + int totalSize, dir; + unsigned int periods = 0; + if ( options ) periods = options->numberOfBuffers; + totalSize = *bufferSize * periods; + + // Set the buffer (or period) size. + snd_pcm_uframes_t periodSize = *bufferSize; + result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + *bufferSize = periodSize; + + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + else periods = totalSize / *bufferSize; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if ( periods < 2 ) periods = 2; + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + + // Install the hardware configuration + result = snd_pcm_hw_params( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca( &sw_params ); + snd_pcm_sw_params_current( phandle, sw_params ); + snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + + result = snd_pcm_sw_params( phandle, sw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); + snd_pcm_sw_params_dump( sw_params, out ); +#endif + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the ApiHandle if necessary and then save. + AlsaHandle *apiInfo = 0; + if ( stream_.apiHandle == 0 ) { + try { + apiInfo = (AlsaHandle *) new AlsaHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; + goto error; + } + + if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; + } + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; + } + apiInfo->handles[mode] = phandle; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.nBuffers = periods; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + // Link the streams if possible. + apiInfo->synchronized = false; + if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) + apiInfo->synchronized = true; + else { + errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; + error( RtError::WARNING ); + } + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &apiInfo->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[0] ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[1] ); + } + + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiAlsa :: startStream() +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. + + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + snd_pcm_state_t state; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + state = snd_pcm_state( handle[0] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + state = snd_pcm_state( handle[1] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } + + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + pthread_cond_signal( &apiInfo->runnable ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) + result = snd_pcm_drop( handle[0] ); + else + result = snd_pcm_drain( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = snd_pcm_drop( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } + + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtError::WARNING ); + goto tryOutput; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } + + tryOutput: + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtError::WARNING ); + goto unlock; + } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +extern "C" void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// +#endif + + +#if defined(__LINUX_OSS__) + +#include +#include +#include +#include +#include "soundcard.h" +#include +#include + +extern "C" void *ossCallbackHandler(void * ptr); + +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; + + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} + +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiOss :: getDeviceCount( void ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return 0; + } + + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return 0; + } + + close( mixerfd ); + return sysinfo.numaudios; +} + +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return info; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return info; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } + + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Probe channels + if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_DUPLEX ) { + if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + } + + // Probe data formats ... do for input + unsigned long mask = ainfo.iformats; + if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) + info.nativeFormats |= RTAUDIO_SINT16; + if ( mask & AFMT_S8 ) + info.nativeFormats |= RTAUDIO_SINT8; + if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) + info.nativeFormats |= RTAUDIO_SINT32; + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i= (int) SAMPLE_RATES[k] ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); + } + } + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + else { + info.probed = true; + info.name = ainfo.name; + } + + return info; +} + + +bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( mode == OUTPUT ) + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; + } + else + flags |= O_RDONLY; + } + + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; + + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + */ + + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = deviceChannels; + + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + } + + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; + + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; + + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Verify the sample rate setup worked. + if ( abs( srate - sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } + + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiOss :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + else + ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + stream_.state = STREAM_STOPPED; + } + + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiOss :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); +} + +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + memset( buffer, 0, samples * formatBytes(format) ); + for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); + if ( result == -1 ) { + errorText_ = "RtApiOss::stopStream: audio write error."; + error( RtError::WARNING ); + } + } + + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiOss :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiOss :: callbackEvent() +{ + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } + + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( buffer, samples, format ); + + if ( stream_.mode == DUPLEX && handle->triggered == false ) { + int trig = 0; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + handle->triggered = true; + } + else + // Write samples to device. + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an underrun, though there isn't a + // specific means for determining that. + handle->xrun[0] = true; + errorText_ = "RtApiOss::callbackEvent: audio write error."; + error( RtError::WARNING ); + // Continue on to input section. + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an overrun, though there isn't a + // specific means for determining that. + handle->xrun[1] = true; + errorText_ = "RtApiOss::callbackEvent: audio read error."; + error( RtError::WARNING ); + goto unlock; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +extern "C" void *ossCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_OSS__ *********************// +#endif + + +// *************************************************** // +// +// Protected common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message printing. +void RtApi :: error( RtError::Type type ) +{ + errorStream_.str(""); // clear the ostringstream + if ( type == RtError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else + throw( RtError( errorText_, type ) ); +} + +void RtApi :: verifyStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtError::INVALID_USE ); + } +} + +void RtApi :: clearStreamInfo() +{ + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; + stream_.sampleRate = 0; + stream_.bufferSize = 0; + stream_.nBuffers = 0; + stream_.userFormat = 0; + stream_.userInterleaved = true; + stream_.streamTime = 0.0; + stream_.apiHandle = 0; + stream_.deviceBuffer = 0; + stream_.callbackInfo.callback = 0; + stream_.callbackInfo.userData = 0; + stream_.callbackInfo.isRunning = false; + for ( int i=0; i<2; i++ ) { + stream_.device[i] = 11111; + stream_.doConvertBuffer[i] = false; + stream_.deviceInterleaved[i] = true; + stream_.doByteSwap[i] = false; + stream_.nUserChannels[i] = 0; + stream_.nDeviceChannels[i] = 0; + stream_.channelOffset[i] = 0; + stream_.deviceFormat[i] = 0; + stream_.latency[i] = 0; + stream_.userBuffer[i] = 0; + stream_.convertInfo[i].channels = 0; + stream_.convertInfo[i].inJump = 0; + stream_.convertInfo[i].outJump = 0; + stream_.convertInfo[i].inFormat = 0; + stream_.convertInfo[i].outFormat = 0; + stream_.convertInfo[i].inOffset.clear(); + stream_.convertInfo[i].outOffset.clear(); + } +} + +unsigned int RtApi :: formatBytes( RtAudioFormat format ) +{ + if ( format == RTAUDIO_SINT16 ) + return 2; + else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || + format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtError::WARNING ); + + return 0; +} + +void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) +{ + if ( mode == INPUT ) { // convert device to user buffer + stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; + stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; + stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; + stream_.convertInfo[mode].outFormat = stream_.userFormat; + } + else { // convert user to device buffer + stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; + stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; + stream_.convertInfo[mode].inFormat = stream_.userFormat; + stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; + } + + if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; + else + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; + + // Set up the interleave/deinterleave offsets. + if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { + if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || + ( mode == INPUT && stream_.userInterleaved ) ) { + for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; k>= 8; + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i>8) | (x<<8); } + //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } + //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } + +void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) +{ + register char val; + register char *ptr; + + ptr = buffer; + if ( format == RTAUDIO_SINT16 ) { + for ( unsigned int i=0; i +#include +#include "RtError.h" + +/*! \typedef typedef unsigned long RtAudioFormat; + \brief RtAudio data format type. + + Support for signed integers and floats. Audio data fed to/from an + RtAudio stream is assumed to ALWAYS be in host byte order. The + internal routines will automatically take care of any necessary + byte-swapping between the host format and the soundcard. Thus, + endian-ness is not a concern in the following format definitions. + + - \e RTAUDIO_SINT8: 8-bit signed integer. + - \e RTAUDIO_SINT16: 16-bit signed integer. + - \e RTAUDIO_SINT24: Upper 3 bytes of 32-bit signed integer. + - \e RTAUDIO_SINT32: 32-bit signed integer. + - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. + - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. +*/ +typedef unsigned long RtAudioFormat; +static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // Lower 3 bytes of 32-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. +static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. +static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. + +/*! \typedef typedef unsigned long RtAudioStreamFlags; + \brief RtAudio stream option flags. + + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows Direct Sound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. +*/ +typedef unsigned int RtAudioStreamFlags; +static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). +static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. +static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. + +/*! \typedef typedef unsigned long RtAudioStreamStatus; + \brief RtAudio stream status (over- or underflow) flags. + + Notification of a stream over- or underflow is indicated by a + non-zero stream \c status argument in the RtAudioCallback function. + The stream status can be one of the following two options, + depending on whether the stream is open for output and/or input: + + - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. + - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. +*/ +typedef unsigned int RtAudioStreamStatus; +static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. +static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. + +//! RtAudio callback function prototype. +/*! + All RtAudio clients must create a function of type RtAudioCallback + to read and/or write data from/to the audio stream. When the + underlying audio system is ready for new input or output data, this + function will be invoked. + + \param outputBuffer For output (or duplex) streams, the client + should write \c nFrames of audio sample frames into this + buffer. This argument should be recast to the datatype + specified when the stream was opened. For input-only + streams, this argument will be NULL. + + \param inputBuffer For input (or duplex) streams, this buffer will + hold \c nFrames of input audio sample frames. This + argument should be recast to the datatype specified when the + stream was opened. For output-only streams, this argument + will be NULL. + + \param nFrames The number of sample frames of input or output + data in the buffers. The actual buffer size in bytes is + dependent on the data type and number of channels in use. + + \param streamTime The number of seconds that have elapsed since the + stream was started. + + \param status If non-zero, this argument indicates a data overflow + or underflow condition for the stream. The particular + condition can be determined by comparison with the + RtAudioStreamStatus flags. + + \param userData A pointer to optional data provided by the client + when opening the stream (default = NULL). + + To continue normal stream operation, the RtAudioCallback function + should return a value of zero. To stop the stream and drain the + output buffer, the function should return a value of one. To abort + the stream immediately, the client should return a value of two. + */ +typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, + unsigned int nFrames, + double streamTime, + RtAudioStreamStatus status, + void *userData ); + + +// **************************************************************** // +// +// RtAudio class declaration. +// +// RtAudio is a "controller" used to select an available audio i/o +// interface. It presents a common API for the user to call but all +// functionality is implemented by the class RtApi and its +// subclasses. RtAudio creates an instance of an RtApi subclass +// based on the user's API choice. If no choice is made, RtAudio +// attempts to make a "logical" API selection. +// +// **************************************************************** // + +class RtApi; + +class RtAudio +{ + public: + + //! Audio API specifier arguments. + enum Api { + UNSPECIFIED, /*!< Search for a working compiled API. */ + LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ + LINUX_OSS, /*!< The Linux Open Sound System API. */ + UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ + MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ + WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ + WINDOWS_DS, /*!< The Microsoft Direct Sound API. */ + RTAUDIO_DUMMY /*!< A compilable but non-functional API. */ + }; + + //! The public device information structure for returning queried values. + struct DeviceInfo { + bool probed; /*!< true if the device capabilities were successfully probed. */ + std::string name; /*!< Character string device identifier. */ + unsigned int outputChannels; /*!< Maximum output channels supported by device. */ + unsigned int inputChannels; /*!< Maximum input channels supported by device. */ + unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */ + bool isDefaultOutput; /*!< true if this is the default output device. */ + bool isDefaultInput; /*!< true if this is the default input device. */ + std::vector sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ + RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */ + + // Default constructor. + DeviceInfo() + :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0), + isDefaultOutput(false), isDefaultInput(false), nativeFormats(0) {} + }; + + //! The structure for specifying input or ouput stream parameters. + struct StreamParameters { + unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */ + unsigned int nChannels; /*!< Number of channels. */ + unsigned int firstChannel; /*!< First channel index on device (default = 0). */ + + // Default constructor. + StreamParameters() + : deviceId(0), nChannels(0), firstChannel(0) {} + }; + + //! The structure for specifying stream options. + /*! + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows Direct Sound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + The \c numberOfBuffers parameter can be used to control stream + latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs + only. A value of two is usually the smallest allowed. Larger + numbers can potentially result in more robust stream performance, + though likely at the cost of stream latency. The value set by the + user is replaced during execution of the RtAudio::openStream() + function by the value actually used by the system. + + The \c streamName parameter can be used to set the client name + when using the Jack API. By default, the client name is set to + RtApiJack. However, if you wish to create multiple instances of + RtAudio with Jack, each instance must have a unique client name. + */ + struct StreamOptions { + RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). */ + unsigned int numberOfBuffers; /*!< Number of stream buffers. */ + std::string streamName; /*!< A stream name (currently used only in Jack). */ + + // Default constructor. + StreamOptions() + : flags(0), numberOfBuffers(0) {} + }; + + //! A static function to determine the available compiled audio APIs. + /*! + The values returned in the std::vector can be compared against + the enumerated list values. Note that there can be more than one + API compiled for certain operating systems. + */ + static void getCompiledApi( std::vector &apis ) throw(); + + //! The class constructor. + /*! + The constructor performs minor initialization tasks. No exceptions + can be thrown. + + If no API argument is specified and multiple API support has been + compiled, the default order of use is JACK, ALSA, OSS (Linux + systems) and ASIO, DS (Windows systems). + */ + RtAudio( RtAudio::Api api=UNSPECIFIED ) throw(); + + //! The destructor. + /*! + If a stream is running or open, it will be stopped and closed + automatically. + */ + ~RtAudio() throw(); + + //! Returns the audio API specifier for the current instance of RtAudio. + RtAudio::Api getCurrentApi( void ) throw(); + + //! A public function that queries for the number of audio devices available. + /*! + This function performs a system query of available devices each time it + is called, thus supporting devices connected \e after instantiation. If + a system error occurs during processing, a warning will be issued. + */ + unsigned int getDeviceCount( void ) throw(); + + //! Return an RtAudio::DeviceInfo structure for a specified device number. + /*! + + Any device integer between 0 and getDeviceCount() - 1 is valid. + If an invalid argument is provided, an RtError (type = INVALID_USE) + will be thrown. If a device is busy or otherwise unavailable, the + structure member "probed" will have a value of "false" and all + other members are undefined. If the specified device is the + current default input or output device, the corresponding + "isDefault" member will have a value of "true". + */ + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + + //! A function that returns the index of the default output device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultOutputDevice( void ) throw(); + + //! A function that returns the index of the default input device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultInputDevice( void ) throw(); + + //! A public function for opening a stream with the specified parameters. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be + opened with the specified parameters or an error occurs during + processing. An RtError (type = INVALID_USE) is thrown if any + invalid device ID or channel number parameters are specified. + + \param outputParameters Specifies output stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For input-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param inputParameters Specifies input stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For output-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param format An RtAudioFormat specifying the desired sample data format. + \param sampleRate The desired sample rate (sample frames per second). + \param *bufferFrames A pointer to a value indicating the desired + internal buffer size in sample frames. The actual value + used by the device is returned via the same pointer. A + value of zero can be specified, in which case the lowest + allowable value is determined. + \param callback A client-defined function that will be invoked + when input data is available and/or output data is needed. + \param userData An optional pointer to data that can be accessed + from within the callback function. + \param options An optional pointer to a structure containing various + global stream options, including a list of OR'ed RtAudioStreamFlags + and a suggested number of stream buffers that can be used to + control stream latency. More buffers typically result in more + robust performance, though at a cost of greater latency. If a + value of zero is specified, a system-specific median value is + chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the + lowest allowable value is used. The actual value used is + returned via the structure argument. The parameter is API dependent. + */ + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData = NULL, RtAudio::StreamOptions *options = NULL ); + + //! A function that closes a stream and frees any associated stream memory. + /*! + If a stream is not open, this function issues a warning and + returns (no exception is thrown). + */ + void closeStream( void ) throw(); + + //! A function that starts a stream. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + running. + */ + void startStream( void ); + + //! Stop a stream, allowing any samples remaining in the output queue to be played. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void stopStream( void ); + + //! Stop a stream, discarding any samples remaining in the input/output queue. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void abortStream( void ); + + //! Returns true if a stream is open and false if not. + bool isStreamOpen( void ) throw(); + + //! Returns true if the stream is running and false if it is stopped or not open. + bool isStreamRunning( void ) throw(); + + //! Returns the number of elapsed seconds since the stream was started. + /*! + If a stream is not open, an RtError (type = INVALID_USE) will be thrown. + */ + double getStreamTime( void ); + + //! Returns the internal stream latency in sample frames. + /*! + The stream latency refers to delay in audio input and/or output + caused by internal buffering by the audio system and/or hardware. + For duplex streams, the returned value will represent the sum of + the input and output latencies. If a stream is not open, an + RtError (type = INVALID_USE) will be thrown. If the API does not + report latency, the return value will be zero. + */ + long getStreamLatency( void ); + + //! Specify whether warning messages should be printed to stderr. + void showWarnings( bool value = true ) throw(); + + protected: + + void openRtApi( RtAudio::Api api ); + RtApi *rtapi_; +}; + +// Operating system dependent thread functionality. +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) + #include + #include + + typedef unsigned long ThreadHandle; + typedef CRITICAL_SECTION StreamMutex; + +#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // Using pthread library for various flavors of unix. + #include + + typedef pthread_t ThreadHandle; + typedef pthread_mutex_t StreamMutex; + +#else // Setup for "dummy" behavior + + #define __RTAUDIO_DUMMY__ + typedef int ThreadHandle; + typedef int StreamMutex; + +#endif + +// This global structure type is used to pass callback information +// between the private RtAudio stream structure and global callback +// handling functions. +struct CallbackInfo { + void *object; // Used as a "this" pointer. + ThreadHandle thread; + void *callback; + void *userData; + void *apiInfo; // void pointer for API specific callback information + bool isRunning; + + // Default constructor. + CallbackInfo() + :object(0), callback(0), userData(0), apiInfo(0), isRunning(false) {} +}; + +// **************************************************************** // +// +// RtApi class declaration. +// +// Subclasses of RtApi contain all API- and OS-specific code necessary +// to fully implement the RtAudio API. +// +// Note that RtApi is an abstract base class and cannot be +// explicitly instantiated. The class RtAudio will create an +// instance of an RtApi subclass (RtApiOss, RtApiAlsa, +// RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio). +// +// **************************************************************** // + +#if defined( HAVE_GETTIMEOFDAY ) + #include +#endif + +#include + +class RtApi +{ +public: + + RtApi(); + virtual ~RtApi(); + virtual RtAudio::Api getCurrentApi( void ) = 0; + virtual unsigned int getDeviceCount( void ) = 0; + virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; + virtual unsigned int getDefaultInputDevice( void ); + virtual unsigned int getDefaultOutputDevice( void ); + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData, RtAudio::StreamOptions *options ); + virtual void closeStream( void ); + virtual void startStream( void ) = 0; + virtual void stopStream( void ) = 0; + virtual void abortStream( void ) = 0; + long getStreamLatency( void ); + virtual double getStreamTime( void ); + bool isStreamOpen( void ) { return stream_.state != STREAM_CLOSED; }; + bool isStreamRunning( void ) { return stream_.state == STREAM_RUNNING; }; + void showWarnings( bool value ) { showWarnings_ = value; }; + + +protected: + + static const unsigned int MAX_SAMPLE_RATES; + static const unsigned int SAMPLE_RATES[]; + + enum { FAILURE, SUCCESS }; + + enum StreamState { + STREAM_STOPPED, + STREAM_RUNNING, + STREAM_CLOSED = -50 + }; + + enum StreamMode { + OUTPUT, + INPUT, + DUPLEX, + UNINITIALIZED = -75 + }; + + // A protected structure used for buffer conversion. + struct ConvertInfo { + int channels; + int inJump, outJump; + RtAudioFormat inFormat, outFormat; + std::vector inOffset; + std::vector outOffset; + }; + + // A protected structure for audio streams. + struct RtApiStream { + unsigned int device[2]; // Playback and record, respectively. + void *apiHandle; // void pointer for API specific stream handle information + StreamMode mode; // OUTPUT, INPUT, or DUPLEX. + StreamState state; // STOPPED, RUNNING, or CLOSED + char *userBuffer[2]; // Playback and record, respectively. + char *deviceBuffer; + bool doConvertBuffer[2]; // Playback and record, respectively. + bool userInterleaved; + bool deviceInterleaved[2]; // Playback and record, respectively. + bool doByteSwap[2]; // Playback and record, respectively. + unsigned int sampleRate; + unsigned int bufferSize; + unsigned int nBuffers; + unsigned int nUserChannels[2]; // Playback and record, respectively. + unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. + unsigned int channelOffset[2]; // Playback and record, respectively. + unsigned long latency[2]; // Playback and record, respectively. + RtAudioFormat userFormat; + RtAudioFormat deviceFormat[2]; // Playback and record, respectively. + StreamMutex mutex; + CallbackInfo callbackInfo; + ConvertInfo convertInfo[2]; + double streamTime; // Number of elapsed seconds since the stream started. + +#if defined(HAVE_GETTIMEOFDAY) + struct timeval lastTickTimestamp; +#endif + + RtApiStream() + :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } + }; + + typedef signed short Int16; + typedef signed int Int32; + typedef float Float32; + typedef double Float64; + + std::ostringstream errorStream_; + std::string errorText_; + bool showWarnings_; + RtApiStream stream_; + + /*! + Protected, api-specific method that attempts to open a device + with the given parameters. This function MUST be implemented by + all subclasses. If an error is encountered during the probe, a + "warning" message is reported and FAILURE is returned. A + successful probe is indicated by a return value of SUCCESS. + */ + virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); + + //! A protected function used to increment the stream time. + void tickStreamTime( void ); + + //! Protected common method to clear an RtApiStream structure. + void clearStreamInfo(); + + /*! + Protected common method that throws an RtError (type = + INVALID_USE) if a stream is not open. + */ + void verifyStream( void ); + + //! Protected common error method to allow global control over error handling. + void error( RtError::Type type ); + + /*! + Protected method used to perform format, channel number, and/or interleaving + conversions between the user and device buffers. + */ + void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); + + //! Protected common method used to perform byte-swapping on buffers. + void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); + + //! Protected common method that returns the number of bytes for a given format. + unsigned int formatBytes( RtAudioFormat format ); + + //! Protected common method that sets up the parameters for buffer conversion. + void setConvertInfo( StreamMode mode, unsigned int firstChannel ); +}; + +// **************************************************************** // +// +// Inline RtAudio definitions. +// +// **************************************************************** // + +inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); } +inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); } +inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } +inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); } +inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); } +inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); } +inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } +inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } +inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } +inline bool RtAudio :: isStreamOpen( void ) throw() { return rtapi_->isStreamOpen(); } +inline bool RtAudio :: isStreamRunning( void ) throw() { return rtapi_->isStreamRunning(); } +inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } +inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } +inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); } + +// RtApi Subclass prototypes. + +#if defined(__MACOSX_CORE__) + +#include + +class RtApiCore: public RtApi +{ +public: + + RtApiCore(); + ~RtApiCore(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); + static const char* getErrorCode( OSStatus code ); +}; + +#endif + +#if defined(__UNIX_JACK__) + +class RtApiJack: public RtApi +{ +public: + + RtApiJack(); + ~RtApiJack(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( unsigned long nframes ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__WINDOWS_ASIO__) + +class RtApiAsio: public RtApi +{ +public: + + RtApiAsio(); + ~RtApiAsio(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( long bufferIndex ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool coInitialized_; + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__WINDOWS_DS__) + +class RtApiDs: public RtApi +{ +public: + + RtApiDs(); + ~RtApiDs(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }; + unsigned int getDeviceCount( void ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + bool coInitialized_; + bool buffersRolling; + long duplexPrerollBytes; + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__LINUX_ALSA__) + +class RtApiAlsa: public RtApi +{ +public: + + RtApiAlsa(); + ~RtApiAlsa(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__LINUX_OSS__) + +class RtApiOss: public RtApi +{ +public: + + RtApiOss(); + ~RtApiOss(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__RTAUDIO_DUMMY__) + +class RtApiDummy: public RtApi +{ +public: + + RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); }; + RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }; + unsigned int getDeviceCount( void ) { return 0; }; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; }; + void closeStream( void ) {}; + void startStream( void ) {}; + void stopStream( void ) {}; + void abortStream( void ) {}; + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) { return false; }; +}; + +#endif + +#endif + +// Indentation settings for Vim and Emacs +// +// Local Variables: +// c-basic-offset: 2 +// indent-tabs-mode: nil +// End: +// +// vim: et sts=2 sw=2 diff --git a/rtaudio/RtError.h b/rtaudio/RtError.h new file mode 100644 index 0000000..900f4a4 --- /dev/null +++ b/rtaudio/RtError.h @@ -0,0 +1,61 @@ +/************************************************************************/ +/*! \class RtError + \brief Exception handling class for RtAudio & RtMidi. + + The RtError class is quite simple but it does allow errors to be + "caught" by RtError::Type. See the RtAudio and RtMidi + documentation to know which methods can throw an RtError. + +*/ +/************************************************************************/ + +#ifndef RTERROR_H +#define RTERROR_H + +#include +//#include +#include +#include + +class RtError : public std::exception +{ + public: + //! Defined RtError types. + enum Type { + WARNING, /*!< A non-critical error. */ + DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ + UNSPECIFIED, /*!< The default, unspecified error type. */ + NO_DEVICES_FOUND, /*!< No devices found on system. */ + INVALID_DEVICE, /*!< An invalid device ID was specified. */ + MEMORY_ERROR, /*!< An error occured during memory allocation. */ + INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ + INVALID_USE, /*!< The function was called incorrectly. */ + DRIVER_ERROR, /*!< A system driver error occured. */ + SYSTEM_ERROR, /*!< A system error occured. */ + THREAD_ERROR /*!< A thread error occured. */ + }; + + //! The constructor. + RtError( const std::string& message, Type type = RtError::UNSPECIFIED ) throw() : message_(message), type_(type) {} + + //! The destructor. + virtual ~RtError( void ) throw() {} + + //! Prints thrown error message to stderr. + virtual void printMessage( ) throw() { std::fprintf(stderr,"%s\n", message_.c_str()); } + + //! Returns the thrown error message type. + virtual const Type& getType(void) throw() { return type_; } + + //! Returns the thrown error message string. + virtual const std::string& getMessage(void) throw() { return message_; } + + //! Returns the thrown error message as a c-style string. + virtual const char* what( void ) const throw() { return message_.c_str(); } + + protected: + std::string message_; + Type type_; +}; + +#endif diff --git a/rtaudio/include/asio.cpp b/rtaudio/include/asio.cpp new file mode 100644 index 0000000..b241663 --- /dev/null +++ b/rtaudio/include/asio.cpp @@ -0,0 +1,257 @@ +/* + Steinberg Audio Stream I/O API + (c) 1996, Steinberg Soft- und Hardware GmbH + + asio.cpp + + asio functions entries which translate the + asio interface to the asiodrvr class methods +*/ + +#include +#include "asiosys.h" // platform definition +#include "asio.h" + +#if MAC +#include "asiodrvr.h" + +#pragma export on + +AsioDriver *theAsioDriver = 0; + +extern "C" +{ + +long main() +{ + return 'ASIO'; +} + +#elif WINDOWS + +#include "windows.h" +#include "iasiodrv.h" +#include "asiodrivers.h" + +IASIO *theAsioDriver = 0; +extern AsioDrivers *asioDrivers; + +#elif SGI || SUN || BEOS || LINUX +#include "asiodrvr.h" +static AsioDriver *theAsioDriver = 0; +#endif + +//----------------------------------------------------------------------------------------------------- +ASIOError ASIOInit(ASIODriverInfo *info) +{ +#if MAC || SGI || SUN || BEOS || LINUX + if(theAsioDriver) + { + delete theAsioDriver; + theAsioDriver = 0; + } + info->driverVersion = 0; + strcpy(info->name, "No ASIO Driver"); + theAsioDriver = getDriver(); + if(!theAsioDriver) + { + strcpy(info->errorMessage, "Not enough memory for the ASIO driver!"); + return ASE_NotPresent; + } + if(!theAsioDriver->init(info->sysRef)) + { + theAsioDriver->getErrorMessage(info->errorMessage); + delete theAsioDriver; + theAsioDriver = 0; + return ASE_NotPresent; + } + strcpy(info->errorMessage, "No ASIO Driver Error"); + theAsioDriver->getDriverName(info->name); + info->driverVersion = theAsioDriver->getDriverVersion(); + return ASE_OK; + +#else + + info->driverVersion = 0; + strcpy(info->name, "No ASIO Driver"); + if(theAsioDriver) // must be loaded! + { + if(!theAsioDriver->init(info->sysRef)) + { + theAsioDriver->getErrorMessage(info->errorMessage); + theAsioDriver = 0; + return ASE_NotPresent; + } + + strcpy(info->errorMessage, "No ASIO Driver Error"); + theAsioDriver->getDriverName(info->name); + info->driverVersion = theAsioDriver->getDriverVersion(); + return ASE_OK; + } + return ASE_NotPresent; + +#endif // !MAC +} + +ASIOError ASIOExit(void) +{ + if(theAsioDriver) + { +#if WINDOWS + asioDrivers->removeCurrentDriver(); +#else + delete theAsioDriver; +#endif + } + theAsioDriver = 0; + return ASE_OK; +} + +ASIOError ASIOStart(void) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->start(); +} + +ASIOError ASIOStop(void) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->stop(); +} + +ASIOError ASIOGetChannels(long *numInputChannels, long *numOutputChannels) +{ + if(!theAsioDriver) + { + *numInputChannels = *numOutputChannels = 0; + return ASE_NotPresent; + } + return theAsioDriver->getChannels(numInputChannels, numOutputChannels); +} + +ASIOError ASIOGetLatencies(long *inputLatency, long *outputLatency) +{ + if(!theAsioDriver) + { + *inputLatency = *outputLatency = 0; + return ASE_NotPresent; + } + return theAsioDriver->getLatencies(inputLatency, outputLatency); +} + +ASIOError ASIOGetBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity) +{ + if(!theAsioDriver) + { + *minSize = *maxSize = *preferredSize = *granularity = 0; + return ASE_NotPresent; + } + return theAsioDriver->getBufferSize(minSize, maxSize, preferredSize, granularity); +} + +ASIOError ASIOCanSampleRate(ASIOSampleRate sampleRate) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->canSampleRate(sampleRate); +} + +ASIOError ASIOGetSampleRate(ASIOSampleRate *currentRate) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->getSampleRate(currentRate); +} + +ASIOError ASIOSetSampleRate(ASIOSampleRate sampleRate) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->setSampleRate(sampleRate); +} + +ASIOError ASIOGetClockSources(ASIOClockSource *clocks, long *numSources) +{ + if(!theAsioDriver) + { + *numSources = 0; + return ASE_NotPresent; + } + return theAsioDriver->getClockSources(clocks, numSources); +} + +ASIOError ASIOSetClockSource(long reference) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->setClockSource(reference); +} + +ASIOError ASIOGetSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->getSamplePosition(sPos, tStamp); +} + +ASIOError ASIOGetChannelInfo(ASIOChannelInfo *info) +{ + if(!theAsioDriver) + { + info->channelGroup = -1; + info->type = ASIOSTInt16MSB; + strcpy(info->name, "None"); + return ASE_NotPresent; + } + return theAsioDriver->getChannelInfo(info); +} + +ASIOError ASIOCreateBuffers(ASIOBufferInfo *bufferInfos, long numChannels, + long bufferSize, ASIOCallbacks *callbacks) +{ + if(!theAsioDriver) + { + ASIOBufferInfo *info = bufferInfos; + for(long i = 0; i < numChannels; i++, info++) + info->buffers[0] = info->buffers[1] = 0; + return ASE_NotPresent; + } + return theAsioDriver->createBuffers(bufferInfos, numChannels, bufferSize, callbacks); +} + +ASIOError ASIODisposeBuffers(void) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->disposeBuffers(); +} + +ASIOError ASIOControlPanel(void) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->controlPanel(); +} + +ASIOError ASIOFuture(long selector, void *opt) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->future(selector, opt); +} + +ASIOError ASIOOutputReady(void) +{ + if(!theAsioDriver) + return ASE_NotPresent; + return theAsioDriver->outputReady(); +} + +#if MAC +} // extern "C" +#pragma export off +#endif + + diff --git a/rtaudio/include/asio.h b/rtaudio/include/asio.h new file mode 100644 index 0000000..8ec811f --- /dev/null +++ b/rtaudio/include/asio.h @@ -0,0 +1,1054 @@ +//--------------------------------------------------------------------------------------------------- +//--------------------------------------------------------------------------------------------------- + +/* + Steinberg Audio Stream I/O API + (c) 1997 - 2005, Steinberg Media Technologies GmbH + + ASIO Interface Specification v 2.1 + + 2005 - Added support for DSD sample data (in cooperation with Sony) + + + basic concept is an i/o synchronous double-buffer scheme: + + on bufferSwitch(index == 0), host will read/write: + + after ASIOStart(), the + read first input buffer A (index 0) + | will be invalid (empty) + * ------------------------ + |------------------------|-----------------------| + | | | + | Input Buffer A (0) | Input Buffer B (1) | + | | | + |------------------------|-----------------------| + | | | + | Output Buffer A (0) | Output Buffer B (1) | + | | | + |------------------------|-----------------------| + * ------------------------- + | before calling ASIOStart(), + write host will have filled output + buffer B (index 1) already + + *please* take special care of proper statement of input + and output latencies (see ASIOGetLatencies()), these + control sequencer sync accuracy + +*/ + +//--------------------------------------------------------------------------------------------------- +//--------------------------------------------------------------------------------------------------- + +/* + +prototypes summary: + +ASIOError ASIOInit(ASIODriverInfo *info); +ASIOError ASIOExit(void); +ASIOError ASIOStart(void); +ASIOError ASIOStop(void); +ASIOError ASIOGetChannels(long *numInputChannels, long *numOutputChannels); +ASIOError ASIOGetLatencies(long *inputLatency, long *outputLatency); +ASIOError ASIOGetBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity); +ASIOError ASIOCanSampleRate(ASIOSampleRate sampleRate); +ASIOError ASIOGetSampleRate(ASIOSampleRate *currentRate); +ASIOError ASIOSetSampleRate(ASIOSampleRate sampleRate); +ASIOError ASIOGetClockSources(ASIOClockSource *clocks, long *numSources); +ASIOError ASIOSetClockSource(long reference); +ASIOError ASIOGetSamplePosition (ASIOSamples *sPos, ASIOTimeStamp *tStamp); +ASIOError ASIOGetChannelInfo(ASIOChannelInfo *info); +ASIOError ASIOCreateBuffers(ASIOBufferInfo *bufferInfos, long numChannels, + long bufferSize, ASIOCallbacks *callbacks); +ASIOError ASIODisposeBuffers(void); +ASIOError ASIOControlPanel(void); +void *ASIOFuture(long selector, void *params); +ASIOError ASIOOutputReady(void); + +*/ + +//--------------------------------------------------------------------------------------------------- +//--------------------------------------------------------------------------------------------------- + +#ifndef __ASIO_H +#define __ASIO_H + +// force 4 byte alignment +#if defined(_MSC_VER) && !defined(__MWERKS__) +#pragma pack(push,4) +#elif PRAGMA_ALIGN_SUPPORTED +#pragma options align = native +#endif + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// Type definitions +//- - - - - - - - - - - - - - - - - - - - - - - - - + +// number of samples data type is 64 bit integer +#if NATIVE_INT64 + typedef long long int ASIOSamples; +#else + typedef struct ASIOSamples { + unsigned long hi; + unsigned long lo; + } ASIOSamples; +#endif + +// Timestamp data type is 64 bit integer, +// Time format is Nanoseconds. +#if NATIVE_INT64 + typedef long long int ASIOTimeStamp ; +#else + typedef struct ASIOTimeStamp { + unsigned long hi; + unsigned long lo; + } ASIOTimeStamp; +#endif + +// Samplerates are expressed in IEEE 754 64 bit double float, +// native format as host computer +#if IEEE754_64FLOAT + typedef double ASIOSampleRate; +#else + typedef struct ASIOSampleRate { + char ieee[8]; + } ASIOSampleRate; +#endif + +// Boolean values are expressed as long +typedef long ASIOBool; +enum { + ASIOFalse = 0, + ASIOTrue = 1 +}; + +// Sample Types are expressed as long +typedef long ASIOSampleType; +enum { + ASIOSTInt16MSB = 0, + ASIOSTInt24MSB = 1, // used for 20 bits as well + ASIOSTInt32MSB = 2, + ASIOSTFloat32MSB = 3, // IEEE 754 32 bit float + ASIOSTFloat64MSB = 4, // IEEE 754 64 bit double float + + // these are used for 32 bit data buffer, with different alignment of the data inside + // 32 bit PCI bus systems can be more easily used with these + ASIOSTInt32MSB16 = 8, // 32 bit data with 16 bit alignment + ASIOSTInt32MSB18 = 9, // 32 bit data with 18 bit alignment + ASIOSTInt32MSB20 = 10, // 32 bit data with 20 bit alignment + ASIOSTInt32MSB24 = 11, // 32 bit data with 24 bit alignment + + ASIOSTInt16LSB = 16, + ASIOSTInt24LSB = 17, // used for 20 bits as well + ASIOSTInt32LSB = 18, + ASIOSTFloat32LSB = 19, // IEEE 754 32 bit float, as found on Intel x86 architecture + ASIOSTFloat64LSB = 20, // IEEE 754 64 bit double float, as found on Intel x86 architecture + + // these are used for 32 bit data buffer, with different alignment of the data inside + // 32 bit PCI bus systems can more easily used with these + ASIOSTInt32LSB16 = 24, // 32 bit data with 18 bit alignment + ASIOSTInt32LSB18 = 25, // 32 bit data with 18 bit alignment + ASIOSTInt32LSB20 = 26, // 32 bit data with 20 bit alignment + ASIOSTInt32LSB24 = 27, // 32 bit data with 24 bit alignment + + // ASIO DSD format. + ASIOSTDSDInt8LSB1 = 32, // DSD 1 bit data, 8 samples per byte. First sample in Least significant bit. + ASIOSTDSDInt8MSB1 = 33, // DSD 1 bit data, 8 samples per byte. First sample in Most significant bit. + ASIOSTDSDInt8NER8 = 40, // DSD 8 bit data, 1 sample per byte. No Endianness required. + + ASIOSTLastEntry +}; + +/*----------------------------------------------------------------------------- +// DSD operation and buffer layout +// Definition by Steinberg/Sony Oxford. +// +// We have tried to treat DSD as PCM and so keep a consistant structure across +// the ASIO interface. +// +// DSD's sample rate is normally referenced as a multiple of 44.1Khz, so +// the standard sample rate is refered to as 64Fs (or 2.8224Mhz). We looked +// at making a special case for DSD and adding a field to the ASIOFuture that +// would allow the user to select the Over Sampleing Rate (OSR) as a seperate +// entity but decided in the end just to treat it as a simple value of +// 2.8224Mhz and use the standard interface to set it. +// +// The second problem was the "word" size, in PCM the word size is always a +// greater than or equal to 8 bits (a byte). This makes life easy as we can +// then pack the samples into the "natural" size for the machine. +// In DSD the "word" size is 1 bit. This is not a major problem and can easily +// be dealt with if we ensure that we always deal with a multiple of 8 samples. +// +// DSD brings with it another twist to the Endianness religion. How are the +// samples packed into the byte. It would be nice to just say the most significant +// bit is always the first sample, however there would then be a performance hit +// on little endian machines. Looking at how some of the processing goes... +// Little endian machines like the first sample to be in the Least Significant Bit, +// this is because when you write it to memory the data is in the correct format +// to be shifted in and out of the words. +// Big endian machine prefer the first sample to be in the Most Significant Bit, +// again for the same reasion. +// +// And just when things were looking really muddy there is a proposed extension to +// DSD that uses 8 bit word sizes. It does not care what endianness you use. +// +// Switching the driver between DSD and PCM mode +// ASIOFuture allows for extending the ASIO API quite transparently. +// See kAsioSetIoFormat, kAsioGetIoFormat, kAsioCanDoIoFormat +// +//-----------------------------------------------------------------------------*/ + + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// Error codes +//- - - - - - - - - - - - - - - - - - - - - - - - - + +typedef long ASIOError; +enum { + ASE_OK = 0, // This value will be returned whenever the call succeeded + ASE_SUCCESS = 0x3f4847a0, // unique success return value for ASIOFuture calls + ASE_NotPresent = -1000, // hardware input or output is not present or available + ASE_HWMalfunction, // hardware is malfunctioning (can be returned by any ASIO function) + ASE_InvalidParameter, // input parameter invalid + ASE_InvalidMode, // hardware is in a bad mode or used in a bad mode + ASE_SPNotAdvancing, // hardware is not running when sample position is inquired + ASE_NoClock, // sample clock or rate cannot be determined or is not present + ASE_NoMemory // not enough memory for completing the request +}; + +//--------------------------------------------------------------------------------------------------- +//--------------------------------------------------------------------------------------------------- + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// Time Info support +//- - - - - - - - - - - - - - - - - - - - - - - - - + +typedef struct ASIOTimeCode +{ + double speed; // speed relation (fraction of nominal speed) + // optional; set to 0. or 1. if not supported + ASIOSamples timeCodeSamples; // time in samples + unsigned long flags; // some information flags (see below) + char future[64]; +} ASIOTimeCode; + +typedef enum ASIOTimeCodeFlags +{ + kTcValid = 1, + kTcRunning = 1 << 1, + kTcReverse = 1 << 2, + kTcOnspeed = 1 << 3, + kTcStill = 1 << 4, + + kTcSpeedValid = 1 << 8 +} ASIOTimeCodeFlags; + +typedef struct AsioTimeInfo +{ + double speed; // absolute speed (1. = nominal) + ASIOTimeStamp systemTime; // system time related to samplePosition, in nanoseconds + // on mac, must be derived from Microseconds() (not UpTime()!) + // on windows, must be derived from timeGetTime() + ASIOSamples samplePosition; + ASIOSampleRate sampleRate; // current rate + unsigned long flags; // (see below) + char reserved[12]; +} AsioTimeInfo; + +typedef enum AsioTimeInfoFlags +{ + kSystemTimeValid = 1, // must always be valid + kSamplePositionValid = 1 << 1, // must always be valid + kSampleRateValid = 1 << 2, + kSpeedValid = 1 << 3, + + kSampleRateChanged = 1 << 4, + kClockSourceChanged = 1 << 5 +} AsioTimeInfoFlags; + +typedef struct ASIOTime // both input/output +{ + long reserved[4]; // must be 0 + struct AsioTimeInfo timeInfo; // required + struct ASIOTimeCode timeCode; // optional, evaluated if (timeCode.flags & kTcValid) +} ASIOTime; + +/* + +using time info: +it is recommended to use the new method with time info even if the asio +device does not support timecode; continuous calls to ASIOGetSamplePosition +and ASIOGetSampleRate are avoided, and there is a more defined relationship +between callback time and the time info. + +see the example below. +to initiate time info mode, after you have received the callbacks pointer in +ASIOCreateBuffers, you will call the asioMessage callback with kAsioSupportsTimeInfo +as the argument. if this returns 1, host has accepted time info mode. +now host expects the new callback bufferSwitchTimeInfo to be used instead +of the old bufferSwitch method. the ASIOTime structure is assumed to be valid +and accessible until the callback returns. + +using time code: +if the device supports reading time code, it will call host's asioMessage callback +with kAsioSupportsTimeCode as the selector. it may then fill the according +fields and set the kTcValid flag. +host will call the future method with the kAsioEnableTimeCodeRead selector when +it wants to enable or disable tc reading by the device. you should also support +the kAsioCanTimeInfo and kAsioCanTimeCode selectors in ASIOFuture (see example). + +note: +the AsioTimeInfo/ASIOTimeCode pair is supposed to work in both directions. +as a matter of convention, the relationship between the sample +position counter and the time code at buffer switch time is +(ignoring offset between tc and sample pos when tc is running): + +on input: sample 0 -> input buffer sample 0 -> time code 0 +on output: sample 0 -> output buffer sample 0 -> time code 0 + +this means that for 'real' calculations, one has to take into account +the according latencies. + +example: + +ASIOTime asioTime; + +in createBuffers() +{ + memset(&asioTime, 0, sizeof(ASIOTime)); + AsioTimeInfo* ti = &asioTime.timeInfo; + ti->sampleRate = theSampleRate; + ASIOTimeCode* tc = &asioTime.timeCode; + tc->speed = 1.; + timeInfoMode = false; + canTimeCode = false; + if(callbacks->asioMessage(kAsioSupportsTimeInfo, 0, 0, 0) == 1) + { + timeInfoMode = true; +#if kCanTimeCode + if(callbacks->asioMessage(kAsioSupportsTimeCode, 0, 0, 0) == 1) + canTimeCode = true; +#endif + } +} + +void switchBuffers(long doubleBufferIndex, bool processNow) +{ + if(timeInfoMode) + { + AsioTimeInfo* ti = &asioTime.timeInfo; + ti->flags = kSystemTimeValid | kSamplePositionValid | kSampleRateValid; + ti->systemTime = theNanoSeconds; + ti->samplePosition = theSamplePosition; + if(ti->sampleRate != theSampleRate) + ti->flags |= kSampleRateChanged; + ti->sampleRate = theSampleRate; + +#if kCanTimeCode + if(canTimeCode && timeCodeEnabled) + { + ASIOTimeCode* tc = &asioTime.timeCode; + tc->timeCodeSamples = tcSamples; // tc in samples + tc->flags = kTcValid | kTcRunning | kTcOnspeed; // if so... + } + ASIOTime* bb = callbacks->bufferSwitchTimeInfo(&asioTime, doubleBufferIndex, processNow ? ASIOTrue : ASIOFalse); +#else + callbacks->bufferSwitchTimeInfo(&asioTime, doubleBufferIndex, processNow ? ASIOTrue : ASIOFalse); +#endif + } + else + callbacks->bufferSwitch(doubleBufferIndex, ASIOFalse); +} + +ASIOError ASIOFuture(long selector, void *params) +{ + switch(selector) + { + case kAsioEnableTimeCodeRead: + timeCodeEnabled = true; + return ASE_SUCCESS; + case kAsioDisableTimeCodeRead: + timeCodeEnabled = false; + return ASE_SUCCESS; + case kAsioCanTimeInfo: + return ASE_SUCCESS; + #if kCanTimeCode + case kAsioCanTimeCode: + return ASE_SUCCESS; + #endif + } + return ASE_NotPresent; +}; + +*/ + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// application's audio stream handler callbacks +//- - - - - - - - - - - - - - - - - - - - - - - - - + +typedef struct ASIOCallbacks +{ + void (*bufferSwitch) (long doubleBufferIndex, ASIOBool directProcess); + // bufferSwitch indicates that both input and output are to be processed. + // the current buffer half index (0 for A, 1 for B) determines + // - the output buffer that the host should start to fill. the other buffer + // will be passed to output hardware regardless of whether it got filled + // in time or not. + // - the input buffer that is now filled with incoming data. Note that + // because of the synchronicity of i/o, the input always has at + // least one buffer latency in relation to the output. + // directProcess suggests to the host whether it should immedeately + // start processing (directProcess == ASIOTrue), or whether its process + // should be deferred because the call comes from a very low level + // (for instance, a high level priority interrupt), and direct processing + // would cause timing instabilities for the rest of the system. If in doubt, + // directProcess should be set to ASIOFalse. + // Note: bufferSwitch may be called at interrupt time for highest efficiency. + + void (*sampleRateDidChange) (ASIOSampleRate sRate); + // gets called when the AudioStreamIO detects a sample rate change + // If sample rate is unknown, 0 is passed (for instance, clock loss + // when externally synchronized). + + long (*asioMessage) (long selector, long value, void* message, double* opt); + // generic callback for various purposes, see selectors below. + // note this is only present if the asio version is 2 or higher + + ASIOTime* (*bufferSwitchTimeInfo) (ASIOTime* params, long doubleBufferIndex, ASIOBool directProcess); + // new callback with time info. makes ASIOGetSamplePosition() and various + // calls to ASIOGetSampleRate obsolete, + // and allows for timecode sync etc. to be preferred; will be used if + // the driver calls asioMessage with selector kAsioSupportsTimeInfo. +} ASIOCallbacks; + +// asioMessage selectors +enum +{ + kAsioSelectorSupported = 1, // selector in , returns 1L if supported, + // 0 otherwise + kAsioEngineVersion, // returns engine (host) asio implementation version, + // 2 or higher + kAsioResetRequest, // request driver reset. if accepted, this + // will close the driver (ASIO_Exit() ) and + // re-open it again (ASIO_Init() etc). some + // drivers need to reconfigure for instance + // when the sample rate changes, or some basic + // changes have been made in ASIO_ControlPanel(). + // returns 1L; note the request is merely passed + // to the application, there is no way to determine + // if it gets accepted at this time (but it usually + // will be). + kAsioBufferSizeChange, // not yet supported, will currently always return 0L. + // for now, use kAsioResetRequest instead. + // once implemented, the new buffer size is expected + // in , and on success returns 1L + kAsioResyncRequest, // the driver went out of sync, such that + // the timestamp is no longer valid. this + // is a request to re-start the engine and + // slave devices (sequencer). returns 1 for ok, + // 0 if not supported. + kAsioLatenciesChanged, // the drivers latencies have changed. The engine + // will refetch the latencies. + kAsioSupportsTimeInfo, // if host returns true here, it will expect the + // callback bufferSwitchTimeInfo to be called instead + // of bufferSwitch + kAsioSupportsTimeCode, // + kAsioMMCCommand, // unused - value: number of commands, message points to mmc commands + kAsioSupportsInputMonitor, // kAsioSupportsXXX return 1 if host supports this + kAsioSupportsInputGain, // unused and undefined + kAsioSupportsInputMeter, // unused and undefined + kAsioSupportsOutputGain, // unused and undefined + kAsioSupportsOutputMeter, // unused and undefined + kAsioOverload, // driver detected an overload + + kAsioNumMessageSelectors +}; + +//--------------------------------------------------------------------------------------------------- +//--------------------------------------------------------------------------------------------------- + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// (De-)Construction +//- - - - - - - - - - - - - - - - - - - - - - - - - + +typedef struct ASIODriverInfo +{ + long asioVersion; // currently, 2 + long driverVersion; // driver specific + char name[32]; + char errorMessage[124]; + void *sysRef; // on input: system reference + // (Windows: application main window handle, Mac & SGI: 0) +} ASIODriverInfo; + +ASIOError ASIOInit(ASIODriverInfo *info); +/* Purpose: + Initialize the AudioStreamIO. + Parameter: + info: pointer to an ASIODriver structure: + - asioVersion: + - on input, the host version. *** Note *** this is 0 for earlier asio + implementations, and the asioMessage callback is implemeted + only if asioVersion is 2 or greater. sorry but due to a design fault + the driver doesn't have access to the host version in ASIOInit :-( + added selector for host (engine) version in the asioMessage callback + so we're ok from now on. + - on return, asio implementation version. + older versions are 1 + if you support this version (namely, ASIO_outputReady() ) + this should be 2 or higher. also see the note in + ASIO_getTimeStamp() ! + - version: on return, the driver version (format is driver specific) + - name: on return, a null-terminated string containing the driver's name + - error message: on return, should contain a user message describing + the type of error that occured during ASIOInit(), if any. + - sysRef: platform specific + Returns: + If neither input nor output is present ASE_NotPresent + will be returned. + ASE_NoMemory, ASE_HWMalfunction are other possible error conditions +*/ + +ASIOError ASIOExit(void); +/* Purpose: + Terminates the AudioStreamIO. + Parameter: + None. + Returns: + If neither input nor output is present ASE_NotPresent + will be returned. + Notes: this implies ASIOStop() and ASIODisposeBuffers(), + meaning that no host callbacks must be accessed after ASIOExit(). +*/ + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// Start/Stop +//- - - - - - - - - - - - - - - - - - - - - - - - - + +ASIOError ASIOStart(void); +/* Purpose: + Start input and output processing synchronously. + This will + - reset the sample counter to zero + - start the hardware (both input and output) + The first call to the hosts' bufferSwitch(index == 0) then tells + the host to read from input buffer A (index 0), and start + processing to output buffer A while output buffer B (which + has been filled by the host prior to calling ASIOStart()) + is possibly sounding (see also ASIOGetLatencies()) + Parameter: + None. + Returns: + If neither input nor output is present, ASE_NotPresent + will be returned. + If the hardware fails to start, ASE_HWMalfunction will be returned. + Notes: + There is no restriction on the time that ASIOStart() takes + to perform (that is, it is not considered a realtime trigger). +*/ + +ASIOError ASIOStop(void); +/* Purpose: + Stops input and output processing altogether. + Parameter: + None. + Returns: + If neither input nor output is present ASE_NotPresent + will be returned. + Notes: + On return from ASIOStop(), the driver must in no + case call the hosts' bufferSwitch() routine. +*/ + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// Inquiry methods and sample rate +//- - - - - - - - - - - - - - - - - - - - - - - - - + +ASIOError ASIOGetChannels(long *numInputChannels, long *numOutputChannels); +/* Purpose: + Returns number of individual input/output channels. + Parameter: + numInputChannels will hold the number of available input channels + numOutputChannels will hold the number of available output channels + Returns: + If no input/output is present ASE_NotPresent will be returned. + If only inputs, or only outputs are available, the according + other parameter will be zero, and ASE_OK is returned. +*/ + +ASIOError ASIOGetLatencies(long *inputLatency, long *outputLatency); +/* Purpose: + Returns the input and output latencies. This includes + device specific delays, like FIFOs etc. + Parameter: + inputLatency will hold the 'age' of the first sample frame + in the input buffer when the hosts reads it in bufferSwitch() + (this is theoretical, meaning it does not include the overhead + and delay between the actual physical switch, and the time + when bufferSitch() enters). + This will usually be the size of one block in sample frames, plus + device specific latencies. + + outputLatency will specify the time between the buffer switch, + and the time when the next play buffer will start to sound. + The next play buffer is defined as the one the host starts + processing after (or at) bufferSwitch(), indicated by the + index parameter (0 for buffer A, 1 for buffer B). + It will usually be either one block, if the host writes directly + to a dma buffer, or two or more blocks if the buffer is 'latched' by + the driver. As an example, on ASIOStart(), the host will have filled + the play buffer at index 1 already; when it gets the callback (with + the parameter index == 0), this tells it to read from the input + buffer 0, and start to fill the play buffer 0 (assuming that now + play buffer 1 is already sounding). In this case, the output + latency is one block. If the driver decides to copy buffer 1 + at that time, and pass it to the hardware at the next slot (which + is most commonly done, but should be avoided), the output latency + becomes two blocks instead, resulting in a total i/o latency of at least + 3 blocks. As memory access is the main bottleneck in native dsp processing, + and to acheive less latency, it is highly recommended to try to avoid + copying (this is also why the driver is the owner of the buffers). To + summarize, the minimum i/o latency can be acheived if the input buffer + is processed by the host into the output buffer which will physically + start to sound on the next time slice. Also note that the host expects + the bufferSwitch() callback to be accessed for each time slice in order + to retain sync, possibly recursively; if it fails to process a block in + time, it will suspend its operation for some time in order to recover. + Returns: + If no input/output is present ASE_NotPresent will be returned. +*/ + +ASIOError ASIOGetBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity); +/* Purpose: + Returns min, max, and preferred buffer sizes for input/output + Parameter: + minSize will hold the minimum buffer size + maxSize will hold the maxium possible buffer size + preferredSize will hold the preferred buffer size (a size which + best fits performance and hardware requirements) + granularity will hold the granularity at which buffer sizes + may differ. Usually, the buffer size will be a power of 2; + in this case, granularity will hold -1 on return, signalling + possible buffer sizes starting from minSize, increased in + powers of 2 up to maxSize. + Returns: + If no input/output is present ASE_NotPresent will be returned. + Notes: + When minimum and maximum buffer size are equal, + the preferred buffer size has to be the same value as well; granularity + should be 0 in this case. +*/ + +ASIOError ASIOCanSampleRate(ASIOSampleRate sampleRate); +/* Purpose: + Inquires the hardware for the available sample rates. + Parameter: + sampleRate is the rate in question. + Returns: + If the inquired sample rate is not supported, ASE_NoClock will be returned. + If no input/output is present ASE_NotPresent will be returned. +*/ +ASIOError ASIOGetSampleRate(ASIOSampleRate *currentRate); +/* Purpose: + Get the current sample Rate. + Parameter: + currentRate will hold the current sample rate on return. + Returns: + If sample rate is unknown, sampleRate will be 0 and ASE_NoClock will be returned. + If no input/output is present ASE_NotPresent will be returned. + Notes: +*/ + +ASIOError ASIOSetSampleRate(ASIOSampleRate sampleRate); +/* Purpose: + Set the hardware to the requested sample Rate. If sampleRate == 0, + enable external sync. + Parameter: + sampleRate: on input, the requested rate + Returns: + If sampleRate is unknown ASE_NoClock will be returned. + If the current clock is external, and sampleRate is != 0, + ASE_InvalidMode will be returned + If no input/output is present ASE_NotPresent will be returned. + Notes: +*/ + +typedef struct ASIOClockSource +{ + long index; // as used for ASIOSetClockSource() + long associatedChannel; // for instance, S/PDIF or AES/EBU + long associatedGroup; // see channel groups (ASIOGetChannelInfo()) + ASIOBool isCurrentSource; // ASIOTrue if this is the current clock source + char name[32]; // for user selection +} ASIOClockSource; + +ASIOError ASIOGetClockSources(ASIOClockSource *clocks, long *numSources); +/* Purpose: + Get the available external audio clock sources + Parameter: + clocks points to an array of ASIOClockSource structures: + - index: this is used to identify the clock source + when ASIOSetClockSource() is accessed, should be + an index counting from zero + - associatedInputChannel: the first channel of an associated + input group, if any. + - associatedGroup: the group index of that channel. + groups of channels are defined to seperate for + instance analog, S/PDIF, AES/EBU, ADAT connectors etc, + when present simultaniously. Note that associated channel + is enumerated according to numInputs/numOutputs, means it + is independant from a group (see also ASIOGetChannelInfo()) + inputs are associated to a clock if the physical connection + transfers both data and clock (like S/PDIF, AES/EBU, or + ADAT inputs). if there is no input channel associated with + the clock source (like Word Clock, or internal oscillator), both + associatedChannel and associatedGroup should be set to -1. + - isCurrentSource: on exit, ASIOTrue if this is the current clock + source, ASIOFalse else + - name: a null-terminated string for user selection of the available sources. + numSources: + on input: the number of allocated array members + on output: the number of available clock sources, at least + 1 (internal clock generator). + Returns: + If no input/output is present ASE_NotPresent will be returned. + Notes: +*/ + +ASIOError ASIOSetClockSource(long index); +/* Purpose: + Set the audio clock source + Parameter: + index as obtained from an inquiry to ASIOGetClockSources() + Returns: + If no input/output is present ASE_NotPresent will be returned. + If the clock can not be selected because an input channel which + carries the current clock source is active, ASE_InvalidMode + *may* be returned (this depends on the properties of the driver + and/or hardware). + Notes: + Should *not* return ASE_NoClock if there is no clock signal present + at the selected source; this will be inquired via ASIOGetSampleRate(). + It should call the host callback procedure sampleRateHasChanged(), + if the switch causes a sample rate change, or if no external clock + is present at the selected source. +*/ + +ASIOError ASIOGetSamplePosition (ASIOSamples *sPos, ASIOTimeStamp *tStamp); +/* Purpose: + Inquires the sample position/time stamp pair. + Parameter: + sPos will hold the sample position on return. The sample + position is reset to zero when ASIOStart() gets called. + tStamp will hold the system time when the sample position + was latched. + Returns: + If no input/output is present, ASE_NotPresent will be returned. + If there is no clock, ASE_SPNotAdvancing will be returned. + Notes: + + in order to be able to synchronise properly, + the sample position / time stamp pair must refer to the current block, + that is, the engine will call ASIOGetSamplePosition() in its bufferSwitch() + callback and expect the time for the current block. thus, when requested + in the very first bufferSwitch after ASIO_Start(), the sample position + should be zero, and the time stamp should refer to the very time where + the stream was started. it also means that the sample position must be + block aligned. the driver must ensure proper interpolation if the system + time can not be determined for the block position. the driver is responsible + for precise time stamps as it usually has most direct access to lower + level resources. proper behaviour of ASIO_GetSamplePosition() and ASIO_GetLatencies() + are essential for precise media synchronization! +*/ + +typedef struct ASIOChannelInfo +{ + long channel; // on input, channel index + ASIOBool isInput; // on input + ASIOBool isActive; // on exit + long channelGroup; // dto + ASIOSampleType type; // dto + char name[32]; // dto +} ASIOChannelInfo; + +ASIOError ASIOGetChannelInfo(ASIOChannelInfo *info); +/* Purpose: + retreive information about the nature of a channel + Parameter: + info: pointer to a ASIOChannelInfo structure with + - channel: on input, the channel index of the channel in question. + - isInput: on input, ASIOTrue if info for an input channel is + requested, else output + - channelGroup: on return, the channel group that the channel + belongs to. For drivers which support different types of + channels, like analog, S/PDIF, AES/EBU, ADAT etc interfaces, + there should be a reasonable grouping of these types. Groups + are always independant form a channel index, that is, a channel + index always counts from 0 to numInputs/numOutputs regardless + of the group it may belong to. + There will always be at least one group (group 0). Please + also note that by default, the host may decide to activate + channels 0 and 1; thus, these should belong to the most + useful type (analog i/o, if present). + - type: on return, contains the sample type of the channel + - isActive: on return, ASIOTrue if channel is active as it was + installed by ASIOCreateBuffers(), ASIOFalse else + - name: describing the type of channel in question. Used to allow + for user selection, and enabling of specific channels. examples: + "Analog In", "SPDIF Out" etc + Returns: + If no input/output is present ASE_NotPresent will be returned. + Notes: + If possible, the string should be organised such that the first + characters are most significantly describing the nature of the + port, to allow for identification even if the view showing the + port name is too small to display more than 8 characters, for + instance. +*/ + +//- - - - - - - - - - - - - - - - - - - - - - - - - +// Buffer preparation +//- - - - - - - - - - - - - - - - - - - - - - - - - + +typedef struct ASIOBufferInfo +{ + ASIOBool isInput; // on input: ASIOTrue: input, else output + long channelNum; // on input: channel index + void *buffers[2]; // on output: double buffer addresses +} ASIOBufferInfo; + +ASIOError ASIOCreateBuffers(ASIOBufferInfo *bufferInfos, long numChannels, + long bufferSize, ASIOCallbacks *callbacks); + +/* Purpose: + Allocates input/output buffers for all input and output channels to be activated. + Parameter: + bufferInfos is a pointer to an array of ASIOBufferInfo structures: + - isInput: on input, ASIOTrue if the buffer is to be allocated + for an input, output buffer else + - channelNum: on input, the index of the channel in question + (counting from 0) + - buffers: on exit, 2 pointers to the halves of the channels' double-buffer. + the size of the buffer(s) of course depend on both the ASIOSampleType + as obtained from ASIOGetChannelInfo(), and bufferSize + numChannels is the sum of all input and output channels to be created; + thus bufferInfos is a pointer to an array of numChannels ASIOBufferInfo + structures. + bufferSize selects one of the possible buffer sizes as obtained from + ASIOGetBufferSizes(). + callbacks is a pointer to an ASIOCallbacks structure. + Returns: + If not enough memory is available ASE_NoMemory will be returned. + If no input/output is present ASE_NotPresent will be returned. + If bufferSize is not supported, or one or more of the bufferInfos elements + contain invalid settings, ASE_InvalidMode will be returned. + Notes: + If individual channel selection is not possible but requested, + the driver has to handle this. namely, bufferSwitch() will only + have filled buffers of enabled outputs. If possible, processing + and buss activities overhead should be avoided for channels which + were not enabled here. +*/ + +ASIOError ASIODisposeBuffers(void); +/* Purpose: + Releases all buffers for the device. + Parameter: + None. + Returns: + If no buffer were ever prepared, ASE_InvalidMode will be returned. + If no input/output is present ASE_NotPresent will be returned. + Notes: + This implies ASIOStop(). +*/ + +ASIOError ASIOControlPanel(void); +/* Purpose: + request the driver to start a control panel component + for device specific user settings. This will not be + accessed on some platforms (where the component is accessed + instead). + Parameter: + None. + Returns: + If no panel is available ASE_NotPresent will be returned. + Actually, the return code is ignored. + Notes: + if the user applied settings which require a re-configuration + of parts or all of the enigine and/or driver (such as a change of + the block size), the asioMessage callback can be used (see + ASIO_Callbacks). +*/ + +ASIOError ASIOFuture(long selector, void *params); +/* Purpose: + various + Parameter: + selector: operation Code as to be defined. zero is reserved for + testing purposes. + params: depends on the selector; usually pointer to a structure + for passing and retreiving any type and amount of parameters. + Returns: + the return value is also selector dependant. if the selector + is unknown, ASE_InvalidParameter should be returned to prevent + further calls with this selector. on success, ASE_SUCCESS + must be returned (note: ASE_OK is *not* sufficient!) + Notes: + see selectors defined below. +*/ + +enum +{ + kAsioEnableTimeCodeRead = 1, // no arguments + kAsioDisableTimeCodeRead, // no arguments + kAsioSetInputMonitor, // ASIOInputMonitor* in params + kAsioTransport, // ASIOTransportParameters* in params + kAsioSetInputGain, // ASIOChannelControls* in params, apply gain + kAsioGetInputMeter, // ASIOChannelControls* in params, fill meter + kAsioSetOutputGain, // ASIOChannelControls* in params, apply gain + kAsioGetOutputMeter, // ASIOChannelControls* in params, fill meter + kAsioCanInputMonitor, // no arguments for kAsioCanXXX selectors + kAsioCanTimeInfo, + kAsioCanTimeCode, + kAsioCanTransport, + kAsioCanInputGain, + kAsioCanInputMeter, + kAsioCanOutputGain, + kAsioCanOutputMeter, + + // DSD support + // The following extensions are required to allow switching + // and control of the DSD subsystem. + kAsioSetIoFormat = 0x23111961, /* ASIOIoFormat * in params. */ + kAsioGetIoFormat = 0x23111983, /* ASIOIoFormat * in params. */ + kAsioCanDoIoFormat = 0x23112004, /* ASIOIoFormat * in params. */ +}; + +typedef struct ASIOInputMonitor +{ + long input; // this input was set to monitor (or off), -1: all + long output; // suggested output for monitoring the input (if so) + long gain; // suggested gain, ranging 0 - 0x7fffffffL (-inf to +12 dB) + ASIOBool state; // ASIOTrue => on, ASIOFalse => off + long pan; // suggested pan, 0 => all left, 0x7fffffff => right +} ASIOInputMonitor; + +typedef struct ASIOChannelControls +{ + long channel; // on input, channel index + ASIOBool isInput; // on input + long gain; // on input, ranges 0 thru 0x7fffffff + long meter; // on return, ranges 0 thru 0x7fffffff + char future[32]; +} ASIOChannelControls; + +typedef struct ASIOTransportParameters +{ + long command; // see enum below + ASIOSamples samplePosition; + long track; + long trackSwitches[16]; // 512 tracks on/off + char future[64]; +} ASIOTransportParameters; + +enum +{ + kTransStart = 1, + kTransStop, + kTransLocate, // to samplePosition + kTransPunchIn, + kTransPunchOut, + kTransArmOn, // track + kTransArmOff, // track + kTransMonitorOn, // track + kTransMonitorOff, // track + kTransArm, // trackSwitches + kTransMonitor // trackSwitches +}; + +/* +// DSD support +// Some notes on how to use ASIOIoFormatType. +// +// The caller will fill the format with the request types. +// If the board can do the request then it will leave the +// values unchanged. If the board does not support the +// request then it will change that entry to Invalid (-1) +// +// So to request DSD then +// +// ASIOIoFormat NeedThis={kASIODSDFormat}; +// +// if(ASE_SUCCESS != ASIOFuture(kAsioSetIoFormat,&NeedThis) ){ +// // If the board did not accept one of the parameters then the +// // whole call will fail and the failing parameter will +// // have had its value changes to -1. +// } +// +// Note: Switching between the formats need to be done before the "prepared" +// state (see ASIO 2 documentation) is entered. +*/ +typedef long int ASIOIoFormatType; +enum ASIOIoFormatType_e +{ + kASIOFormatInvalid = -1, + kASIOPCMFormat = 0, + kASIODSDFormat = 1, +}; + +typedef struct ASIOIoFormat_s +{ + ASIOIoFormatType FormatType; + char future[512-sizeof(ASIOIoFormatType)]; +} ASIOIoFormat; + + +ASIOError ASIOOutputReady(void); +/* Purpose: + this tells the driver that the host has completed processing + the output buffers. if the data format required by the hardware + differs from the supported asio formats, but the hardware + buffers are DMA buffers, the driver will have to convert + the audio stream data; as the bufferSwitch callback is + usually issued at dma block switch time, the driver will + have to convert the *previous* host buffer, which increases + the output latency by one block. + when the host finds out that ASIOOutputReady() returns + true, it will issue this call whenever it completed + output processing. then the driver can convert the + host data directly to the dma buffer to be played next, + reducing output latency by one block. + another way to look at it is, that the buffer switch is called + in order to pass the *input* stream to the host, so that it can + process the input into the output, and the output stream is passed + to the driver when the host has completed its process. + Parameter: + None + Returns: + only if the above mentioned scenario is given, and a reduction + of output latency can be acheived by this mechanism, should + ASE_OK be returned. otherwise (and usually), ASE_NotPresent + should be returned in order to prevent further calls to this + function. note that the host may want to determine if it is + to use this when the system is not yet fully initialized, so + ASE_OK should always be returned if the mechanism makes sense. + Notes: + please remeber to adjust ASIOGetLatencies() according to + whether ASIOOutputReady() was ever called or not, if your + driver supports this scenario. + also note that the engine may fail to call ASIO_OutputReady() + in time in overload cases. as already mentioned, bufferSwitch + should be called for every block regardless of whether a block + could be processed in time. +*/ + +// restore old alignment +#if defined(_MSC_VER) && !defined(__MWERKS__) +#pragma pack(pop) +#elif PRAGMA_ALIGN_SUPPORTED +#pragma options align = reset +#endif + +#endif + diff --git a/rtaudio/include/asiodrivers.cpp b/rtaudio/include/asiodrivers.cpp new file mode 100644 index 0000000..5f56454 --- /dev/null +++ b/rtaudio/include/asiodrivers.cpp @@ -0,0 +1,186 @@ +#include +#include "asiodrivers.h" + +AsioDrivers* asioDrivers = 0; + +bool loadAsioDriver(char *name); + +bool loadAsioDriver(char *name) +{ + if(!asioDrivers) + asioDrivers = new AsioDrivers(); + if(asioDrivers) + return asioDrivers->loadDriver(name); + return false; +} + +//------------------------------------------------------------------------------------ + +#if MAC + +bool resolveASIO(unsigned long aconnID); + +AsioDrivers::AsioDrivers() : CodeFragments("ASIO Drivers", 'AsDr', 'Asio') +{ + connID = -1; + curIndex = -1; +} + +AsioDrivers::~AsioDrivers() +{ + removeCurrentDriver(); +} + +bool AsioDrivers::getCurrentDriverName(char *name) +{ + if(curIndex >= 0) + return getName(curIndex, name); + return false; +} + +long AsioDrivers::getDriverNames(char **names, long maxDrivers) +{ + for(long i = 0; i < getNumFragments() && i < maxDrivers; i++) + getName(i, names[i]); + return getNumFragments() < maxDrivers ? getNumFragments() : maxDrivers; +} + +bool AsioDrivers::loadDriver(char *name) +{ + char dname[64]; + unsigned long newID; + + for(long i = 0; i < getNumFragments(); i++) + { + if(getName(i, dname) && !strcmp(name, dname)) + { + if(newInstance(i, &newID)) + { + if(resolveASIO(newID)) + { + if(connID != -1) + removeInstance(curIndex, connID); + curIndex = i; + connID = newID; + return true; + } + } + break; + } + } + return false; +} + +void AsioDrivers::removeCurrentDriver() +{ + if(connID != -1) + removeInstance(curIndex, connID); + connID = -1; + curIndex = -1; +} + +//------------------------------------------------------------------------------------ + +#elif WINDOWS + +#include "iasiodrv.h" + +extern IASIO* theAsioDriver; + +AsioDrivers::AsioDrivers() : AsioDriverList() +{ + curIndex = -1; +} + +AsioDrivers::~AsioDrivers() +{ +} + +bool AsioDrivers::getCurrentDriverName(char *name) +{ + if(curIndex >= 0) + return asioGetDriverName(curIndex, name, 32) == 0 ? true : false; + name[0] = 0; + return false; +} + +long AsioDrivers::getDriverNames(char **names, long maxDrivers) +{ + for(long i = 0; i < asioGetNumDev() && i < maxDrivers; i++) + asioGetDriverName(i, names[i], 32); + return asioGetNumDev() < maxDrivers ? asioGetNumDev() : maxDrivers; +} + +bool AsioDrivers::loadDriver(char *name) +{ + char dname[64]; + char curName[64]; + + for(long i = 0; i < asioGetNumDev(); i++) + { + if(!asioGetDriverName(i, dname, 32) && !strcmp(name, dname)) + { + curName[0] = 0; + getCurrentDriverName(curName); // in case we fail... + removeCurrentDriver(); + + if(!asioOpenDriver(i, (void **)&theAsioDriver)) + { + curIndex = i; + return true; + } + else + { + theAsioDriver = 0; + if(curName[0] && strcmp(dname, curName)) + loadDriver(curName); // try restore + } + break; + } + } + return false; +} + +void AsioDrivers::removeCurrentDriver() +{ + if(curIndex != -1) + asioCloseDriver(curIndex); + curIndex = -1; +} + +#elif SGI || BEOS + +#include "asiolist.h" + +AsioDrivers::AsioDrivers() + : AsioDriverList() +{ + curIndex = -1; +} + +AsioDrivers::~AsioDrivers() +{ +} + +bool AsioDrivers::getCurrentDriverName(char *name) +{ + return false; +} + +long AsioDrivers::getDriverNames(char **names, long maxDrivers) +{ + return 0; +} + +bool AsioDrivers::loadDriver(char *name) +{ + return false; +} + +void AsioDrivers::removeCurrentDriver() +{ +} + +#else +#error implement me +#endif diff --git a/rtaudio/include/asiodrivers.h b/rtaudio/include/asiodrivers.h new file mode 100644 index 0000000..2ddf7ad --- /dev/null +++ b/rtaudio/include/asiodrivers.h @@ -0,0 +1,41 @@ +#ifndef __AsioDrivers__ +#define __AsioDrivers__ + +#include "ginclude.h" + +#if MAC +#include "CodeFragments.hpp" + +class AsioDrivers : public CodeFragments + +#elif WINDOWS +#include +#include "asiolist.h" + +class AsioDrivers : public AsioDriverList + +#elif SGI || BEOS +#include "asiolist.h" + +class AsioDrivers : public AsioDriverList + +#else +#error implement me +#endif + +{ +public: + AsioDrivers(); + ~AsioDrivers(); + + bool getCurrentDriverName(char *name); + long getDriverNames(char **names, long maxDrivers); + bool loadDriver(char *name); + void removeCurrentDriver(); + long getCurrentDriverIndex() {return curIndex;} +protected: + unsigned long connID; + long curIndex; +}; + +#endif diff --git a/rtaudio/include/asiodrvr.h b/rtaudio/include/asiodrvr.h new file mode 100644 index 0000000..663f75a --- /dev/null +++ b/rtaudio/include/asiodrvr.h @@ -0,0 +1,76 @@ +/* + Steinberg Audio Stream I/O API + (c) 1996, Steinberg Soft- und Hardware GmbH + charlie (May 1996) + + asiodrvr.h + c++ superclass to implement asio functionality. from this, + you can derive whatever required +*/ + +#ifndef _asiodrvr_ +#define _asiodrvr_ + +// cpu and os system we are running on +#include "asiosys.h" +// basic "C" interface +#include "asio.h" + +class AsioDriver; +extern AsioDriver *getDriver(); // for generic constructor + +#if WINDOWS +#include +#include "combase.h" +#include "iasiodrv.h" +class AsioDriver : public IASIO ,public CUnknown +{ +public: + AsioDriver(LPUNKNOWN pUnk, HRESULT *phr); + + DECLARE_IUNKNOWN + // Factory method + static CUnknown *CreateInstance(LPUNKNOWN pUnk, HRESULT *phr); + // IUnknown + virtual HRESULT STDMETHODCALLTYPE NonDelegatingQueryInterface(REFIID riid,void **ppvObject); + +#else + +class AsioDriver +{ +public: + AsioDriver(); +#endif + virtual ~AsioDriver(); + + virtual ASIOBool init(void* sysRef); + virtual void getDriverName(char *name); // max 32 bytes incl. terminating zero + virtual long getDriverVersion(); + virtual void getErrorMessage(char *string); // max 124 bytes incl. + + virtual ASIOError start(); + virtual ASIOError stop(); + + virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels); + virtual ASIOError getLatencies(long *inputLatency, long *outputLatency); + virtual ASIOError getBufferSize(long *minSize, long *maxSize, + long *preferredSize, long *granularity); + + virtual ASIOError canSampleRate(ASIOSampleRate sampleRate); + virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate); + virtual ASIOError setSampleRate(ASIOSampleRate sampleRate); + virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources); + virtual ASIOError setClockSource(long reference); + + virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp); + virtual ASIOError getChannelInfo(ASIOChannelInfo *info); + + virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels, + long bufferSize, ASIOCallbacks *callbacks); + virtual ASIOError disposeBuffers(); + + virtual ASIOError controlPanel(); + virtual ASIOError future(long selector, void *opt); + virtual ASIOError outputReady(); +}; +#endif diff --git a/rtaudio/include/asiolist.cpp b/rtaudio/include/asiolist.cpp new file mode 100644 index 0000000..5a62f5b --- /dev/null +++ b/rtaudio/include/asiolist.cpp @@ -0,0 +1,268 @@ +#include +#include "iasiodrv.h" +#include "asiolist.h" + +#define ASIODRV_DESC "description" +#define INPROC_SERVER "InprocServer32" +#define ASIO_PATH "software\\asio" +#define COM_CLSID "clsid" + +// ****************************************************************** +// Local Functions +// ****************************************************************** +static LONG findDrvPath (char *clsidstr,char *dllpath,int dllpathsize) +{ + HKEY hkEnum,hksub,hkpath; + char databuf[512]; + LONG cr,rc = -1; + DWORD datatype,datasize; + DWORD index; + OFSTRUCT ofs; + HFILE hfile; + BOOL found = FALSE; + + CharLowerBuff(clsidstr,strlen(clsidstr)); + if ((cr = RegOpenKey(HKEY_CLASSES_ROOT,COM_CLSID,&hkEnum)) == ERROR_SUCCESS) { + + index = 0; + while (cr == ERROR_SUCCESS && !found) { + cr = RegEnumKey(hkEnum,index++,(LPTSTR)databuf,512); + if (cr == ERROR_SUCCESS) { + CharLowerBuff(databuf,strlen(databuf)); + if (!(strcmp(databuf,clsidstr))) { + if ((cr = RegOpenKeyEx(hkEnum,(LPCTSTR)databuf,0,KEY_READ,&hksub)) == ERROR_SUCCESS) { + if ((cr = RegOpenKeyEx(hksub,(LPCTSTR)INPROC_SERVER,0,KEY_READ,&hkpath)) == ERROR_SUCCESS) { + datatype = REG_SZ; datasize = (DWORD)dllpathsize; + cr = RegQueryValueEx(hkpath,0,0,&datatype,(LPBYTE)dllpath,&datasize); + if (cr == ERROR_SUCCESS) { + memset(&ofs,0,sizeof(OFSTRUCT)); + ofs.cBytes = sizeof(OFSTRUCT); + hfile = OpenFile(dllpath,&ofs,OF_EXIST); + if (hfile) rc = 0; + } + RegCloseKey(hkpath); + } + RegCloseKey(hksub); + } + found = TRUE; // break out + } + } + } + RegCloseKey(hkEnum); + } + return rc; +} + + +static LPASIODRVSTRUCT newDrvStruct (HKEY hkey,char *keyname,int drvID,LPASIODRVSTRUCT lpdrv) +{ + HKEY hksub; + char databuf[256]; + char dllpath[MAXPATHLEN]; + WORD wData[100]; + CLSID clsid; + DWORD datatype,datasize; + LONG cr,rc; + + if (!lpdrv) { + if ((cr = RegOpenKeyEx(hkey,(LPCTSTR)keyname,0,KEY_READ,&hksub)) == ERROR_SUCCESS) { + + datatype = REG_SZ; datasize = 256; + cr = RegQueryValueEx(hksub,COM_CLSID,0,&datatype,(LPBYTE)databuf,&datasize); + if (cr == ERROR_SUCCESS) { + rc = findDrvPath (databuf,dllpath,MAXPATHLEN); + if (rc == 0) { + lpdrv = new ASIODRVSTRUCT[1]; + if (lpdrv) { + memset(lpdrv,0,sizeof(ASIODRVSTRUCT)); + lpdrv->drvID = drvID; + MultiByteToWideChar(CP_ACP,0,(LPCSTR)databuf,-1,(LPWSTR)wData,100); + if ((cr = CLSIDFromString((LPOLESTR)wData,(LPCLSID)&clsid)) == S_OK) { + memcpy(&lpdrv->clsid,&clsid,sizeof(CLSID)); + } + + datatype = REG_SZ; datasize = 256; + cr = RegQueryValueEx(hksub,ASIODRV_DESC,0,&datatype,(LPBYTE)databuf,&datasize); + if (cr == ERROR_SUCCESS) { + strcpy(lpdrv->drvname,databuf); + } + else strcpy(lpdrv->drvname,keyname); + } + } + } + RegCloseKey(hksub); + } + } + else lpdrv->next = newDrvStruct(hkey,keyname,drvID+1,lpdrv->next); + + return lpdrv; +} + +static void deleteDrvStruct (LPASIODRVSTRUCT lpdrv) +{ + IASIO *iasio; + + if (lpdrv != 0) { + deleteDrvStruct(lpdrv->next); + if (lpdrv->asiodrv) { + iasio = (IASIO *)lpdrv->asiodrv; + iasio->Release(); + } + delete lpdrv; + } +} + + +static LPASIODRVSTRUCT getDrvStruct (int drvID,LPASIODRVSTRUCT lpdrv) +{ + while (lpdrv) { + if (lpdrv->drvID == drvID) return lpdrv; + lpdrv = lpdrv->next; + } + return 0; +} +// ****************************************************************** + + +// ****************************************************************** +// AsioDriverList +// ****************************************************************** +AsioDriverList::AsioDriverList () +{ + HKEY hkEnum = 0; + char keyname[MAXDRVNAMELEN]; + LPASIODRVSTRUCT pdl; + LONG cr; + DWORD index = 0; + BOOL fin = FALSE; + + numdrv = 0; + lpdrvlist = 0; + + cr = RegOpenKey(HKEY_LOCAL_MACHINE,ASIO_PATH,&hkEnum); + while (cr == ERROR_SUCCESS) { + if ((cr = RegEnumKey(hkEnum,index++,(LPTSTR)keyname,MAXDRVNAMELEN))== ERROR_SUCCESS) { + lpdrvlist = newDrvStruct (hkEnum,keyname,0,lpdrvlist); + } + else fin = TRUE; + } + if (hkEnum) RegCloseKey(hkEnum); + + pdl = lpdrvlist; + while (pdl) { + numdrv++; + pdl = pdl->next; + } + + if (numdrv) CoInitialize(0); // initialize COM +} + +AsioDriverList::~AsioDriverList () +{ + if (numdrv) { + deleteDrvStruct(lpdrvlist); + CoUninitialize(); + } +} + + +LONG AsioDriverList::asioGetNumDev (VOID) +{ + return (LONG)numdrv; +} + + +LONG AsioDriverList::asioOpenDriver (int drvID,LPVOID *asiodrv) +{ + LPASIODRVSTRUCT lpdrv = 0; + long rc; + + if (!asiodrv) return DRVERR_INVALID_PARAM; + + if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) { + if (!lpdrv->asiodrv) { + rc = CoCreateInstance(lpdrv->clsid,0,CLSCTX_INPROC_SERVER,lpdrv->clsid,asiodrv); + if (rc == S_OK) { + lpdrv->asiodrv = *asiodrv; + return 0; + } + // else if (rc == REGDB_E_CLASSNOTREG) + // strcpy (info->messageText, "Driver not registered in the Registration Database!"); + } + else rc = DRVERR_DEVICE_ALREADY_OPEN; + } + else rc = DRVERR_DEVICE_NOT_FOUND; + + return rc; +} + + +LONG AsioDriverList::asioCloseDriver (int drvID) +{ + LPASIODRVSTRUCT lpdrv = 0; + IASIO *iasio; + + if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) { + if (lpdrv->asiodrv) { + iasio = (IASIO *)lpdrv->asiodrv; + iasio->Release(); + lpdrv->asiodrv = 0; + } + } + + return 0; +} + +LONG AsioDriverList::asioGetDriverName (int drvID,char *drvname,int drvnamesize) +{ + LPASIODRVSTRUCT lpdrv = 0; + + if (!drvname) return DRVERR_INVALID_PARAM; + + if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) { + if (strlen(lpdrv->drvname) < (unsigned int)drvnamesize) { + strcpy(drvname,lpdrv->drvname); + } + else { + memcpy(drvname,lpdrv->drvname,drvnamesize-4); + drvname[drvnamesize-4] = '.'; + drvname[drvnamesize-3] = '.'; + drvname[drvnamesize-2] = '.'; + drvname[drvnamesize-1] = 0; + } + return 0; + } + return DRVERR_DEVICE_NOT_FOUND; +} + +LONG AsioDriverList::asioGetDriverPath (int drvID,char *dllpath,int dllpathsize) +{ + LPASIODRVSTRUCT lpdrv = 0; + + if (!dllpath) return DRVERR_INVALID_PARAM; + + if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) { + if (strlen(lpdrv->dllpath) < (unsigned int)dllpathsize) { + strcpy(dllpath,lpdrv->dllpath); + return 0; + } + dllpath[0] = 0; + return DRVERR_INVALID_PARAM; + } + return DRVERR_DEVICE_NOT_FOUND; +} + +LONG AsioDriverList::asioGetDriverCLSID (int drvID,CLSID *clsid) +{ + LPASIODRVSTRUCT lpdrv = 0; + + if (!clsid) return DRVERR_INVALID_PARAM; + + if ((lpdrv = getDrvStruct(drvID,lpdrvlist)) != 0) { + memcpy(clsid,&lpdrv->clsid,sizeof(CLSID)); + return 0; + } + return DRVERR_DEVICE_NOT_FOUND; +} + + diff --git a/rtaudio/include/asiolist.h b/rtaudio/include/asiolist.h new file mode 100644 index 0000000..01c64f0 --- /dev/null +++ b/rtaudio/include/asiolist.h @@ -0,0 +1,46 @@ +#ifndef __asiolist__ +#define __asiolist__ + +#define DRVERR -5000 +#define DRVERR_INVALID_PARAM DRVERR-1 +#define DRVERR_DEVICE_ALREADY_OPEN DRVERR-2 +#define DRVERR_DEVICE_NOT_FOUND DRVERR-3 + +#define MAXPATHLEN 512 +#define MAXDRVNAMELEN 128 + +struct asiodrvstruct +{ + int drvID; + CLSID clsid; + char dllpath[MAXPATHLEN]; + char drvname[MAXDRVNAMELEN]; + LPVOID asiodrv; + struct asiodrvstruct *next; +}; + +typedef struct asiodrvstruct ASIODRVSTRUCT; +typedef ASIODRVSTRUCT *LPASIODRVSTRUCT; + +class AsioDriverList { +public: + AsioDriverList(); + ~AsioDriverList(); + + LONG asioOpenDriver (int,VOID **); + LONG asioCloseDriver (int); + + // nice to have + LONG asioGetNumDev (VOID); + LONG asioGetDriverName (int,char *,int); + LONG asioGetDriverPath (int,char *,int); + LONG asioGetDriverCLSID (int,CLSID *); + + // or use directly access + LPASIODRVSTRUCT lpdrvlist; + int numdrv; +}; + +typedef class AsioDriverList *LPASIODRIVERLIST; + +#endif diff --git a/rtaudio/include/asiosys.h b/rtaudio/include/asiosys.h new file mode 100644 index 0000000..37f7a48 --- /dev/null +++ b/rtaudio/include/asiosys.h @@ -0,0 +1,82 @@ +#ifndef __asiosys__ + #define __asiosys__ + + #ifdef WIN32 + #undef MAC + #define PPC 0 + #define WINDOWS 1 + #define SGI 0 + #define SUN 0 + #define LINUX 0 + #define BEOS 0 + + #define NATIVE_INT64 0 + #define IEEE754_64FLOAT 1 + + #elif BEOS + #define MAC 0 + #define PPC 0 + #define WINDOWS 0 + #define PC 0 + #define SGI 0 + #define SUN 0 + #define LINUX 0 + + #define NATIVE_INT64 0 + #define IEEE754_64FLOAT 1 + + #ifndef DEBUG + #define DEBUG 0 + #if DEBUG + void DEBUGGERMESSAGE(char *string); + #else + #define DEBUGGERMESSAGE(a) + #endif + #endif + + #elif SGI + #define MAC 0 + #define PPC 0 + #define WINDOWS 0 + #define PC 0 + #define SUN 0 + #define LINUX 0 + #define BEOS 0 + + #define NATIVE_INT64 0 + #define IEEE754_64FLOAT 1 + + #ifndef DEBUG + #define DEBUG 0 + #if DEBUG + void DEBUGGERMESSAGE(char *string); + #else + #define DEBUGGERMESSAGE(a) + #endif + #endif + + #else // MAC + + #define MAC 1 + #define PPC 1 + #define WINDOWS 0 + #define PC 0 + #define SGI 0 + #define SUN 0 + #define LINUX 0 + #define BEOS 0 + + #define NATIVE_INT64 0 + #define IEEE754_64FLOAT 1 + + #ifndef DEBUG + #define DEBUG 0 + #if DEBUG + void DEBUGGERMESSAGE(char *string); + #else + #define DEBUGGERMESSAGE(a) + #endif + #endif + #endif + +#endif diff --git a/rtaudio/include/dsound.h b/rtaudio/include/dsound.h new file mode 100644 index 0000000..cb19cca --- /dev/null +++ b/rtaudio/include/dsound.h @@ -0,0 +1,2369 @@ +/*==========================================================================; + * + * Copyright (c) Microsoft Corporation. All rights reserved. + * + * File: dsound.h + * Content: DirectSound include file + * + **************************************************************************/ + +#define COM_NO_WINDOWS_H +#include +#include + +#ifndef DIRECTSOUND_VERSION +#define DIRECTSOUND_VERSION 0x0900 /* Version 9.0 */ +#endif + +#ifdef __cplusplus +extern "C" { +#endif // __cplusplus + +#ifndef __DSOUND_INCLUDED__ +#define __DSOUND_INCLUDED__ + +/* Type definitions shared with Direct3D */ + +#ifndef DX_SHARED_DEFINES + +typedef float D3DVALUE, *LPD3DVALUE; + +#ifndef D3DCOLOR_DEFINED +typedef DWORD D3DCOLOR; +#define D3DCOLOR_DEFINED +#endif + +#ifndef LPD3DCOLOR_DEFINED +typedef DWORD *LPD3DCOLOR; +#define LPD3DCOLOR_DEFINED +#endif + +#ifndef D3DVECTOR_DEFINED +typedef struct _D3DVECTOR { + float x; + float y; + float z; +} D3DVECTOR; +#define D3DVECTOR_DEFINED +#endif + +#ifndef LPD3DVECTOR_DEFINED +typedef D3DVECTOR *LPD3DVECTOR; +#define LPD3DVECTOR_DEFINED +#endif + +#define DX_SHARED_DEFINES +#endif // DX_SHARED_DEFINES + +#define _FACDS 0x878 /* DirectSound's facility code */ +#define MAKE_DSHRESULT(code) MAKE_HRESULT(1, _FACDS, code) + +// DirectSound Component GUID {47D4D946-62E8-11CF-93BC-444553540000} +DEFINE_GUID(CLSID_DirectSound, 0x47d4d946, 0x62e8, 0x11cf, 0x93, 0xbc, 0x44, 0x45, 0x53, 0x54, 0x0, 0x0); + +// DirectSound 8.0 Component GUID {3901CC3F-84B5-4FA4-BA35-AA8172B8A09B} +DEFINE_GUID(CLSID_DirectSound8, 0x3901cc3f, 0x84b5, 0x4fa4, 0xba, 0x35, 0xaa, 0x81, 0x72, 0xb8, 0xa0, 0x9b); + +// DirectSound Capture Component GUID {B0210780-89CD-11D0-AF08-00A0C925CD16} +DEFINE_GUID(CLSID_DirectSoundCapture, 0xb0210780, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16); + +// DirectSound 8.0 Capture Component GUID {E4BCAC13-7F99-4908-9A8E-74E3BF24B6E1} +DEFINE_GUID(CLSID_DirectSoundCapture8, 0xe4bcac13, 0x7f99, 0x4908, 0x9a, 0x8e, 0x74, 0xe3, 0xbf, 0x24, 0xb6, 0xe1); + +// DirectSound Full Duplex Component GUID {FEA4300C-7959-4147-B26A-2377B9E7A91D} +DEFINE_GUID(CLSID_DirectSoundFullDuplex, 0xfea4300c, 0x7959, 0x4147, 0xb2, 0x6a, 0x23, 0x77, 0xb9, 0xe7, 0xa9, 0x1d); + + +// DirectSound default playback device GUID {DEF00000-9C6D-47ED-AAF1-4DDA8F2B5C03} +DEFINE_GUID(DSDEVID_DefaultPlayback, 0xdef00000, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03); + +// DirectSound default capture device GUID {DEF00001-9C6D-47ED-AAF1-4DDA8F2B5C03} +DEFINE_GUID(DSDEVID_DefaultCapture, 0xdef00001, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03); + +// DirectSound default device for voice playback {DEF00002-9C6D-47ED-AAF1-4DDA8F2B5C03} +DEFINE_GUID(DSDEVID_DefaultVoicePlayback, 0xdef00002, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03); + +// DirectSound default device for voice capture {DEF00003-9C6D-47ED-AAF1-4DDA8F2B5C03} +DEFINE_GUID(DSDEVID_DefaultVoiceCapture, 0xdef00003, 0x9c6d, 0x47ed, 0xaa, 0xf1, 0x4d, 0xda, 0x8f, 0x2b, 0x5c, 0x03); + + +// +// Forward declarations for interfaces. +// 'struct' not 'class' per the way DECLARE_INTERFACE_ is defined +// + +#ifdef __cplusplus +struct IDirectSound; +struct IDirectSoundBuffer; +struct IDirectSound3DListener; +struct IDirectSound3DBuffer; +struct IDirectSoundCapture; +struct IDirectSoundCaptureBuffer; +struct IDirectSoundNotify; +#endif // __cplusplus + + +// +// DirectSound 8.0 interfaces. +// + +#if DIRECTSOUND_VERSION >= 0x0800 + +#ifdef __cplusplus +struct IDirectSound8; +struct IDirectSoundBuffer8; +struct IDirectSoundCaptureBuffer8; +struct IDirectSoundFXGargle; +struct IDirectSoundFXChorus; +struct IDirectSoundFXFlanger; +struct IDirectSoundFXEcho; +struct IDirectSoundFXDistortion; +struct IDirectSoundFXCompressor; +struct IDirectSoundFXParamEq; +struct IDirectSoundFXWavesReverb; +struct IDirectSoundFXI3DL2Reverb; +struct IDirectSoundCaptureFXAec; +struct IDirectSoundCaptureFXNoiseSuppress; +struct IDirectSoundFullDuplex; +#endif // __cplusplus + +// IDirectSound8, IDirectSoundBuffer8 and IDirectSoundCaptureBuffer8 are the +// only DirectSound 7.0 interfaces with changed functionality in version 8.0. +// The other level 8 interfaces as equivalent to their level 7 counterparts: + +#define IDirectSoundCapture8 IDirectSoundCapture +#define IDirectSound3DListener8 IDirectSound3DListener +#define IDirectSound3DBuffer8 IDirectSound3DBuffer +#define IDirectSoundNotify8 IDirectSoundNotify +#define IDirectSoundFXGargle8 IDirectSoundFXGargle +#define IDirectSoundFXChorus8 IDirectSoundFXChorus +#define IDirectSoundFXFlanger8 IDirectSoundFXFlanger +#define IDirectSoundFXEcho8 IDirectSoundFXEcho +#define IDirectSoundFXDistortion8 IDirectSoundFXDistortion +#define IDirectSoundFXCompressor8 IDirectSoundFXCompressor +#define IDirectSoundFXParamEq8 IDirectSoundFXParamEq +#define IDirectSoundFXWavesReverb8 IDirectSoundFXWavesReverb +#define IDirectSoundFXI3DL2Reverb8 IDirectSoundFXI3DL2Reverb +#define IDirectSoundCaptureFXAec8 IDirectSoundCaptureFXAec +#define IDirectSoundCaptureFXNoiseSuppress8 IDirectSoundCaptureFXNoiseSuppress +#define IDirectSoundFullDuplex8 IDirectSoundFullDuplex + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +typedef struct IDirectSound *LPDIRECTSOUND; +typedef struct IDirectSoundBuffer *LPDIRECTSOUNDBUFFER; +typedef struct IDirectSound3DListener *LPDIRECTSOUND3DLISTENER; +typedef struct IDirectSound3DBuffer *LPDIRECTSOUND3DBUFFER; +typedef struct IDirectSoundCapture *LPDIRECTSOUNDCAPTURE; +typedef struct IDirectSoundCaptureBuffer *LPDIRECTSOUNDCAPTUREBUFFER; +typedef struct IDirectSoundNotify *LPDIRECTSOUNDNOTIFY; + + +#if DIRECTSOUND_VERSION >= 0x0800 + +typedef struct IDirectSoundFXGargle *LPDIRECTSOUNDFXGARGLE; +typedef struct IDirectSoundFXChorus *LPDIRECTSOUNDFXCHORUS; +typedef struct IDirectSoundFXFlanger *LPDIRECTSOUNDFXFLANGER; +typedef struct IDirectSoundFXEcho *LPDIRECTSOUNDFXECHO; +typedef struct IDirectSoundFXDistortion *LPDIRECTSOUNDFXDISTORTION; +typedef struct IDirectSoundFXCompressor *LPDIRECTSOUNDFXCOMPRESSOR; +typedef struct IDirectSoundFXParamEq *LPDIRECTSOUNDFXPARAMEQ; +typedef struct IDirectSoundFXWavesReverb *LPDIRECTSOUNDFXWAVESREVERB; +typedef struct IDirectSoundFXI3DL2Reverb *LPDIRECTSOUNDFXI3DL2REVERB; +typedef struct IDirectSoundCaptureFXAec *LPDIRECTSOUNDCAPTUREFXAEC; +typedef struct IDirectSoundCaptureFXNoiseSuppress *LPDIRECTSOUNDCAPTUREFXNOISESUPPRESS; +typedef struct IDirectSoundFullDuplex *LPDIRECTSOUNDFULLDUPLEX; + +typedef struct IDirectSound8 *LPDIRECTSOUND8; +typedef struct IDirectSoundBuffer8 *LPDIRECTSOUNDBUFFER8; +typedef struct IDirectSound3DListener8 *LPDIRECTSOUND3DLISTENER8; +typedef struct IDirectSound3DBuffer8 *LPDIRECTSOUND3DBUFFER8; +typedef struct IDirectSoundCapture8 *LPDIRECTSOUNDCAPTURE8; +typedef struct IDirectSoundCaptureBuffer8 *LPDIRECTSOUNDCAPTUREBUFFER8; +typedef struct IDirectSoundNotify8 *LPDIRECTSOUNDNOTIFY8; +typedef struct IDirectSoundFXGargle8 *LPDIRECTSOUNDFXGARGLE8; +typedef struct IDirectSoundFXChorus8 *LPDIRECTSOUNDFXCHORUS8; +typedef struct IDirectSoundFXFlanger8 *LPDIRECTSOUNDFXFLANGER8; +typedef struct IDirectSoundFXEcho8 *LPDIRECTSOUNDFXECHO8; +typedef struct IDirectSoundFXDistortion8 *LPDIRECTSOUNDFXDISTORTION8; +typedef struct IDirectSoundFXCompressor8 *LPDIRECTSOUNDFXCOMPRESSOR8; +typedef struct IDirectSoundFXParamEq8 *LPDIRECTSOUNDFXPARAMEQ8; +typedef struct IDirectSoundFXWavesReverb8 *LPDIRECTSOUNDFXWAVESREVERB8; +typedef struct IDirectSoundFXI3DL2Reverb8 *LPDIRECTSOUNDFXI3DL2REVERB8; +typedef struct IDirectSoundCaptureFXAec8 *LPDIRECTSOUNDCAPTUREFXAEC8; +typedef struct IDirectSoundCaptureFXNoiseSuppress8 *LPDIRECTSOUNDCAPTUREFXNOISESUPPRESS8; +typedef struct IDirectSoundFullDuplex8 *LPDIRECTSOUNDFULLDUPLEX8; + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// IID definitions for the unchanged DirectSound 8.0 interfaces +// + +#if DIRECTSOUND_VERSION >= 0x0800 + +#define IID_IDirectSoundCapture8 IID_IDirectSoundCapture +#define IID_IDirectSound3DListener8 IID_IDirectSound3DListener +#define IID_IDirectSound3DBuffer8 IID_IDirectSound3DBuffer +#define IID_IDirectSoundNotify8 IID_IDirectSoundNotify +#define IID_IDirectSoundFXGargle8 IID_IDirectSoundFXGargle +#define IID_IDirectSoundFXChorus8 IID_IDirectSoundFXChorus +#define IID_IDirectSoundFXFlanger8 IID_IDirectSoundFXFlanger +#define IID_IDirectSoundFXEcho8 IID_IDirectSoundFXEcho +#define IID_IDirectSoundFXDistortion8 IID_IDirectSoundFXDistortion +#define IID_IDirectSoundFXCompressor8 IID_IDirectSoundFXCompressor +#define IID_IDirectSoundFXParamEq8 IID_IDirectSoundFXParamEq +#define IID_IDirectSoundFXWavesReverb8 IID_IDirectSoundFXWavesReverb +#define IID_IDirectSoundFXI3DL2Reverb8 IID_IDirectSoundFXI3DL2Reverb +#define IID_IDirectSoundCaptureFXAec8 IID_IDirectSoundCaptureFXAec +#define IID_IDirectSoundCaptureFXNoiseSuppress8 IID_IDirectSoundCaptureFXNoiseSuppress +#define IID_IDirectSoundFullDuplex8 IID_IDirectSoundFullDuplex + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// Compatibility typedefs +// + +#ifndef _LPCWAVEFORMATEX_DEFINED +#define _LPCWAVEFORMATEX_DEFINED +typedef const WAVEFORMATEX *LPCWAVEFORMATEX; +#endif // _LPCWAVEFORMATEX_DEFINED + +#ifndef __LPCGUID_DEFINED__ +#define __LPCGUID_DEFINED__ +typedef const GUID *LPCGUID; +#endif // __LPCGUID_DEFINED__ + +typedef LPDIRECTSOUND *LPLPDIRECTSOUND; +typedef LPDIRECTSOUNDBUFFER *LPLPDIRECTSOUNDBUFFER; +typedef LPDIRECTSOUND3DLISTENER *LPLPDIRECTSOUND3DLISTENER; +typedef LPDIRECTSOUND3DBUFFER *LPLPDIRECTSOUND3DBUFFER; +typedef LPDIRECTSOUNDCAPTURE *LPLPDIRECTSOUNDCAPTURE; +typedef LPDIRECTSOUNDCAPTUREBUFFER *LPLPDIRECTSOUNDCAPTUREBUFFER; +typedef LPDIRECTSOUNDNOTIFY *LPLPDIRECTSOUNDNOTIFY; + +#if DIRECTSOUND_VERSION >= 0x0800 +typedef LPDIRECTSOUND8 *LPLPDIRECTSOUND8; +typedef LPDIRECTSOUNDBUFFER8 *LPLPDIRECTSOUNDBUFFER8; +typedef LPDIRECTSOUNDCAPTURE8 *LPLPDIRECTSOUNDCAPTURE8; +typedef LPDIRECTSOUNDCAPTUREBUFFER8 *LPLPDIRECTSOUNDCAPTUREBUFFER8; +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// Structures +// + +typedef struct _DSCAPS +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwMinSecondarySampleRate; + DWORD dwMaxSecondarySampleRate; + DWORD dwPrimaryBuffers; + DWORD dwMaxHwMixingAllBuffers; + DWORD dwMaxHwMixingStaticBuffers; + DWORD dwMaxHwMixingStreamingBuffers; + DWORD dwFreeHwMixingAllBuffers; + DWORD dwFreeHwMixingStaticBuffers; + DWORD dwFreeHwMixingStreamingBuffers; + DWORD dwMaxHw3DAllBuffers; + DWORD dwMaxHw3DStaticBuffers; + DWORD dwMaxHw3DStreamingBuffers; + DWORD dwFreeHw3DAllBuffers; + DWORD dwFreeHw3DStaticBuffers; + DWORD dwFreeHw3DStreamingBuffers; + DWORD dwTotalHwMemBytes; + DWORD dwFreeHwMemBytes; + DWORD dwMaxContigFreeHwMemBytes; + DWORD dwUnlockTransferRateHwBuffers; + DWORD dwPlayCpuOverheadSwBuffers; + DWORD dwReserved1; + DWORD dwReserved2; +} DSCAPS, *LPDSCAPS; + +typedef const DSCAPS *LPCDSCAPS; + +typedef struct _DSBCAPS +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwUnlockTransferRate; + DWORD dwPlayCpuOverhead; +} DSBCAPS, *LPDSBCAPS; + +typedef const DSBCAPS *LPCDSBCAPS; + +#if DIRECTSOUND_VERSION >= 0x0800 + + typedef struct _DSEFFECTDESC + { + DWORD dwSize; + DWORD dwFlags; + GUID guidDSFXClass; + DWORD_PTR dwReserved1; + DWORD_PTR dwReserved2; + } DSEFFECTDESC, *LPDSEFFECTDESC; + typedef const DSEFFECTDESC *LPCDSEFFECTDESC; + + #define DSFX_LOCHARDWARE 0x00000001 + #define DSFX_LOCSOFTWARE 0x00000002 + + enum + { + DSFXR_PRESENT, // 0 + DSFXR_LOCHARDWARE, // 1 + DSFXR_LOCSOFTWARE, // 2 + DSFXR_UNALLOCATED, // 3 + DSFXR_FAILED, // 4 + DSFXR_UNKNOWN, // 5 + DSFXR_SENDLOOP // 6 + }; + + typedef struct _DSCEFFECTDESC + { + DWORD dwSize; + DWORD dwFlags; + GUID guidDSCFXClass; + GUID guidDSCFXInstance; + DWORD dwReserved1; + DWORD dwReserved2; + } DSCEFFECTDESC, *LPDSCEFFECTDESC; + typedef const DSCEFFECTDESC *LPCDSCEFFECTDESC; + + #define DSCFX_LOCHARDWARE 0x00000001 + #define DSCFX_LOCSOFTWARE 0x00000002 + + #define DSCFXR_LOCHARDWARE 0x00000010 + #define DSCFXR_LOCSOFTWARE 0x00000020 + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +typedef struct _DSBUFFERDESC +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; + LPWAVEFORMATEX lpwfxFormat; +#if DIRECTSOUND_VERSION >= 0x0700 + GUID guid3DAlgorithm; +#endif +} DSBUFFERDESC, *LPDSBUFFERDESC; + +typedef const DSBUFFERDESC *LPCDSBUFFERDESC; + +// Older version of this structure: + +typedef struct _DSBUFFERDESC1 +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; + LPWAVEFORMATEX lpwfxFormat; +} DSBUFFERDESC1, *LPDSBUFFERDESC1; + +typedef const DSBUFFERDESC1 *LPCDSBUFFERDESC1; + +typedef struct _DS3DBUFFER +{ + DWORD dwSize; + D3DVECTOR vPosition; + D3DVECTOR vVelocity; + DWORD dwInsideConeAngle; + DWORD dwOutsideConeAngle; + D3DVECTOR vConeOrientation; + LONG lConeOutsideVolume; + D3DVALUE flMinDistance; + D3DVALUE flMaxDistance; + DWORD dwMode; +} DS3DBUFFER, *LPDS3DBUFFER; + +typedef const DS3DBUFFER *LPCDS3DBUFFER; + +typedef struct _DS3DLISTENER +{ + DWORD dwSize; + D3DVECTOR vPosition; + D3DVECTOR vVelocity; + D3DVECTOR vOrientFront; + D3DVECTOR vOrientTop; + D3DVALUE flDistanceFactor; + D3DVALUE flRolloffFactor; + D3DVALUE flDopplerFactor; +} DS3DLISTENER, *LPDS3DLISTENER; + +typedef const DS3DLISTENER *LPCDS3DLISTENER; + +typedef struct _DSCCAPS +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwFormats; + DWORD dwChannels; +} DSCCAPS, *LPDSCCAPS; + +typedef const DSCCAPS *LPCDSCCAPS; + +typedef struct _DSCBUFFERDESC1 +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; + LPWAVEFORMATEX lpwfxFormat; +} DSCBUFFERDESC1, *LPDSCBUFFERDESC1; + +typedef struct _DSCBUFFERDESC +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; + LPWAVEFORMATEX lpwfxFormat; +#if DIRECTSOUND_VERSION >= 0x0800 + DWORD dwFXCount; + LPDSCEFFECTDESC lpDSCFXDesc; +#endif +} DSCBUFFERDESC, *LPDSCBUFFERDESC; + +typedef const DSCBUFFERDESC *LPCDSCBUFFERDESC; + +typedef struct _DSCBCAPS +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; +} DSCBCAPS, *LPDSCBCAPS; + +typedef const DSCBCAPS *LPCDSCBCAPS; + +typedef struct _DSBPOSITIONNOTIFY +{ + DWORD dwOffset; + HANDLE hEventNotify; +} DSBPOSITIONNOTIFY, *LPDSBPOSITIONNOTIFY; + +typedef const DSBPOSITIONNOTIFY *LPCDSBPOSITIONNOTIFY; + +// +// DirectSound API +// + +typedef BOOL (CALLBACK *LPDSENUMCALLBACKA)(LPGUID, LPCSTR, LPCSTR, LPVOID); +typedef BOOL (CALLBACK *LPDSENUMCALLBACKW)(LPGUID, LPCWSTR, LPCWSTR, LPVOID); + +extern HRESULT WINAPI DirectSoundCreate(LPCGUID pcGuidDevice, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter); +extern HRESULT WINAPI DirectSoundEnumerateA(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext); +extern HRESULT WINAPI DirectSoundEnumerateW(LPDSENUMCALLBACKW pDSEnumCallback, LPVOID pContext); + +extern HRESULT WINAPI DirectSoundCaptureCreate(LPCGUID pcGuidDevice, LPDIRECTSOUNDCAPTURE *ppDSC, LPUNKNOWN pUnkOuter); +extern HRESULT WINAPI DirectSoundCaptureEnumerateA(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext); +extern HRESULT WINAPI DirectSoundCaptureEnumerateW(LPDSENUMCALLBACKW pDSEnumCallback, LPVOID pContext); + +#if DIRECTSOUND_VERSION >= 0x0800 +extern HRESULT WINAPI DirectSoundCreate8(LPCGUID pcGuidDevice, LPDIRECTSOUND8 *ppDS8, LPUNKNOWN pUnkOuter); +extern HRESULT WINAPI DirectSoundCaptureCreate8(LPCGUID pcGuidDevice, LPDIRECTSOUNDCAPTURE8 *ppDSC8, LPUNKNOWN pUnkOuter); +extern HRESULT WINAPI DirectSoundFullDuplexCreate(LPCGUID pcGuidCaptureDevice, LPCGUID pcGuidRenderDevice, + LPCDSCBUFFERDESC pcDSCBufferDesc, LPCDSBUFFERDESC pcDSBufferDesc, HWND hWnd, + DWORD dwLevel, LPDIRECTSOUNDFULLDUPLEX* ppDSFD, LPDIRECTSOUNDCAPTUREBUFFER8 *ppDSCBuffer8, + LPDIRECTSOUNDBUFFER8 *ppDSBuffer8, LPUNKNOWN pUnkOuter); +#define DirectSoundFullDuplexCreate8 DirectSoundFullDuplexCreate + +extern HRESULT WINAPI GetDeviceID(LPCGUID pGuidSrc, LPGUID pGuidDest); +#endif // DIRECTSOUND_VERSION >= 0x0800 + +#ifdef UNICODE +#define LPDSENUMCALLBACK LPDSENUMCALLBACKW +#define DirectSoundEnumerate DirectSoundEnumerateW +#define DirectSoundCaptureEnumerate DirectSoundCaptureEnumerateW +#else // UNICODE +#define LPDSENUMCALLBACK LPDSENUMCALLBACKA +#define DirectSoundEnumerate DirectSoundEnumerateA +#define DirectSoundCaptureEnumerate DirectSoundCaptureEnumerateA +#endif // UNICODE + +// +// IUnknown +// + +#if !defined(__cplusplus) || defined(CINTERFACE) +#ifndef IUnknown_QueryInterface +#define IUnknown_QueryInterface(p,a,b) (p)->lpVtbl->QueryInterface(p,a,b) +#endif // IUnknown_QueryInterface +#ifndef IUnknown_AddRef +#define IUnknown_AddRef(p) (p)->lpVtbl->AddRef(p) +#endif // IUnknown_AddRef +#ifndef IUnknown_Release +#define IUnknown_Release(p) (p)->lpVtbl->Release(p) +#endif // IUnknown_Release +#else // !defined(__cplusplus) || defined(CINTERFACE) +#ifndef IUnknown_QueryInterface +#define IUnknown_QueryInterface(p,a,b) (p)->QueryInterface(a,b) +#endif // IUnknown_QueryInterface +#ifndef IUnknown_AddRef +#define IUnknown_AddRef(p) (p)->AddRef() +#endif // IUnknown_AddRef +#ifndef IUnknown_Release +#define IUnknown_Release(p) (p)->Release() +#endif // IUnknown_Release +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#ifndef __IReferenceClock_INTERFACE_DEFINED__ +#define __IReferenceClock_INTERFACE_DEFINED__ + +typedef LONGLONG REFERENCE_TIME; +typedef REFERENCE_TIME *LPREFERENCE_TIME; + +DEFINE_GUID(IID_IReferenceClock, 0x56a86897, 0x0ad4, 0x11ce, 0xb0, 0x3a, 0x00, 0x20, 0xaf, 0x0b, 0xa7, 0x70); + +#undef INTERFACE +#define INTERFACE IReferenceClock + +DECLARE_INTERFACE_(IReferenceClock, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IReferenceClock methods + STDMETHOD(GetTime) (THIS_ REFERENCE_TIME *pTime) PURE; + STDMETHOD(AdviseTime) (THIS_ REFERENCE_TIME rtBaseTime, REFERENCE_TIME rtStreamTime, + HANDLE hEvent, LPDWORD pdwAdviseCookie) PURE; + STDMETHOD(AdvisePeriodic) (THIS_ REFERENCE_TIME rtStartTime, REFERENCE_TIME rtPeriodTime, + HANDLE hSemaphore, LPDWORD pdwAdviseCookie) PURE; + STDMETHOD(Unadvise) (THIS_ DWORD dwAdviseCookie) PURE; +}; + +#endif // __IReferenceClock_INTERFACE_DEFINED__ + +#ifndef IReferenceClock_QueryInterface + +#define IReferenceClock_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IReferenceClock_AddRef(p) IUnknown_AddRef(p) +#define IReferenceClock_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IReferenceClock_GetTime(p,a) (p)->lpVtbl->GetTime(p,a) +#define IReferenceClock_AdviseTime(p,a,b,c,d) (p)->lpVtbl->AdviseTime(p,a,b,c,d) +#define IReferenceClock_AdvisePeriodic(p,a,b,c,d) (p)->lpVtbl->AdvisePeriodic(p,a,b,c,d) +#define IReferenceClock_Unadvise(p,a) (p)->lpVtbl->Unadvise(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IReferenceClock_GetTime(p,a) (p)->GetTime(a) +#define IReferenceClock_AdviseTime(p,a,b,c,d) (p)->AdviseTime(a,b,c,d) +#define IReferenceClock_AdvisePeriodic(p,a,b,c,d) (p)->AdvisePeriodic(a,b,c,d) +#define IReferenceClock_Unadvise(p,a) (p)->Unadvise(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#endif // IReferenceClock_QueryInterface + +// +// IDirectSound +// + +DEFINE_GUID(IID_IDirectSound, 0x279AFA83, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60); + +#undef INTERFACE +#define INTERFACE IDirectSound + +DECLARE_INTERFACE_(IDirectSound, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSound methods + STDMETHOD(CreateSoundBuffer) (THIS_ LPCDSBUFFERDESC pcDSBufferDesc, LPDIRECTSOUNDBUFFER *ppDSBuffer, LPUNKNOWN pUnkOuter) PURE; + STDMETHOD(GetCaps) (THIS_ LPDSCAPS pDSCaps) PURE; + STDMETHOD(DuplicateSoundBuffer) (THIS_ LPDIRECTSOUNDBUFFER pDSBufferOriginal, LPDIRECTSOUNDBUFFER *ppDSBufferDuplicate) PURE; + STDMETHOD(SetCooperativeLevel) (THIS_ HWND hwnd, DWORD dwLevel) PURE; + STDMETHOD(Compact) (THIS) PURE; + STDMETHOD(GetSpeakerConfig) (THIS_ LPDWORD pdwSpeakerConfig) PURE; + STDMETHOD(SetSpeakerConfig) (THIS_ DWORD dwSpeakerConfig) PURE; + STDMETHOD(Initialize) (THIS_ LPCGUID pcGuidDevice) PURE; +}; + +#define IDirectSound_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSound_AddRef(p) IUnknown_AddRef(p) +#define IDirectSound_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound_CreateSoundBuffer(p,a,b,c) (p)->lpVtbl->CreateSoundBuffer(p,a,b,c) +#define IDirectSound_GetCaps(p,a) (p)->lpVtbl->GetCaps(p,a) +#define IDirectSound_DuplicateSoundBuffer(p,a,b) (p)->lpVtbl->DuplicateSoundBuffer(p,a,b) +#define IDirectSound_SetCooperativeLevel(p,a,b) (p)->lpVtbl->SetCooperativeLevel(p,a,b) +#define IDirectSound_Compact(p) (p)->lpVtbl->Compact(p) +#define IDirectSound_GetSpeakerConfig(p,a) (p)->lpVtbl->GetSpeakerConfig(p,a) +#define IDirectSound_SetSpeakerConfig(p,b) (p)->lpVtbl->SetSpeakerConfig(p,b) +#define IDirectSound_Initialize(p,a) (p)->lpVtbl->Initialize(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound_CreateSoundBuffer(p,a,b,c) (p)->CreateSoundBuffer(a,b,c) +#define IDirectSound_GetCaps(p,a) (p)->GetCaps(a) +#define IDirectSound_DuplicateSoundBuffer(p,a,b) (p)->DuplicateSoundBuffer(a,b) +#define IDirectSound_SetCooperativeLevel(p,a,b) (p)->SetCooperativeLevel(a,b) +#define IDirectSound_Compact(p) (p)->Compact() +#define IDirectSound_GetSpeakerConfig(p,a) (p)->GetSpeakerConfig(a) +#define IDirectSound_SetSpeakerConfig(p,b) (p)->SetSpeakerConfig(b) +#define IDirectSound_Initialize(p,a) (p)->Initialize(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#if DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSound8 +// + +DEFINE_GUID(IID_IDirectSound8, 0xC50A7E93, 0xF395, 0x4834, 0x9E, 0xF6, 0x7F, 0xA9, 0x9D, 0xE5, 0x09, 0x66); + +#undef INTERFACE +#define INTERFACE IDirectSound8 + +DECLARE_INTERFACE_(IDirectSound8, IDirectSound) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSound methods + STDMETHOD(CreateSoundBuffer) (THIS_ LPCDSBUFFERDESC pcDSBufferDesc, LPDIRECTSOUNDBUFFER *ppDSBuffer, LPUNKNOWN pUnkOuter) PURE; + STDMETHOD(GetCaps) (THIS_ LPDSCAPS pDSCaps) PURE; + STDMETHOD(DuplicateSoundBuffer) (THIS_ LPDIRECTSOUNDBUFFER pDSBufferOriginal, LPDIRECTSOUNDBUFFER *ppDSBufferDuplicate) PURE; + STDMETHOD(SetCooperativeLevel) (THIS_ HWND hwnd, DWORD dwLevel) PURE; + STDMETHOD(Compact) (THIS) PURE; + STDMETHOD(GetSpeakerConfig) (THIS_ LPDWORD pdwSpeakerConfig) PURE; + STDMETHOD(SetSpeakerConfig) (THIS_ DWORD dwSpeakerConfig) PURE; + STDMETHOD(Initialize) (THIS_ LPCGUID pcGuidDevice) PURE; + + // IDirectSound8 methods + STDMETHOD(VerifyCertification) (THIS_ LPDWORD pdwCertified) PURE; +}; + +#define IDirectSound8_QueryInterface(p,a,b) IDirectSound_QueryInterface(p,a,b) +#define IDirectSound8_AddRef(p) IDirectSound_AddRef(p) +#define IDirectSound8_Release(p) IDirectSound_Release(p) +#define IDirectSound8_CreateSoundBuffer(p,a,b,c) IDirectSound_CreateSoundBuffer(p,a,b,c) +#define IDirectSound8_GetCaps(p,a) IDirectSound_GetCaps(p,a) +#define IDirectSound8_DuplicateSoundBuffer(p,a,b) IDirectSound_DuplicateSoundBuffer(p,a,b) +#define IDirectSound8_SetCooperativeLevel(p,a,b) IDirectSound_SetCooperativeLevel(p,a,b) +#define IDirectSound8_Compact(p) IDirectSound_Compact(p) +#define IDirectSound8_GetSpeakerConfig(p,a) IDirectSound_GetSpeakerConfig(p,a) +#define IDirectSound8_SetSpeakerConfig(p,a) IDirectSound_SetSpeakerConfig(p,a) +#define IDirectSound8_Initialize(p,a) IDirectSound_Initialize(p,a) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound8_VerifyCertification(p,a) (p)->lpVtbl->VerifyCertification(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound8_VerifyCertification(p,a) (p)->VerifyCertification(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSoundBuffer +// + +DEFINE_GUID(IID_IDirectSoundBuffer, 0x279AFA85, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60); + +#undef INTERFACE +#define INTERFACE IDirectSoundBuffer + +DECLARE_INTERFACE_(IDirectSoundBuffer, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundBuffer methods + STDMETHOD(GetCaps) (THIS_ LPDSBCAPS pDSBufferCaps) PURE; + STDMETHOD(GetCurrentPosition) (THIS_ LPDWORD pdwCurrentPlayCursor, LPDWORD pdwCurrentWriteCursor) PURE; + STDMETHOD(GetFormat) (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE; + STDMETHOD(GetVolume) (THIS_ LPLONG plVolume) PURE; + STDMETHOD(GetPan) (THIS_ LPLONG plPan) PURE; + STDMETHOD(GetFrequency) (THIS_ LPDWORD pdwFrequency) PURE; + STDMETHOD(GetStatus) (THIS_ LPDWORD pdwStatus) PURE; + STDMETHOD(Initialize) (THIS_ LPDIRECTSOUND pDirectSound, LPCDSBUFFERDESC pcDSBufferDesc) PURE; + STDMETHOD(Lock) (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1, + LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE; + STDMETHOD(Play) (THIS_ DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) PURE; + STDMETHOD(SetCurrentPosition) (THIS_ DWORD dwNewPosition) PURE; + STDMETHOD(SetFormat) (THIS_ LPCWAVEFORMATEX pcfxFormat) PURE; + STDMETHOD(SetVolume) (THIS_ LONG lVolume) PURE; + STDMETHOD(SetPan) (THIS_ LONG lPan) PURE; + STDMETHOD(SetFrequency) (THIS_ DWORD dwFrequency) PURE; + STDMETHOD(Stop) (THIS) PURE; + STDMETHOD(Unlock) (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE; + STDMETHOD(Restore) (THIS) PURE; +}; + +#define IDirectSoundBuffer_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundBuffer_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundBuffer_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundBuffer_GetCaps(p,a) (p)->lpVtbl->GetCaps(p,a) +#define IDirectSoundBuffer_GetCurrentPosition(p,a,b) (p)->lpVtbl->GetCurrentPosition(p,a,b) +#define IDirectSoundBuffer_GetFormat(p,a,b,c) (p)->lpVtbl->GetFormat(p,a,b,c) +#define IDirectSoundBuffer_GetVolume(p,a) (p)->lpVtbl->GetVolume(p,a) +#define IDirectSoundBuffer_GetPan(p,a) (p)->lpVtbl->GetPan(p,a) +#define IDirectSoundBuffer_GetFrequency(p,a) (p)->lpVtbl->GetFrequency(p,a) +#define IDirectSoundBuffer_GetStatus(p,a) (p)->lpVtbl->GetStatus(p,a) +#define IDirectSoundBuffer_Initialize(p,a,b) (p)->lpVtbl->Initialize(p,a,b) +#define IDirectSoundBuffer_Lock(p,a,b,c,d,e,f,g) (p)->lpVtbl->Lock(p,a,b,c,d,e,f,g) +#define IDirectSoundBuffer_Play(p,a,b,c) (p)->lpVtbl->Play(p,a,b,c) +#define IDirectSoundBuffer_SetCurrentPosition(p,a) (p)->lpVtbl->SetCurrentPosition(p,a) +#define IDirectSoundBuffer_SetFormat(p,a) (p)->lpVtbl->SetFormat(p,a) +#define IDirectSoundBuffer_SetVolume(p,a) (p)->lpVtbl->SetVolume(p,a) +#define IDirectSoundBuffer_SetPan(p,a) (p)->lpVtbl->SetPan(p,a) +#define IDirectSoundBuffer_SetFrequency(p,a) (p)->lpVtbl->SetFrequency(p,a) +#define IDirectSoundBuffer_Stop(p) (p)->lpVtbl->Stop(p) +#define IDirectSoundBuffer_Unlock(p,a,b,c,d) (p)->lpVtbl->Unlock(p,a,b,c,d) +#define IDirectSoundBuffer_Restore(p) (p)->lpVtbl->Restore(p) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundBuffer_GetCaps(p,a) (p)->GetCaps(a) +#define IDirectSoundBuffer_GetCurrentPosition(p,a,b) (p)->GetCurrentPosition(a,b) +#define IDirectSoundBuffer_GetFormat(p,a,b,c) (p)->GetFormat(a,b,c) +#define IDirectSoundBuffer_GetVolume(p,a) (p)->GetVolume(a) +#define IDirectSoundBuffer_GetPan(p,a) (p)->GetPan(a) +#define IDirectSoundBuffer_GetFrequency(p,a) (p)->GetFrequency(a) +#define IDirectSoundBuffer_GetStatus(p,a) (p)->GetStatus(a) +#define IDirectSoundBuffer_Initialize(p,a,b) (p)->Initialize(a,b) +#define IDirectSoundBuffer_Lock(p,a,b,c,d,e,f,g) (p)->Lock(a,b,c,d,e,f,g) +#define IDirectSoundBuffer_Play(p,a,b,c) (p)->Play(a,b,c) +#define IDirectSoundBuffer_SetCurrentPosition(p,a) (p)->SetCurrentPosition(a) +#define IDirectSoundBuffer_SetFormat(p,a) (p)->SetFormat(a) +#define IDirectSoundBuffer_SetVolume(p,a) (p)->SetVolume(a) +#define IDirectSoundBuffer_SetPan(p,a) (p)->SetPan(a) +#define IDirectSoundBuffer_SetFrequency(p,a) (p)->SetFrequency(a) +#define IDirectSoundBuffer_Stop(p) (p)->Stop() +#define IDirectSoundBuffer_Unlock(p,a,b,c,d) (p)->Unlock(a,b,c,d) +#define IDirectSoundBuffer_Restore(p) (p)->Restore() +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#if DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSoundBuffer8 +// + +DEFINE_GUID(IID_IDirectSoundBuffer8, 0x6825a449, 0x7524, 0x4d82, 0x92, 0x0f, 0x50, 0xe3, 0x6a, 0xb3, 0xab, 0x1e); + +#undef INTERFACE +#define INTERFACE IDirectSoundBuffer8 + +DECLARE_INTERFACE_(IDirectSoundBuffer8, IDirectSoundBuffer) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundBuffer methods + STDMETHOD(GetCaps) (THIS_ LPDSBCAPS pDSBufferCaps) PURE; + STDMETHOD(GetCurrentPosition) (THIS_ LPDWORD pdwCurrentPlayCursor, LPDWORD pdwCurrentWriteCursor) PURE; + STDMETHOD(GetFormat) (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE; + STDMETHOD(GetVolume) (THIS_ LPLONG plVolume) PURE; + STDMETHOD(GetPan) (THIS_ LPLONG plPan) PURE; + STDMETHOD(GetFrequency) (THIS_ LPDWORD pdwFrequency) PURE; + STDMETHOD(GetStatus) (THIS_ LPDWORD pdwStatus) PURE; + STDMETHOD(Initialize) (THIS_ LPDIRECTSOUND pDirectSound, LPCDSBUFFERDESC pcDSBufferDesc) PURE; + STDMETHOD(Lock) (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1, + LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE; + STDMETHOD(Play) (THIS_ DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) PURE; + STDMETHOD(SetCurrentPosition) (THIS_ DWORD dwNewPosition) PURE; + STDMETHOD(SetFormat) (THIS_ LPCWAVEFORMATEX pcfxFormat) PURE; + STDMETHOD(SetVolume) (THIS_ LONG lVolume) PURE; + STDMETHOD(SetPan) (THIS_ LONG lPan) PURE; + STDMETHOD(SetFrequency) (THIS_ DWORD dwFrequency) PURE; + STDMETHOD(Stop) (THIS) PURE; + STDMETHOD(Unlock) (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE; + STDMETHOD(Restore) (THIS) PURE; + + // IDirectSoundBuffer8 methods + STDMETHOD(SetFX) (THIS_ DWORD dwEffectsCount, LPDSEFFECTDESC pDSFXDesc, LPDWORD pdwResultCodes) PURE; + STDMETHOD(AcquireResources) (THIS_ DWORD dwFlags, DWORD dwEffectsCount, LPDWORD pdwResultCodes) PURE; + STDMETHOD(GetObjectInPath) (THIS_ REFGUID rguidObject, DWORD dwIndex, REFGUID rguidInterface, LPVOID *ppObject) PURE; +}; + +// Special GUID meaning "select all objects" for use in GetObjectInPath() +DEFINE_GUID(GUID_All_Objects, 0xaa114de5, 0xc262, 0x4169, 0xa1, 0xc8, 0x23, 0xd6, 0x98, 0xcc, 0x73, 0xb5); + +#define IDirectSoundBuffer8_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundBuffer8_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundBuffer8_Release(p) IUnknown_Release(p) + +#define IDirectSoundBuffer8_GetCaps(p,a) IDirectSoundBuffer_GetCaps(p,a) +#define IDirectSoundBuffer8_GetCurrentPosition(p,a,b) IDirectSoundBuffer_GetCurrentPosition(p,a,b) +#define IDirectSoundBuffer8_GetFormat(p,a,b,c) IDirectSoundBuffer_GetFormat(p,a,b,c) +#define IDirectSoundBuffer8_GetVolume(p,a) IDirectSoundBuffer_GetVolume(p,a) +#define IDirectSoundBuffer8_GetPan(p,a) IDirectSoundBuffer_GetPan(p,a) +#define IDirectSoundBuffer8_GetFrequency(p,a) IDirectSoundBuffer_GetFrequency(p,a) +#define IDirectSoundBuffer8_GetStatus(p,a) IDirectSoundBuffer_GetStatus(p,a) +#define IDirectSoundBuffer8_Initialize(p,a,b) IDirectSoundBuffer_Initialize(p,a,b) +#define IDirectSoundBuffer8_Lock(p,a,b,c,d,e,f,g) IDirectSoundBuffer_Lock(p,a,b,c,d,e,f,g) +#define IDirectSoundBuffer8_Play(p,a,b,c) IDirectSoundBuffer_Play(p,a,b,c) +#define IDirectSoundBuffer8_SetCurrentPosition(p,a) IDirectSoundBuffer_SetCurrentPosition(p,a) +#define IDirectSoundBuffer8_SetFormat(p,a) IDirectSoundBuffer_SetFormat(p,a) +#define IDirectSoundBuffer8_SetVolume(p,a) IDirectSoundBuffer_SetVolume(p,a) +#define IDirectSoundBuffer8_SetPan(p,a) IDirectSoundBuffer_SetPan(p,a) +#define IDirectSoundBuffer8_SetFrequency(p,a) IDirectSoundBuffer_SetFrequency(p,a) +#define IDirectSoundBuffer8_Stop(p) IDirectSoundBuffer_Stop(p) +#define IDirectSoundBuffer8_Unlock(p,a,b,c,d) IDirectSoundBuffer_Unlock(p,a,b,c,d) +#define IDirectSoundBuffer8_Restore(p) IDirectSoundBuffer_Restore(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundBuffer8_SetFX(p,a,b,c) (p)->lpVtbl->SetFX(p,a,b,c) +#define IDirectSoundBuffer8_AcquireResources(p,a,b,c) (p)->lpVtbl->AcquireResources(p,a,b,c) +#define IDirectSoundBuffer8_GetObjectInPath(p,a,b,c,d) (p)->lpVtbl->GetObjectInPath(p,a,b,c,d) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundBuffer8_SetFX(p,a,b,c) (p)->SetFX(a,b,c) +#define IDirectSoundBuffer8_AcquireResources(p,a,b,c) (p)->AcquireResources(a,b,c) +#define IDirectSoundBuffer8_GetObjectInPath(p,a,b,c,d) (p)->GetObjectInPath(a,b,c,d) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSound3DListener +// + +DEFINE_GUID(IID_IDirectSound3DListener, 0x279AFA84, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60); + +#undef INTERFACE +#define INTERFACE IDirectSound3DListener + +DECLARE_INTERFACE_(IDirectSound3DListener, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSound3DListener methods + STDMETHOD(GetAllParameters) (THIS_ LPDS3DLISTENER pListener) PURE; + STDMETHOD(GetDistanceFactor) (THIS_ D3DVALUE* pflDistanceFactor) PURE; + STDMETHOD(GetDopplerFactor) (THIS_ D3DVALUE* pflDopplerFactor) PURE; + STDMETHOD(GetOrientation) (THIS_ D3DVECTOR* pvOrientFront, D3DVECTOR* pvOrientTop) PURE; + STDMETHOD(GetPosition) (THIS_ D3DVECTOR* pvPosition) PURE; + STDMETHOD(GetRolloffFactor) (THIS_ D3DVALUE* pflRolloffFactor) PURE; + STDMETHOD(GetVelocity) (THIS_ D3DVECTOR* pvVelocity) PURE; + STDMETHOD(SetAllParameters) (THIS_ LPCDS3DLISTENER pcListener, DWORD dwApply) PURE; + STDMETHOD(SetDistanceFactor) (THIS_ D3DVALUE flDistanceFactor, DWORD dwApply) PURE; + STDMETHOD(SetDopplerFactor) (THIS_ D3DVALUE flDopplerFactor, DWORD dwApply) PURE; + STDMETHOD(SetOrientation) (THIS_ D3DVALUE xFront, D3DVALUE yFront, D3DVALUE zFront, + D3DVALUE xTop, D3DVALUE yTop, D3DVALUE zTop, DWORD dwApply) PURE; + STDMETHOD(SetPosition) (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE; + STDMETHOD(SetRolloffFactor) (THIS_ D3DVALUE flRolloffFactor, DWORD dwApply) PURE; + STDMETHOD(SetVelocity) (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE; + STDMETHOD(CommitDeferredSettings) (THIS) PURE; +}; + +#define IDirectSound3DListener_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSound3DListener_AddRef(p) IUnknown_AddRef(p) +#define IDirectSound3DListener_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound3DListener_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#define IDirectSound3DListener_GetDistanceFactor(p,a) (p)->lpVtbl->GetDistanceFactor(p,a) +#define IDirectSound3DListener_GetDopplerFactor(p,a) (p)->lpVtbl->GetDopplerFactor(p,a) +#define IDirectSound3DListener_GetOrientation(p,a,b) (p)->lpVtbl->GetOrientation(p,a,b) +#define IDirectSound3DListener_GetPosition(p,a) (p)->lpVtbl->GetPosition(p,a) +#define IDirectSound3DListener_GetRolloffFactor(p,a) (p)->lpVtbl->GetRolloffFactor(p,a) +#define IDirectSound3DListener_GetVelocity(p,a) (p)->lpVtbl->GetVelocity(p,a) +#define IDirectSound3DListener_SetAllParameters(p,a,b) (p)->lpVtbl->SetAllParameters(p,a,b) +#define IDirectSound3DListener_SetDistanceFactor(p,a,b) (p)->lpVtbl->SetDistanceFactor(p,a,b) +#define IDirectSound3DListener_SetDopplerFactor(p,a,b) (p)->lpVtbl->SetDopplerFactor(p,a,b) +#define IDirectSound3DListener_SetOrientation(p,a,b,c,d,e,f,g) (p)->lpVtbl->SetOrientation(p,a,b,c,d,e,f,g) +#define IDirectSound3DListener_SetPosition(p,a,b,c,d) (p)->lpVtbl->SetPosition(p,a,b,c,d) +#define IDirectSound3DListener_SetRolloffFactor(p,a,b) (p)->lpVtbl->SetRolloffFactor(p,a,b) +#define IDirectSound3DListener_SetVelocity(p,a,b,c,d) (p)->lpVtbl->SetVelocity(p,a,b,c,d) +#define IDirectSound3DListener_CommitDeferredSettings(p) (p)->lpVtbl->CommitDeferredSettings(p) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound3DListener_GetAllParameters(p,a) (p)->GetAllParameters(a) +#define IDirectSound3DListener_GetDistanceFactor(p,a) (p)->GetDistanceFactor(a) +#define IDirectSound3DListener_GetDopplerFactor(p,a) (p)->GetDopplerFactor(a) +#define IDirectSound3DListener_GetOrientation(p,a,b) (p)->GetOrientation(a,b) +#define IDirectSound3DListener_GetPosition(p,a) (p)->GetPosition(a) +#define IDirectSound3DListener_GetRolloffFactor(p,a) (p)->GetRolloffFactor(a) +#define IDirectSound3DListener_GetVelocity(p,a) (p)->GetVelocity(a) +#define IDirectSound3DListener_SetAllParameters(p,a,b) (p)->SetAllParameters(a,b) +#define IDirectSound3DListener_SetDistanceFactor(p,a,b) (p)->SetDistanceFactor(a,b) +#define IDirectSound3DListener_SetDopplerFactor(p,a,b) (p)->SetDopplerFactor(a,b) +#define IDirectSound3DListener_SetOrientation(p,a,b,c,d,e,f,g) (p)->SetOrientation(a,b,c,d,e,f,g) +#define IDirectSound3DListener_SetPosition(p,a,b,c,d) (p)->SetPosition(a,b,c,d) +#define IDirectSound3DListener_SetRolloffFactor(p,a,b) (p)->SetRolloffFactor(a,b) +#define IDirectSound3DListener_SetVelocity(p,a,b,c,d) (p)->SetVelocity(a,b,c,d) +#define IDirectSound3DListener_CommitDeferredSettings(p) (p)->CommitDeferredSettings() +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSound3DBuffer +// + +DEFINE_GUID(IID_IDirectSound3DBuffer, 0x279AFA86, 0x4981, 0x11CE, 0xA5, 0x21, 0x00, 0x20, 0xAF, 0x0B, 0xE5, 0x60); + +#undef INTERFACE +#define INTERFACE IDirectSound3DBuffer + +DECLARE_INTERFACE_(IDirectSound3DBuffer, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSound3DBuffer methods + STDMETHOD(GetAllParameters) (THIS_ LPDS3DBUFFER pDs3dBuffer) PURE; + STDMETHOD(GetConeAngles) (THIS_ LPDWORD pdwInsideConeAngle, LPDWORD pdwOutsideConeAngle) PURE; + STDMETHOD(GetConeOrientation) (THIS_ D3DVECTOR* pvOrientation) PURE; + STDMETHOD(GetConeOutsideVolume) (THIS_ LPLONG plConeOutsideVolume) PURE; + STDMETHOD(GetMaxDistance) (THIS_ D3DVALUE* pflMaxDistance) PURE; + STDMETHOD(GetMinDistance) (THIS_ D3DVALUE* pflMinDistance) PURE; + STDMETHOD(GetMode) (THIS_ LPDWORD pdwMode) PURE; + STDMETHOD(GetPosition) (THIS_ D3DVECTOR* pvPosition) PURE; + STDMETHOD(GetVelocity) (THIS_ D3DVECTOR* pvVelocity) PURE; + STDMETHOD(SetAllParameters) (THIS_ LPCDS3DBUFFER pcDs3dBuffer, DWORD dwApply) PURE; + STDMETHOD(SetConeAngles) (THIS_ DWORD dwInsideConeAngle, DWORD dwOutsideConeAngle, DWORD dwApply) PURE; + STDMETHOD(SetConeOrientation) (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE; + STDMETHOD(SetConeOutsideVolume) (THIS_ LONG lConeOutsideVolume, DWORD dwApply) PURE; + STDMETHOD(SetMaxDistance) (THIS_ D3DVALUE flMaxDistance, DWORD dwApply) PURE; + STDMETHOD(SetMinDistance) (THIS_ D3DVALUE flMinDistance, DWORD dwApply) PURE; + STDMETHOD(SetMode) (THIS_ DWORD dwMode, DWORD dwApply) PURE; + STDMETHOD(SetPosition) (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE; + STDMETHOD(SetVelocity) (THIS_ D3DVALUE x, D3DVALUE y, D3DVALUE z, DWORD dwApply) PURE; +}; + +#define IDirectSound3DBuffer_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSound3DBuffer_AddRef(p) IUnknown_AddRef(p) +#define IDirectSound3DBuffer_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound3DBuffer_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#define IDirectSound3DBuffer_GetConeAngles(p,a,b) (p)->lpVtbl->GetConeAngles(p,a,b) +#define IDirectSound3DBuffer_GetConeOrientation(p,a) (p)->lpVtbl->GetConeOrientation(p,a) +#define IDirectSound3DBuffer_GetConeOutsideVolume(p,a) (p)->lpVtbl->GetConeOutsideVolume(p,a) +#define IDirectSound3DBuffer_GetPosition(p,a) (p)->lpVtbl->GetPosition(p,a) +#define IDirectSound3DBuffer_GetMinDistance(p,a) (p)->lpVtbl->GetMinDistance(p,a) +#define IDirectSound3DBuffer_GetMaxDistance(p,a) (p)->lpVtbl->GetMaxDistance(p,a) +#define IDirectSound3DBuffer_GetMode(p,a) (p)->lpVtbl->GetMode(p,a) +#define IDirectSound3DBuffer_GetVelocity(p,a) (p)->lpVtbl->GetVelocity(p,a) +#define IDirectSound3DBuffer_SetAllParameters(p,a,b) (p)->lpVtbl->SetAllParameters(p,a,b) +#define IDirectSound3DBuffer_SetConeAngles(p,a,b,c) (p)->lpVtbl->SetConeAngles(p,a,b,c) +#define IDirectSound3DBuffer_SetConeOrientation(p,a,b,c,d) (p)->lpVtbl->SetConeOrientation(p,a,b,c,d) +#define IDirectSound3DBuffer_SetConeOutsideVolume(p,a,b) (p)->lpVtbl->SetConeOutsideVolume(p,a,b) +#define IDirectSound3DBuffer_SetPosition(p,a,b,c,d) (p)->lpVtbl->SetPosition(p,a,b,c,d) +#define IDirectSound3DBuffer_SetMinDistance(p,a,b) (p)->lpVtbl->SetMinDistance(p,a,b) +#define IDirectSound3DBuffer_SetMaxDistance(p,a,b) (p)->lpVtbl->SetMaxDistance(p,a,b) +#define IDirectSound3DBuffer_SetMode(p,a,b) (p)->lpVtbl->SetMode(p,a,b) +#define IDirectSound3DBuffer_SetVelocity(p,a,b,c,d) (p)->lpVtbl->SetVelocity(p,a,b,c,d) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSound3DBuffer_GetAllParameters(p,a) (p)->GetAllParameters(a) +#define IDirectSound3DBuffer_GetConeAngles(p,a,b) (p)->GetConeAngles(a,b) +#define IDirectSound3DBuffer_GetConeOrientation(p,a) (p)->GetConeOrientation(a) +#define IDirectSound3DBuffer_GetConeOutsideVolume(p,a) (p)->GetConeOutsideVolume(a) +#define IDirectSound3DBuffer_GetPosition(p,a) (p)->GetPosition(a) +#define IDirectSound3DBuffer_GetMinDistance(p,a) (p)->GetMinDistance(a) +#define IDirectSound3DBuffer_GetMaxDistance(p,a) (p)->GetMaxDistance(a) +#define IDirectSound3DBuffer_GetMode(p,a) (p)->GetMode(a) +#define IDirectSound3DBuffer_GetVelocity(p,a) (p)->GetVelocity(a) +#define IDirectSound3DBuffer_SetAllParameters(p,a,b) (p)->SetAllParameters(a,b) +#define IDirectSound3DBuffer_SetConeAngles(p,a,b,c) (p)->SetConeAngles(a,b,c) +#define IDirectSound3DBuffer_SetConeOrientation(p,a,b,c,d) (p)->SetConeOrientation(a,b,c,d) +#define IDirectSound3DBuffer_SetConeOutsideVolume(p,a,b) (p)->SetConeOutsideVolume(a,b) +#define IDirectSound3DBuffer_SetPosition(p,a,b,c,d) (p)->SetPosition(a,b,c,d) +#define IDirectSound3DBuffer_SetMinDistance(p,a,b) (p)->SetMinDistance(a,b) +#define IDirectSound3DBuffer_SetMaxDistance(p,a,b) (p)->SetMaxDistance(a,b) +#define IDirectSound3DBuffer_SetMode(p,a,b) (p)->SetMode(a,b) +#define IDirectSound3DBuffer_SetVelocity(p,a,b,c,d) (p)->SetVelocity(a,b,c,d) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundCapture +// + +DEFINE_GUID(IID_IDirectSoundCapture, 0xb0210781, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16); + +#undef INTERFACE +#define INTERFACE IDirectSoundCapture + +DECLARE_INTERFACE_(IDirectSoundCapture, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundCapture methods + STDMETHOD(CreateCaptureBuffer) (THIS_ LPCDSCBUFFERDESC pcDSCBufferDesc, LPDIRECTSOUNDCAPTUREBUFFER *ppDSCBuffer, LPUNKNOWN pUnkOuter) PURE; + STDMETHOD(GetCaps) (THIS_ LPDSCCAPS pDSCCaps) PURE; + STDMETHOD(Initialize) (THIS_ LPCGUID pcGuidDevice) PURE; +}; + +#define IDirectSoundCapture_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundCapture_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundCapture_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCapture_CreateCaptureBuffer(p,a,b,c) (p)->lpVtbl->CreateCaptureBuffer(p,a,b,c) +#define IDirectSoundCapture_GetCaps(p,a) (p)->lpVtbl->GetCaps(p,a) +#define IDirectSoundCapture_Initialize(p,a) (p)->lpVtbl->Initialize(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCapture_CreateCaptureBuffer(p,a,b,c) (p)->CreateCaptureBuffer(a,b,c) +#define IDirectSoundCapture_GetCaps(p,a) (p)->GetCaps(a) +#define IDirectSoundCapture_Initialize(p,a) (p)->Initialize(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundCaptureBuffer +// + +DEFINE_GUID(IID_IDirectSoundCaptureBuffer, 0xb0210782, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16); + +#undef INTERFACE +#define INTERFACE IDirectSoundCaptureBuffer + +DECLARE_INTERFACE_(IDirectSoundCaptureBuffer, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundCaptureBuffer methods + STDMETHOD(GetCaps) (THIS_ LPDSCBCAPS pDSCBCaps) PURE; + STDMETHOD(GetCurrentPosition) (THIS_ LPDWORD pdwCapturePosition, LPDWORD pdwReadPosition) PURE; + STDMETHOD(GetFormat) (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE; + STDMETHOD(GetStatus) (THIS_ LPDWORD pdwStatus) PURE; + STDMETHOD(Initialize) (THIS_ LPDIRECTSOUNDCAPTURE pDirectSoundCapture, LPCDSCBUFFERDESC pcDSCBufferDesc) PURE; + STDMETHOD(Lock) (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1, + LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE; + STDMETHOD(Start) (THIS_ DWORD dwFlags) PURE; + STDMETHOD(Stop) (THIS) PURE; + STDMETHOD(Unlock) (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE; +}; + +#define IDirectSoundCaptureBuffer_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundCaptureBuffer_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundCaptureBuffer_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureBuffer_GetCaps(p,a) (p)->lpVtbl->GetCaps(p,a) +#define IDirectSoundCaptureBuffer_GetCurrentPosition(p,a,b) (p)->lpVtbl->GetCurrentPosition(p,a,b) +#define IDirectSoundCaptureBuffer_GetFormat(p,a,b,c) (p)->lpVtbl->GetFormat(p,a,b,c) +#define IDirectSoundCaptureBuffer_GetStatus(p,a) (p)->lpVtbl->GetStatus(p,a) +#define IDirectSoundCaptureBuffer_Initialize(p,a,b) (p)->lpVtbl->Initialize(p,a,b) +#define IDirectSoundCaptureBuffer_Lock(p,a,b,c,d,e,f,g) (p)->lpVtbl->Lock(p,a,b,c,d,e,f,g) +#define IDirectSoundCaptureBuffer_Start(p,a) (p)->lpVtbl->Start(p,a) +#define IDirectSoundCaptureBuffer_Stop(p) (p)->lpVtbl->Stop(p) +#define IDirectSoundCaptureBuffer_Unlock(p,a,b,c,d) (p)->lpVtbl->Unlock(p,a,b,c,d) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureBuffer_GetCaps(p,a) (p)->GetCaps(a) +#define IDirectSoundCaptureBuffer_GetCurrentPosition(p,a,b) (p)->GetCurrentPosition(a,b) +#define IDirectSoundCaptureBuffer_GetFormat(p,a,b,c) (p)->GetFormat(a,b,c) +#define IDirectSoundCaptureBuffer_GetStatus(p,a) (p)->GetStatus(a) +#define IDirectSoundCaptureBuffer_Initialize(p,a,b) (p)->Initialize(a,b) +#define IDirectSoundCaptureBuffer_Lock(p,a,b,c,d,e,f,g) (p)->Lock(a,b,c,d,e,f,g) +#define IDirectSoundCaptureBuffer_Start(p,a) (p)->Start(a) +#define IDirectSoundCaptureBuffer_Stop(p) (p)->Stop() +#define IDirectSoundCaptureBuffer_Unlock(p,a,b,c,d) (p)->Unlock(a,b,c,d) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + + +#if DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSoundCaptureBuffer8 +// + +DEFINE_GUID(IID_IDirectSoundCaptureBuffer8, 0x990df4, 0xdbb, 0x4872, 0x83, 0x3e, 0x6d, 0x30, 0x3e, 0x80, 0xae, 0xb6); + +#undef INTERFACE +#define INTERFACE IDirectSoundCaptureBuffer8 + +DECLARE_INTERFACE_(IDirectSoundCaptureBuffer8, IDirectSoundCaptureBuffer) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundCaptureBuffer methods + STDMETHOD(GetCaps) (THIS_ LPDSCBCAPS pDSCBCaps) PURE; + STDMETHOD(GetCurrentPosition) (THIS_ LPDWORD pdwCapturePosition, LPDWORD pdwReadPosition) PURE; + STDMETHOD(GetFormat) (THIS_ LPWAVEFORMATEX pwfxFormat, DWORD dwSizeAllocated, LPDWORD pdwSizeWritten) PURE; + STDMETHOD(GetStatus) (THIS_ LPDWORD pdwStatus) PURE; + STDMETHOD(Initialize) (THIS_ LPDIRECTSOUNDCAPTURE pDirectSoundCapture, LPCDSCBUFFERDESC pcDSCBufferDesc) PURE; + STDMETHOD(Lock) (THIS_ DWORD dwOffset, DWORD dwBytes, LPVOID *ppvAudioPtr1, LPDWORD pdwAudioBytes1, + LPVOID *ppvAudioPtr2, LPDWORD pdwAudioBytes2, DWORD dwFlags) PURE; + STDMETHOD(Start) (THIS_ DWORD dwFlags) PURE; + STDMETHOD(Stop) (THIS) PURE; + STDMETHOD(Unlock) (THIS_ LPVOID pvAudioPtr1, DWORD dwAudioBytes1, LPVOID pvAudioPtr2, DWORD dwAudioBytes2) PURE; + + // IDirectSoundCaptureBuffer8 methods + STDMETHOD(GetObjectInPath) (THIS_ REFGUID rguidObject, DWORD dwIndex, REFGUID rguidInterface, LPVOID *ppObject) PURE; + STDMETHOD(GetFXStatus) (DWORD dwFXCount, LPDWORD pdwFXStatus) PURE; +}; + +#define IDirectSoundCaptureBuffer8_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundCaptureBuffer8_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundCaptureBuffer8_Release(p) IUnknown_Release(p) + +#define IDirectSoundCaptureBuffer8_GetCaps(p,a) IDirectSoundCaptureBuffer_GetCaps(p,a) +#define IDirectSoundCaptureBuffer8_GetCurrentPosition(p,a,b) IDirectSoundCaptureBuffer_GetCurrentPosition(p,a,b) +#define IDirectSoundCaptureBuffer8_GetFormat(p,a,b,c) IDirectSoundCaptureBuffer_GetFormat(p,a,b,c) +#define IDirectSoundCaptureBuffer8_GetStatus(p,a) IDirectSoundCaptureBuffer_GetStatus(p,a) +#define IDirectSoundCaptureBuffer8_Initialize(p,a,b) IDirectSoundCaptureBuffer_Initialize(p,a,b) +#define IDirectSoundCaptureBuffer8_Lock(p,a,b,c,d,e,f,g) IDirectSoundCaptureBuffer_Lock(p,a,b,c,d,e,f,g) +#define IDirectSoundCaptureBuffer8_Start(p,a) IDirectSoundCaptureBuffer_Start(p,a) +#define IDirectSoundCaptureBuffer8_Stop(p) IDirectSoundCaptureBuffer_Stop(p)) +#define IDirectSoundCaptureBuffer8_Unlock(p,a,b,c,d) IDirectSoundCaptureBuffer_Unlock(p,a,b,c,d) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureBuffer8_GetObjectInPath(p,a,b,c,d) (p)->lpVtbl->GetObjectInPath(p,a,b,c,d) +#define IDirectSoundCaptureBuffer8_GetFXStatus(p,a,b) (p)->lpVtbl->GetFXStatus(p,a,b) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureBuffer8_GetObjectInPath(p,a,b,c,d) (p)->GetObjectInPath(a,b,c,d) +#define IDirectSoundCaptureBuffer8_GetFXStatus(p,a,b) (p)->GetFXStatus(a,b) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSoundNotify +// + +DEFINE_GUID(IID_IDirectSoundNotify, 0xb0210783, 0x89cd, 0x11d0, 0xaf, 0x8, 0x0, 0xa0, 0xc9, 0x25, 0xcd, 0x16); + +#undef INTERFACE +#define INTERFACE IDirectSoundNotify + +DECLARE_INTERFACE_(IDirectSoundNotify, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundNotify methods + STDMETHOD(SetNotificationPositions) (THIS_ DWORD dwPositionNotifies, LPCDSBPOSITIONNOTIFY pcPositionNotifies) PURE; +}; + +#define IDirectSoundNotify_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundNotify_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundNotify_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundNotify_SetNotificationPositions(p,a,b) (p)->lpVtbl->SetNotificationPositions(p,a,b) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundNotify_SetNotificationPositions(p,a,b) (p)->SetNotificationPositions(a,b) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IKsPropertySet +// + +#ifndef _IKsPropertySet_ +#define _IKsPropertySet_ + +#ifdef __cplusplus +// 'struct' not 'class' per the way DECLARE_INTERFACE_ is defined +struct IKsPropertySet; +#endif // __cplusplus + +typedef struct IKsPropertySet *LPKSPROPERTYSET; + +#define KSPROPERTY_SUPPORT_GET 0x00000001 +#define KSPROPERTY_SUPPORT_SET 0x00000002 + +DEFINE_GUID(IID_IKsPropertySet, 0x31efac30, 0x515c, 0x11d0, 0xa9, 0xaa, 0x00, 0xaa, 0x00, 0x61, 0xbe, 0x93); + +#undef INTERFACE +#define INTERFACE IKsPropertySet + +DECLARE_INTERFACE_(IKsPropertySet, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IKsPropertySet methods + STDMETHOD(Get) (THIS_ REFGUID rguidPropSet, ULONG ulId, LPVOID pInstanceData, ULONG ulInstanceLength, + LPVOID pPropertyData, ULONG ulDataLength, PULONG pulBytesReturned) PURE; + STDMETHOD(Set) (THIS_ REFGUID rguidPropSet, ULONG ulId, LPVOID pInstanceData, ULONG ulInstanceLength, + LPVOID pPropertyData, ULONG ulDataLength) PURE; + STDMETHOD(QuerySupport) (THIS_ REFGUID rguidPropSet, ULONG ulId, PULONG pulTypeSupport) PURE; +}; + +#define IKsPropertySet_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IKsPropertySet_AddRef(p) IUnknown_AddRef(p) +#define IKsPropertySet_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IKsPropertySet_Get(p,a,b,c,d,e,f,g) (p)->lpVtbl->Get(p,a,b,c,d,e,f,g) +#define IKsPropertySet_Set(p,a,b,c,d,e,f) (p)->lpVtbl->Set(p,a,b,c,d,e,f) +#define IKsPropertySet_QuerySupport(p,a,b,c) (p)->lpVtbl->QuerySupport(p,a,b,c) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IKsPropertySet_Get(p,a,b,c,d,e,f,g) (p)->Get(a,b,c,d,e,f,g) +#define IKsPropertySet_Set(p,a,b,c,d,e,f) (p)->Set(a,b,c,d,e,f) +#define IKsPropertySet_QuerySupport(p,a,b,c) (p)->QuerySupport(a,b,c) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#endif // _IKsPropertySet_ + +#if DIRECTSOUND_VERSION >= 0x0800 + +// +// IDirectSoundFXGargle +// + +DEFINE_GUID(IID_IDirectSoundFXGargle, 0xd616f352, 0xd622, 0x11ce, 0xaa, 0xc5, 0x00, 0x20, 0xaf, 0x0b, 0x99, 0xa3); + +typedef struct _DSFXGargle +{ + DWORD dwRateHz; // Rate of modulation in hz + DWORD dwWaveShape; // DSFXGARGLE_WAVE_xxx +} DSFXGargle, *LPDSFXGargle; + +#define DSFXGARGLE_WAVE_TRIANGLE 0 +#define DSFXGARGLE_WAVE_SQUARE 1 + +typedef const DSFXGargle *LPCDSFXGargle; + +#define DSFXGARGLE_RATEHZ_MIN 1 +#define DSFXGARGLE_RATEHZ_MAX 1000 + +#undef INTERFACE +#define INTERFACE IDirectSoundFXGargle + +DECLARE_INTERFACE_(IDirectSoundFXGargle, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXGargle methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXGargle pcDsFxGargle) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXGargle pDsFxGargle) PURE; +}; + +#define IDirectSoundFXGargle_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXGargle_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXGargle_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXGargle_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXGargle_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXGargle_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXGargle_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXChorus +// + +DEFINE_GUID(IID_IDirectSoundFXChorus, 0x880842e3, 0x145f, 0x43e6, 0xa9, 0x34, 0xa7, 0x18, 0x06, 0xe5, 0x05, 0x47); + +typedef struct _DSFXChorus +{ + FLOAT fWetDryMix; + FLOAT fDepth; + FLOAT fFeedback; + FLOAT fFrequency; + LONG lWaveform; // LFO shape; DSFXCHORUS_WAVE_xxx + FLOAT fDelay; + LONG lPhase; +} DSFXChorus, *LPDSFXChorus; + +typedef const DSFXChorus *LPCDSFXChorus; + +#define DSFXCHORUS_WAVE_TRIANGLE 0 +#define DSFXCHORUS_WAVE_SIN 1 + +#define DSFXCHORUS_WETDRYMIX_MIN 0.0f +#define DSFXCHORUS_WETDRYMIX_MAX 100.0f +#define DSFXCHORUS_DEPTH_MIN 0.0f +#define DSFXCHORUS_DEPTH_MAX 100.0f +#define DSFXCHORUS_FEEDBACK_MIN -99.0f +#define DSFXCHORUS_FEEDBACK_MAX 99.0f +#define DSFXCHORUS_FREQUENCY_MIN 0.0f +#define DSFXCHORUS_FREQUENCY_MAX 10.0f +#define DSFXCHORUS_DELAY_MIN 0.0f +#define DSFXCHORUS_DELAY_MAX 20.0f +#define DSFXCHORUS_PHASE_MIN 0 +#define DSFXCHORUS_PHASE_MAX 4 + +#define DSFXCHORUS_PHASE_NEG_180 0 +#define DSFXCHORUS_PHASE_NEG_90 1 +#define DSFXCHORUS_PHASE_ZERO 2 +#define DSFXCHORUS_PHASE_90 3 +#define DSFXCHORUS_PHASE_180 4 + +#undef INTERFACE +#define INTERFACE IDirectSoundFXChorus + +DECLARE_INTERFACE_(IDirectSoundFXChorus, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXChorus methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXChorus pcDsFxChorus) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXChorus pDsFxChorus) PURE; +}; + +#define IDirectSoundFXChorus_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXChorus_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXChorus_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXChorus_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXChorus_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXChorus_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXChorus_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXFlanger +// + +DEFINE_GUID(IID_IDirectSoundFXFlanger, 0x903e9878, 0x2c92, 0x4072, 0x9b, 0x2c, 0xea, 0x68, 0xf5, 0x39, 0x67, 0x83); + +typedef struct _DSFXFlanger +{ + FLOAT fWetDryMix; + FLOAT fDepth; + FLOAT fFeedback; + FLOAT fFrequency; + LONG lWaveform; + FLOAT fDelay; + LONG lPhase; +} DSFXFlanger, *LPDSFXFlanger; + +typedef const DSFXFlanger *LPCDSFXFlanger; + +#define DSFXFLANGER_WAVE_TRIANGLE 0 +#define DSFXFLANGER_WAVE_SIN 1 + +#define DSFXFLANGER_WETDRYMIX_MIN 0.0f +#define DSFXFLANGER_WETDRYMIX_MAX 100.0f +#define DSFXFLANGER_FREQUENCY_MIN 0.0f +#define DSFXFLANGER_FREQUENCY_MAX 10.0f +#define DSFXFLANGER_DEPTH_MIN 0.0f +#define DSFXFLANGER_DEPTH_MAX 100.0f +#define DSFXFLANGER_PHASE_MIN 0 +#define DSFXFLANGER_PHASE_MAX 4 +#define DSFXFLANGER_FEEDBACK_MIN -99.0f +#define DSFXFLANGER_FEEDBACK_MAX 99.0f +#define DSFXFLANGER_DELAY_MIN 0.0f +#define DSFXFLANGER_DELAY_MAX 4.0f + +#define DSFXFLANGER_PHASE_NEG_180 0 +#define DSFXFLANGER_PHASE_NEG_90 1 +#define DSFXFLANGER_PHASE_ZERO 2 +#define DSFXFLANGER_PHASE_90 3 +#define DSFXFLANGER_PHASE_180 4 + +#undef INTERFACE +#define INTERFACE IDirectSoundFXFlanger + +DECLARE_INTERFACE_(IDirectSoundFXFlanger, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXFlanger methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXFlanger pcDsFxFlanger) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXFlanger pDsFxFlanger) PURE; +}; + +#define IDirectSoundFXFlanger_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXFlanger_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXFlanger_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXFlanger_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXFlanger_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXFlanger_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXFlanger_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXEcho +// + +DEFINE_GUID(IID_IDirectSoundFXEcho, 0x8bd28edf, 0x50db, 0x4e92, 0xa2, 0xbd, 0x44, 0x54, 0x88, 0xd1, 0xed, 0x42); + +typedef struct _DSFXEcho +{ + FLOAT fWetDryMix; + FLOAT fFeedback; + FLOAT fLeftDelay; + FLOAT fRightDelay; + LONG lPanDelay; +} DSFXEcho, *LPDSFXEcho; + +typedef const DSFXEcho *LPCDSFXEcho; + +#define DSFXECHO_WETDRYMIX_MIN 0.0f +#define DSFXECHO_WETDRYMIX_MAX 100.0f +#define DSFXECHO_FEEDBACK_MIN 0.0f +#define DSFXECHO_FEEDBACK_MAX 100.0f +#define DSFXECHO_LEFTDELAY_MIN 1.0f +#define DSFXECHO_LEFTDELAY_MAX 2000.0f +#define DSFXECHO_RIGHTDELAY_MIN 1.0f +#define DSFXECHO_RIGHTDELAY_MAX 2000.0f +#define DSFXECHO_PANDELAY_MIN 0 +#define DSFXECHO_PANDELAY_MAX 1 + +#undef INTERFACE +#define INTERFACE IDirectSoundFXEcho + +DECLARE_INTERFACE_(IDirectSoundFXEcho, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXEcho methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXEcho pcDsFxEcho) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXEcho pDsFxEcho) PURE; +}; + +#define IDirectSoundFXEcho_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXEcho_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXEcho_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXEcho_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXEcho_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXEcho_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXEcho_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXDistortion +// + +DEFINE_GUID(IID_IDirectSoundFXDistortion, 0x8ecf4326, 0x455f, 0x4d8b, 0xbd, 0xa9, 0x8d, 0x5d, 0x3e, 0x9e, 0x3e, 0x0b); + +typedef struct _DSFXDistortion +{ + FLOAT fGain; + FLOAT fEdge; + FLOAT fPostEQCenterFrequency; + FLOAT fPostEQBandwidth; + FLOAT fPreLowpassCutoff; +} DSFXDistortion, *LPDSFXDistortion; + +typedef const DSFXDistortion *LPCDSFXDistortion; + +#define DSFXDISTORTION_GAIN_MIN -60.0f +#define DSFXDISTORTION_GAIN_MAX 0.0f +#define DSFXDISTORTION_EDGE_MIN 0.0f +#define DSFXDISTORTION_EDGE_MAX 100.0f +#define DSFXDISTORTION_POSTEQCENTERFREQUENCY_MIN 100.0f +#define DSFXDISTORTION_POSTEQCENTERFREQUENCY_MAX 8000.0f +#define DSFXDISTORTION_POSTEQBANDWIDTH_MIN 100.0f +#define DSFXDISTORTION_POSTEQBANDWIDTH_MAX 8000.0f +#define DSFXDISTORTION_PRELOWPASSCUTOFF_MIN 100.0f +#define DSFXDISTORTION_PRELOWPASSCUTOFF_MAX 8000.0f + +#undef INTERFACE +#define INTERFACE IDirectSoundFXDistortion + +DECLARE_INTERFACE_(IDirectSoundFXDistortion, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXDistortion methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXDistortion pcDsFxDistortion) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXDistortion pDsFxDistortion) PURE; +}; + +#define IDirectSoundFXDistortion_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXDistortion_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXDistortion_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXDistortion_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXDistortion_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXDistortion_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXDistortion_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXCompressor +// + +DEFINE_GUID(IID_IDirectSoundFXCompressor, 0x4bbd1154, 0x62f6, 0x4e2c, 0xa1, 0x5c, 0xd3, 0xb6, 0xc4, 0x17, 0xf7, 0xa0); + +typedef struct _DSFXCompressor +{ + FLOAT fGain; + FLOAT fAttack; + FLOAT fRelease; + FLOAT fThreshold; + FLOAT fRatio; + FLOAT fPredelay; +} DSFXCompressor, *LPDSFXCompressor; + +typedef const DSFXCompressor *LPCDSFXCompressor; + +#define DSFXCOMPRESSOR_GAIN_MIN -60.0f +#define DSFXCOMPRESSOR_GAIN_MAX 60.0f +#define DSFXCOMPRESSOR_ATTACK_MIN 0.01f +#define DSFXCOMPRESSOR_ATTACK_MAX 500.0f +#define DSFXCOMPRESSOR_RELEASE_MIN 50.0f +#define DSFXCOMPRESSOR_RELEASE_MAX 3000.0f +#define DSFXCOMPRESSOR_THRESHOLD_MIN -60.0f +#define DSFXCOMPRESSOR_THRESHOLD_MAX 0.0f +#define DSFXCOMPRESSOR_RATIO_MIN 1.0f +#define DSFXCOMPRESSOR_RATIO_MAX 100.0f +#define DSFXCOMPRESSOR_PREDELAY_MIN 0.0f +#define DSFXCOMPRESSOR_PREDELAY_MAX 4.0f + +#undef INTERFACE +#define INTERFACE IDirectSoundFXCompressor + +DECLARE_INTERFACE_(IDirectSoundFXCompressor, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXCompressor methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXCompressor pcDsFxCompressor) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXCompressor pDsFxCompressor) PURE; +}; + +#define IDirectSoundFXCompressor_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXCompressor_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXCompressor_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXCompressor_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXCompressor_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXCompressor_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXCompressor_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXParamEq +// + +DEFINE_GUID(IID_IDirectSoundFXParamEq, 0xc03ca9fe, 0xfe90, 0x4204, 0x80, 0x78, 0x82, 0x33, 0x4c, 0xd1, 0x77, 0xda); + +typedef struct _DSFXParamEq +{ + FLOAT fCenter; + FLOAT fBandwidth; + FLOAT fGain; +} DSFXParamEq, *LPDSFXParamEq; + +typedef const DSFXParamEq *LPCDSFXParamEq; + +#define DSFXPARAMEQ_CENTER_MIN 80.0f +#define DSFXPARAMEQ_CENTER_MAX 16000.0f +#define DSFXPARAMEQ_BANDWIDTH_MIN 1.0f +#define DSFXPARAMEQ_BANDWIDTH_MAX 36.0f +#define DSFXPARAMEQ_GAIN_MIN -15.0f +#define DSFXPARAMEQ_GAIN_MAX 15.0f + +#undef INTERFACE +#define INTERFACE IDirectSoundFXParamEq + +DECLARE_INTERFACE_(IDirectSoundFXParamEq, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXParamEq methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXParamEq pcDsFxParamEq) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXParamEq pDsFxParamEq) PURE; +}; + +#define IDirectSoundFXParamEq_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXParamEq_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXParamEq_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXParamEq_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXParamEq_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXParamEq_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXParamEq_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXI3DL2Reverb +// + +DEFINE_GUID(IID_IDirectSoundFXI3DL2Reverb, 0x4b166a6a, 0x0d66, 0x43f3, 0x80, 0xe3, 0xee, 0x62, 0x80, 0xde, 0xe1, 0xa4); + +typedef struct _DSFXI3DL2Reverb +{ + LONG lRoom; // [-10000, 0] default: -1000 mB + LONG lRoomHF; // [-10000, 0] default: 0 mB + FLOAT flRoomRolloffFactor; // [0.0, 10.0] default: 0.0 + FLOAT flDecayTime; // [0.1, 20.0] default: 1.49s + FLOAT flDecayHFRatio; // [0.1, 2.0] default: 0.83 + LONG lReflections; // [-10000, 1000] default: -2602 mB + FLOAT flReflectionsDelay; // [0.0, 0.3] default: 0.007 s + LONG lReverb; // [-10000, 2000] default: 200 mB + FLOAT flReverbDelay; // [0.0, 0.1] default: 0.011 s + FLOAT flDiffusion; // [0.0, 100.0] default: 100.0 % + FLOAT flDensity; // [0.0, 100.0] default: 100.0 % + FLOAT flHFReference; // [20.0, 20000.0] default: 5000.0 Hz +} DSFXI3DL2Reverb, *LPDSFXI3DL2Reverb; + +typedef const DSFXI3DL2Reverb *LPCDSFXI3DL2Reverb; + +#define DSFX_I3DL2REVERB_ROOM_MIN (-10000) +#define DSFX_I3DL2REVERB_ROOM_MAX 0 +#define DSFX_I3DL2REVERB_ROOM_DEFAULT (-1000) + +#define DSFX_I3DL2REVERB_ROOMHF_MIN (-10000) +#define DSFX_I3DL2REVERB_ROOMHF_MAX 0 +#define DSFX_I3DL2REVERB_ROOMHF_DEFAULT (-100) + +#define DSFX_I3DL2REVERB_ROOMROLLOFFFACTOR_MIN 0.0f +#define DSFX_I3DL2REVERB_ROOMROLLOFFFACTOR_MAX 10.0f +#define DSFX_I3DL2REVERB_ROOMROLLOFFFACTOR_DEFAULT 0.0f + +#define DSFX_I3DL2REVERB_DECAYTIME_MIN 0.1f +#define DSFX_I3DL2REVERB_DECAYTIME_MAX 20.0f +#define DSFX_I3DL2REVERB_DECAYTIME_DEFAULT 1.49f + +#define DSFX_I3DL2REVERB_DECAYHFRATIO_MIN 0.1f +#define DSFX_I3DL2REVERB_DECAYHFRATIO_MAX 2.0f +#define DSFX_I3DL2REVERB_DECAYHFRATIO_DEFAULT 0.83f + +#define DSFX_I3DL2REVERB_REFLECTIONS_MIN (-10000) +#define DSFX_I3DL2REVERB_REFLECTIONS_MAX 1000 +#define DSFX_I3DL2REVERB_REFLECTIONS_DEFAULT (-2602) + +#define DSFX_I3DL2REVERB_REFLECTIONSDELAY_MIN 0.0f +#define DSFX_I3DL2REVERB_REFLECTIONSDELAY_MAX 0.3f +#define DSFX_I3DL2REVERB_REFLECTIONSDELAY_DEFAULT 0.007f + +#define DSFX_I3DL2REVERB_REVERB_MIN (-10000) +#define DSFX_I3DL2REVERB_REVERB_MAX 2000 +#define DSFX_I3DL2REVERB_REVERB_DEFAULT (200) + +#define DSFX_I3DL2REVERB_REVERBDELAY_MIN 0.0f +#define DSFX_I3DL2REVERB_REVERBDELAY_MAX 0.1f +#define DSFX_I3DL2REVERB_REVERBDELAY_DEFAULT 0.011f + +#define DSFX_I3DL2REVERB_DIFFUSION_MIN 0.0f +#define DSFX_I3DL2REVERB_DIFFUSION_MAX 100.0f +#define DSFX_I3DL2REVERB_DIFFUSION_DEFAULT 100.0f + +#define DSFX_I3DL2REVERB_DENSITY_MIN 0.0f +#define DSFX_I3DL2REVERB_DENSITY_MAX 100.0f +#define DSFX_I3DL2REVERB_DENSITY_DEFAULT 100.0f + +#define DSFX_I3DL2REVERB_HFREFERENCE_MIN 20.0f +#define DSFX_I3DL2REVERB_HFREFERENCE_MAX 20000.0f +#define DSFX_I3DL2REVERB_HFREFERENCE_DEFAULT 5000.0f + +#define DSFX_I3DL2REVERB_QUALITY_MIN 0 +#define DSFX_I3DL2REVERB_QUALITY_MAX 3 +#define DSFX_I3DL2REVERB_QUALITY_DEFAULT 2 + +#undef INTERFACE +#define INTERFACE IDirectSoundFXI3DL2Reverb + +DECLARE_INTERFACE_(IDirectSoundFXI3DL2Reverb, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXI3DL2Reverb methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXI3DL2Reverb pcDsFxI3DL2Reverb) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXI3DL2Reverb pDsFxI3DL2Reverb) PURE; + STDMETHOD(SetPreset) (THIS_ DWORD dwPreset) PURE; + STDMETHOD(GetPreset) (THIS_ LPDWORD pdwPreset) PURE; + STDMETHOD(SetQuality) (THIS_ LONG lQuality) PURE; + STDMETHOD(GetQuality) (THIS_ LONG *plQuality) PURE; +}; + +#define IDirectSoundFXI3DL2Reverb_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXI3DL2Reverb_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXI3DL2Reverb_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXI3DL2Reverb_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXI3DL2Reverb_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#define IDirectSoundFXI3DL2Reverb_SetPreset(p,a) (p)->lpVtbl->SetPreset(p,a) +#define IDirectSoundFXI3DL2Reverb_GetPreset(p,a) (p)->lpVtbl->GetPreset(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXI3DL2Reverb_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXI3DL2Reverb_GetAllParameters(p,a) (p)->GetAllParameters(a) +#define IDirectSoundFXI3DL2Reverb_SetPreset(p,a) (p)->SetPreset(a) +#define IDirectSoundFXI3DL2Reverb_GetPreset(p,a) (p)->GetPreset(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundFXWavesReverb +// + +DEFINE_GUID(IID_IDirectSoundFXWavesReverb,0x46858c3a,0x0dc6,0x45e3,0xb7,0x60,0xd4,0xee,0xf1,0x6c,0xb3,0x25); + +typedef struct _DSFXWavesReverb +{ + FLOAT fInGain; // [-96.0,0.0] default: 0.0 dB + FLOAT fReverbMix; // [-96.0,0.0] default: 0.0 db + FLOAT fReverbTime; // [0.001,3000.0] default: 1000.0 ms + FLOAT fHighFreqRTRatio; // [0.001,0.999] default: 0.001 +} DSFXWavesReverb, *LPDSFXWavesReverb; + +typedef const DSFXWavesReverb *LPCDSFXWavesReverb; + +#define DSFX_WAVESREVERB_INGAIN_MIN -96.0f +#define DSFX_WAVESREVERB_INGAIN_MAX 0.0f +#define DSFX_WAVESREVERB_INGAIN_DEFAULT 0.0f +#define DSFX_WAVESREVERB_REVERBMIX_MIN -96.0f +#define DSFX_WAVESREVERB_REVERBMIX_MAX 0.0f +#define DSFX_WAVESREVERB_REVERBMIX_DEFAULT 0.0f +#define DSFX_WAVESREVERB_REVERBTIME_MIN 0.001f +#define DSFX_WAVESREVERB_REVERBTIME_MAX 3000.0f +#define DSFX_WAVESREVERB_REVERBTIME_DEFAULT 1000.0f +#define DSFX_WAVESREVERB_HIGHFREQRTRATIO_MIN 0.001f +#define DSFX_WAVESREVERB_HIGHFREQRTRATIO_MAX 0.999f +#define DSFX_WAVESREVERB_HIGHFREQRTRATIO_DEFAULT 0.001f + +#undef INTERFACE +#define INTERFACE IDirectSoundFXWavesReverb + +DECLARE_INTERFACE_(IDirectSoundFXWavesReverb, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFXWavesReverb methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSFXWavesReverb pcDsFxWavesReverb) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSFXWavesReverb pDsFxWavesReverb) PURE; +}; + +#define IDirectSoundFXWavesReverb_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFXWavesReverb_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFXWavesReverb_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXWavesReverb_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundFXWavesReverb_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFXWavesReverb_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundFXWavesReverb_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +// +// IDirectSoundCaptureFXAec +// + +DEFINE_GUID(IID_IDirectSoundCaptureFXAec, 0xad74143d, 0x903d, 0x4ab7, 0x80, 0x66, 0x28, 0xd3, 0x63, 0x03, 0x6d, 0x65); + +typedef struct _DSCFXAec +{ + BOOL fEnable; + BOOL fNoiseFill; + DWORD dwMode; +} DSCFXAec, *LPDSCFXAec; + +typedef const DSCFXAec *LPCDSCFXAec; + +// These match the AEC_MODE_* constants in the DDK's ksmedia.h file +#define DSCFX_AEC_MODE_PASS_THROUGH 0x0 +#define DSCFX_AEC_MODE_HALF_DUPLEX 0x1 +#define DSCFX_AEC_MODE_FULL_DUPLEX 0x2 + +// These match the AEC_STATUS_* constants in ksmedia.h +#define DSCFX_AEC_STATUS_HISTORY_UNINITIALIZED 0x0 +#define DSCFX_AEC_STATUS_HISTORY_CONTINUOUSLY_CONVERGED 0x1 +#define DSCFX_AEC_STATUS_HISTORY_PREVIOUSLY_DIVERGED 0x2 +#define DSCFX_AEC_STATUS_CURRENTLY_CONVERGED 0x8 + +#undef INTERFACE +#define INTERFACE IDirectSoundCaptureFXAec + +DECLARE_INTERFACE_(IDirectSoundCaptureFXAec, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundCaptureFXAec methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSCFXAec pDscFxAec) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSCFXAec pDscFxAec) PURE; + STDMETHOD(GetStatus) (THIS_ PDWORD pdwStatus) PURE; + STDMETHOD(Reset) (THIS) PURE; +}; + +#define IDirectSoundCaptureFXAec_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundCaptureFXAec_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundCaptureFXAec_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureFXAec_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundCaptureFXAec_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureFXAec_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundCaptureFXAec_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + + +// +// IDirectSoundCaptureFXNoiseSuppress +// + +DEFINE_GUID(IID_IDirectSoundCaptureFXNoiseSuppress, 0xed311e41, 0xfbae, 0x4175, 0x96, 0x25, 0xcd, 0x8, 0x54, 0xf6, 0x93, 0xca); + +typedef struct _DSCFXNoiseSuppress +{ + BOOL fEnable; +} DSCFXNoiseSuppress, *LPDSCFXNoiseSuppress; + +typedef const DSCFXNoiseSuppress *LPCDSCFXNoiseSuppress; + +#undef INTERFACE +#define INTERFACE IDirectSoundCaptureFXNoiseSuppress + +DECLARE_INTERFACE_(IDirectSoundCaptureFXNoiseSuppress, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundCaptureFXNoiseSuppress methods + STDMETHOD(SetAllParameters) (THIS_ LPCDSCFXNoiseSuppress pcDscFxNoiseSuppress) PURE; + STDMETHOD(GetAllParameters) (THIS_ LPDSCFXNoiseSuppress pDscFxNoiseSuppress) PURE; + STDMETHOD(Reset) (THIS) PURE; +}; + +#define IDirectSoundCaptureFXNoiseSuppress_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundCaptureFXNoiseSuppress_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundCaptureFXNoiseSuppress_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureFXNoiseSuppress_SetAllParameters(p,a) (p)->lpVtbl->SetAllParameters(p,a) +#define IDirectSoundCaptureFXNoiseSuppress_GetAllParameters(p,a) (p)->lpVtbl->GetAllParameters(p,a) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundCaptureFXNoiseSuppress_SetAllParameters(p,a) (p)->SetAllParameters(a) +#define IDirectSoundCaptureFXNoiseSuppress_GetAllParameters(p,a) (p)->GetAllParameters(a) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + + +// +// IDirectSoundFullDuplex +// + +#ifndef _IDirectSoundFullDuplex_ +#define _IDirectSoundFullDuplex_ + +#ifdef __cplusplus +// 'struct' not 'class' per the way DECLARE_INTERFACE_ is defined +struct IDirectSoundFullDuplex; +#endif // __cplusplus + +typedef struct IDirectSoundFullDuplex *LPDIRECTSOUNDFULLDUPLEX; + +DEFINE_GUID(IID_IDirectSoundFullDuplex, 0xedcb4c7a, 0xdaab, 0x4216, 0xa4, 0x2e, 0x6c, 0x50, 0x59, 0x6d, 0xdc, 0x1d); + +#undef INTERFACE +#define INTERFACE IDirectSoundFullDuplex + +DECLARE_INTERFACE_(IDirectSoundFullDuplex, IUnknown) +{ + // IUnknown methods + STDMETHOD(QueryInterface) (THIS_ REFIID, LPVOID *) PURE; + STDMETHOD_(ULONG,AddRef) (THIS) PURE; + STDMETHOD_(ULONG,Release) (THIS) PURE; + + // IDirectSoundFullDuplex methods + STDMETHOD(Initialize) (THIS_ LPCGUID pCaptureGuid, LPCGUID pRenderGuid, LPCDSCBUFFERDESC lpDscBufferDesc, LPCDSBUFFERDESC lpDsBufferDesc, HWND hWnd, DWORD dwLevel, LPLPDIRECTSOUNDCAPTUREBUFFER8 lplpDirectSoundCaptureBuffer8, LPLPDIRECTSOUNDBUFFER8 lplpDirectSoundBuffer8) PURE; +}; + +#define IDirectSoundFullDuplex_QueryInterface(p,a,b) IUnknown_QueryInterface(p,a,b) +#define IDirectSoundFullDuplex_AddRef(p) IUnknown_AddRef(p) +#define IDirectSoundFullDuplex_Release(p) IUnknown_Release(p) + +#if !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFullDuplex_Initialize(p,a,b,c,d,e,f,g,h) (p)->lpVtbl->Initialize(p,a,b,c,d,e,f,g,h) +#else // !defined(__cplusplus) || defined(CINTERFACE) +#define IDirectSoundFullDuplex_Initialize(p,a,b,c,d,e,f,g,h) (p)->Initialize(a,b,c,d,e,f,g,h) +#endif // !defined(__cplusplus) || defined(CINTERFACE) + +#endif // _IDirectSoundFullDuplex_ + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +// +// Return Codes +// + +// The function completed successfully +#define DS_OK S_OK + +// The call succeeded, but we had to substitute the 3D algorithm +#define DS_NO_VIRTUALIZATION MAKE_HRESULT(0, _FACDS, 10) + +// The call failed because resources (such as a priority level) +// were already being used by another caller +#define DSERR_ALLOCATED MAKE_DSHRESULT(10) + +// The control (vol, pan, etc.) requested by the caller is not available +#define DSERR_CONTROLUNAVAIL MAKE_DSHRESULT(30) + +// An invalid parameter was passed to the returning function +#define DSERR_INVALIDPARAM E_INVALIDARG + +// This call is not valid for the current state of this object +#define DSERR_INVALIDCALL MAKE_DSHRESULT(50) + +// An undetermined error occurred inside the DirectSound subsystem +#define DSERR_GENERIC E_FAIL + +// The caller does not have the priority level required for the function to +// succeed +#define DSERR_PRIOLEVELNEEDED MAKE_DSHRESULT(70) + +// Not enough free memory is available to complete the operation +#define DSERR_OUTOFMEMORY E_OUTOFMEMORY + +// The specified WAVE format is not supported +#define DSERR_BADFORMAT MAKE_DSHRESULT(100) + +// The function called is not supported at this time +#define DSERR_UNSUPPORTED E_NOTIMPL + +// No sound driver is available for use +#define DSERR_NODRIVER MAKE_DSHRESULT(120) +// This object is already initialized +#define DSERR_ALREADYINITIALIZED MAKE_DSHRESULT(130) + +// This object does not support aggregation +#define DSERR_NOAGGREGATION CLASS_E_NOAGGREGATION + +// The buffer memory has been lost, and must be restored +#define DSERR_BUFFERLOST MAKE_DSHRESULT(150) + +// Another app has a higher priority level, preventing this call from +// succeeding +#define DSERR_OTHERAPPHASPRIO MAKE_DSHRESULT(160) + +// This object has not been initialized +#define DSERR_UNINITIALIZED MAKE_DSHRESULT(170) + +// The requested COM interface is not available +#define DSERR_NOINTERFACE E_NOINTERFACE + +// Access is denied +#define DSERR_ACCESSDENIED E_ACCESSDENIED + +// Tried to create a DSBCAPS_CTRLFX buffer shorter than DSBSIZE_FX_MIN milliseconds +#define DSERR_BUFFERTOOSMALL MAKE_DSHRESULT(180) + +// Attempt to use DirectSound 8 functionality on an older DirectSound object +#define DSERR_DS8_REQUIRED MAKE_DSHRESULT(190) + +// A circular loop of send effects was detected +#define DSERR_SENDLOOP MAKE_DSHRESULT(200) + +// The GUID specified in an audiopath file does not match a valid MIXIN buffer +#define DSERR_BADSENDBUFFERGUID MAKE_DSHRESULT(210) + +// The object requested was not found (numerically equal to DMUS_E_NOT_FOUND) +#define DSERR_OBJECTNOTFOUND MAKE_DSHRESULT(4449) + +// The effects requested could not be found on the system, or they were found +// but in the wrong order, or in the wrong hardware/software locations. +#define DSERR_FXUNAVAILABLE MAKE_DSHRESULT(220) + +// +// Flags +// + +#define DSCAPS_PRIMARYMONO 0x00000001 +#define DSCAPS_PRIMARYSTEREO 0x00000002 +#define DSCAPS_PRIMARY8BIT 0x00000004 +#define DSCAPS_PRIMARY16BIT 0x00000008 +#define DSCAPS_CONTINUOUSRATE 0x00000010 +#define DSCAPS_EMULDRIVER 0x00000020 +#define DSCAPS_CERTIFIED 0x00000040 +#define DSCAPS_SECONDARYMONO 0x00000100 +#define DSCAPS_SECONDARYSTEREO 0x00000200 +#define DSCAPS_SECONDARY8BIT 0x00000400 +#define DSCAPS_SECONDARY16BIT 0x00000800 + +#define DSSCL_NORMAL 0x00000001 +#define DSSCL_PRIORITY 0x00000002 +#define DSSCL_EXCLUSIVE 0x00000003 +#define DSSCL_WRITEPRIMARY 0x00000004 + +#define DSSPEAKER_DIRECTOUT 0x00000000 +#define DSSPEAKER_HEADPHONE 0x00000001 +#define DSSPEAKER_MONO 0x00000002 +#define DSSPEAKER_QUAD 0x00000003 +#define DSSPEAKER_STEREO 0x00000004 +#define DSSPEAKER_SURROUND 0x00000005 +#define DSSPEAKER_5POINT1 0x00000006 // obsolete 5.1 setting +#define DSSPEAKER_7POINT1 0x00000007 // obsolete 7.1 setting +#define DSSPEAKER_7POINT1_SURROUND 0x00000008 // correct 7.1 Home Theater setting +#define DSSPEAKER_7POINT1_WIDE DSSPEAKER_7POINT1 +#if (DIRECTSOUND_VERSION >= 0x1000) + #define DSSPEAKER_5POINT1_SURROUND 0x00000009 // correct 5.1 setting + #define DSSPEAKER_5POINT1_BACK DSSPEAKER_5POINT1 +#endif + +#define DSSPEAKER_GEOMETRY_MIN 0x00000005 // 5 degrees +#define DSSPEAKER_GEOMETRY_NARROW 0x0000000A // 10 degrees +#define DSSPEAKER_GEOMETRY_WIDE 0x00000014 // 20 degrees +#define DSSPEAKER_GEOMETRY_MAX 0x000000B4 // 180 degrees + +#define DSSPEAKER_COMBINED(c, g) ((DWORD)(((BYTE)(c)) | ((DWORD)((BYTE)(g))) << 16)) +#define DSSPEAKER_CONFIG(a) ((BYTE)(a)) +#define DSSPEAKER_GEOMETRY(a) ((BYTE)(((DWORD)(a) >> 16) & 0x00FF)) + +#define DSBCAPS_PRIMARYBUFFER 0x00000001 +#define DSBCAPS_STATIC 0x00000002 +#define DSBCAPS_LOCHARDWARE 0x00000004 +#define DSBCAPS_LOCSOFTWARE 0x00000008 +#define DSBCAPS_CTRL3D 0x00000010 +#define DSBCAPS_CTRLFREQUENCY 0x00000020 +#define DSBCAPS_CTRLPAN 0x00000040 +#define DSBCAPS_CTRLVOLUME 0x00000080 +#define DSBCAPS_CTRLPOSITIONNOTIFY 0x00000100 +#define DSBCAPS_CTRLFX 0x00000200 +#define DSBCAPS_STICKYFOCUS 0x00004000 +#define DSBCAPS_GLOBALFOCUS 0x00008000 +#define DSBCAPS_GETCURRENTPOSITION2 0x00010000 +#define DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000 +#define DSBCAPS_LOCDEFER 0x00040000 +#if (DIRECTSOUND_VERSION >= 0x1000) + // Force GetCurrentPosition() to return a buffer's true play position; + // unmodified by aids to enhance backward compatibility. + #define DSBCAPS_TRUEPLAYPOSITION 0x00080000 +#endif + +#define DSBPLAY_LOOPING 0x00000001 +#define DSBPLAY_LOCHARDWARE 0x00000002 +#define DSBPLAY_LOCSOFTWARE 0x00000004 +#define DSBPLAY_TERMINATEBY_TIME 0x00000008 +#define DSBPLAY_TERMINATEBY_DISTANCE 0x000000010 +#define DSBPLAY_TERMINATEBY_PRIORITY 0x000000020 + +#define DSBSTATUS_PLAYING 0x00000001 +#define DSBSTATUS_BUFFERLOST 0x00000002 +#define DSBSTATUS_LOOPING 0x00000004 +#define DSBSTATUS_LOCHARDWARE 0x00000008 +#define DSBSTATUS_LOCSOFTWARE 0x00000010 +#define DSBSTATUS_TERMINATED 0x00000020 + +#define DSBLOCK_FROMWRITECURSOR 0x00000001 +#define DSBLOCK_ENTIREBUFFER 0x00000002 + +#define DSBFREQUENCY_ORIGINAL 0 +#define DSBFREQUENCY_MIN 100 +#if DIRECTSOUND_VERSION >= 0x0900 +#define DSBFREQUENCY_MAX 200000 +#else +#define DSBFREQUENCY_MAX 100000 +#endif + +#define DSBPAN_LEFT -10000 +#define DSBPAN_CENTER 0 +#define DSBPAN_RIGHT 10000 + +#define DSBVOLUME_MIN -10000 +#define DSBVOLUME_MAX 0 + +#define DSBSIZE_MIN 4 +#define DSBSIZE_MAX 0x0FFFFFFF +#define DSBSIZE_FX_MIN 150 // NOTE: Milliseconds, not bytes + +#define DSBNOTIFICATIONS_MAX 100000UL + +#define DS3DMODE_NORMAL 0x00000000 +#define DS3DMODE_HEADRELATIVE 0x00000001 +#define DS3DMODE_DISABLE 0x00000002 + +#define DS3D_IMMEDIATE 0x00000000 +#define DS3D_DEFERRED 0x00000001 + +#define DS3D_MINDISTANCEFACTOR FLT_MIN +#define DS3D_MAXDISTANCEFACTOR FLT_MAX +#define DS3D_DEFAULTDISTANCEFACTOR 1.0f + +#define DS3D_MINROLLOFFFACTOR 0.0f +#define DS3D_MAXROLLOFFFACTOR 10.0f +#define DS3D_DEFAULTROLLOFFFACTOR 1.0f + +#define DS3D_MINDOPPLERFACTOR 0.0f +#define DS3D_MAXDOPPLERFACTOR 10.0f +#define DS3D_DEFAULTDOPPLERFACTOR 1.0f + +#define DS3D_DEFAULTMINDISTANCE 1.0f +#define DS3D_DEFAULTMAXDISTANCE 1000000000.0f + +#define DS3D_MINCONEANGLE 0 +#define DS3D_MAXCONEANGLE 360 +#define DS3D_DEFAULTCONEANGLE 360 + +#define DS3D_DEFAULTCONEOUTSIDEVOLUME DSBVOLUME_MAX + +// IDirectSoundCapture attributes + +#define DSCCAPS_EMULDRIVER DSCAPS_EMULDRIVER +#define DSCCAPS_CERTIFIED DSCAPS_CERTIFIED +#define DSCCAPS_MULTIPLECAPTURE 0x00000001 + +// IDirectSoundCaptureBuffer attributes + +#define DSCBCAPS_WAVEMAPPED 0x80000000 + +#if DIRECTSOUND_VERSION >= 0x0800 +#define DSCBCAPS_CTRLFX 0x00000200 +#endif + + +#define DSCBLOCK_ENTIREBUFFER 0x00000001 + +#define DSCBSTATUS_CAPTURING 0x00000001 +#define DSCBSTATUS_LOOPING 0x00000002 + +#define DSCBSTART_LOOPING 0x00000001 + +#define DSBPN_OFFSETSTOP 0xFFFFFFFF + +#define DS_CERTIFIED 0x00000000 +#define DS_UNCERTIFIED 0x00000001 + + +// +// Flags for the I3DL2 effects +// + +// +// I3DL2 Material Presets +// + +enum +{ + DSFX_I3DL2_MATERIAL_PRESET_SINGLEWINDOW, + DSFX_I3DL2_MATERIAL_PRESET_DOUBLEWINDOW, + DSFX_I3DL2_MATERIAL_PRESET_THINDOOR, + DSFX_I3DL2_MATERIAL_PRESET_THICKDOOR, + DSFX_I3DL2_MATERIAL_PRESET_WOODWALL, + DSFX_I3DL2_MATERIAL_PRESET_BRICKWALL, + DSFX_I3DL2_MATERIAL_PRESET_STONEWALL, + DSFX_I3DL2_MATERIAL_PRESET_CURTAIN +}; + +#define I3DL2_MATERIAL_PRESET_SINGLEWINDOW -2800,0.71f +#define I3DL2_MATERIAL_PRESET_DOUBLEWINDOW -5000,0.40f +#define I3DL2_MATERIAL_PRESET_THINDOOR -1800,0.66f +#define I3DL2_MATERIAL_PRESET_THICKDOOR -4400,0.64f +#define I3DL2_MATERIAL_PRESET_WOODWALL -4000,0.50f +#define I3DL2_MATERIAL_PRESET_BRICKWALL -5000,0.60f +#define I3DL2_MATERIAL_PRESET_STONEWALL -6000,0.68f +#define I3DL2_MATERIAL_PRESET_CURTAIN -1200,0.15f + +enum +{ + DSFX_I3DL2_ENVIRONMENT_PRESET_DEFAULT, + DSFX_I3DL2_ENVIRONMENT_PRESET_GENERIC, + DSFX_I3DL2_ENVIRONMENT_PRESET_PADDEDCELL, + DSFX_I3DL2_ENVIRONMENT_PRESET_ROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_BATHROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_LIVINGROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_STONEROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_AUDITORIUM, + DSFX_I3DL2_ENVIRONMENT_PRESET_CONCERTHALL, + DSFX_I3DL2_ENVIRONMENT_PRESET_CAVE, + DSFX_I3DL2_ENVIRONMENT_PRESET_ARENA, + DSFX_I3DL2_ENVIRONMENT_PRESET_HANGAR, + DSFX_I3DL2_ENVIRONMENT_PRESET_CARPETEDHALLWAY, + DSFX_I3DL2_ENVIRONMENT_PRESET_HALLWAY, + DSFX_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR, + DSFX_I3DL2_ENVIRONMENT_PRESET_ALLEY, + DSFX_I3DL2_ENVIRONMENT_PRESET_FOREST, + DSFX_I3DL2_ENVIRONMENT_PRESET_CITY, + DSFX_I3DL2_ENVIRONMENT_PRESET_MOUNTAINS, + DSFX_I3DL2_ENVIRONMENT_PRESET_QUARRY, + DSFX_I3DL2_ENVIRONMENT_PRESET_PLAIN, + DSFX_I3DL2_ENVIRONMENT_PRESET_PARKINGLOT, + DSFX_I3DL2_ENVIRONMENT_PRESET_SEWERPIPE, + DSFX_I3DL2_ENVIRONMENT_PRESET_UNDERWATER, + DSFX_I3DL2_ENVIRONMENT_PRESET_SMALLROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_MEDIUMROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_LARGEROOM, + DSFX_I3DL2_ENVIRONMENT_PRESET_MEDIUMHALL, + DSFX_I3DL2_ENVIRONMENT_PRESET_LARGEHALL, + DSFX_I3DL2_ENVIRONMENT_PRESET_PLATE +}; + +// +// I3DL2 Reverberation Presets Values +// + +#define I3DL2_ENVIRONMENT_PRESET_DEFAULT -1000, -100, 0.0f, 1.49f, 0.83f, -2602, 0.007f, 200, 0.011f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_GENERIC -1000, -100, 0.0f, 1.49f, 0.83f, -2602, 0.007f, 200, 0.011f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_PADDEDCELL -1000,-6000, 0.0f, 0.17f, 0.10f, -1204, 0.001f, 207, 0.002f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_ROOM -1000, -454, 0.0f, 0.40f, 0.83f, -1646, 0.002f, 53, 0.003f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_BATHROOM -1000,-1200, 0.0f, 1.49f, 0.54f, -370, 0.007f, 1030, 0.011f, 100.0f, 60.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_LIVINGROOM -1000,-6000, 0.0f, 0.50f, 0.10f, -1376, 0.003f, -1104, 0.004f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_STONEROOM -1000, -300, 0.0f, 2.31f, 0.64f, -711, 0.012f, 83, 0.017f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_AUDITORIUM -1000, -476, 0.0f, 4.32f, 0.59f, -789, 0.020f, -289, 0.030f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_CONCERTHALL -1000, -500, 0.0f, 3.92f, 0.70f, -1230, 0.020f, -2, 0.029f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_CAVE -1000, 0, 0.0f, 2.91f, 1.30f, -602, 0.015f, -302, 0.022f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_ARENA -1000, -698, 0.0f, 7.24f, 0.33f, -1166, 0.020f, 16, 0.030f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_HANGAR -1000,-1000, 0.0f,10.05f, 0.23f, -602, 0.020f, 198, 0.030f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_CARPETEDHALLWAY -1000,-4000, 0.0f, 0.30f, 0.10f, -1831, 0.002f, -1630, 0.030f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_HALLWAY -1000, -300, 0.0f, 1.49f, 0.59f, -1219, 0.007f, 441, 0.011f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR -1000, -237, 0.0f, 2.70f, 0.79f, -1214, 0.013f, 395, 0.020f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_ALLEY -1000, -270, 0.0f, 1.49f, 0.86f, -1204, 0.007f, -4, 0.011f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_FOREST -1000,-3300, 0.0f, 1.49f, 0.54f, -2560, 0.162f, -613, 0.088f, 79.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_CITY -1000, -800, 0.0f, 1.49f, 0.67f, -2273, 0.007f, -2217, 0.011f, 50.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_MOUNTAINS -1000,-2500, 0.0f, 1.49f, 0.21f, -2780, 0.300f, -2014, 0.100f, 27.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_QUARRY -1000,-1000, 0.0f, 1.49f, 0.83f,-10000, 0.061f, 500, 0.025f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_PLAIN -1000,-2000, 0.0f, 1.49f, 0.50f, -2466, 0.179f, -2514, 0.100f, 21.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_PARKINGLOT -1000, 0, 0.0f, 1.65f, 1.50f, -1363, 0.008f, -1153, 0.012f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_SEWERPIPE -1000,-1000, 0.0f, 2.81f, 0.14f, 429, 0.014f, 648, 0.021f, 80.0f, 60.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_UNDERWATER -1000,-4000, 0.0f, 1.49f, 0.10f, -449, 0.007f, 1700, 0.011f, 100.0f, 100.0f, 5000.0f + +// +// Examples simulating 'musical' reverb presets +// +// Name Decay time Description +// Small Room 1.1s A small size room with a length of 5m or so. +// Medium Room 1.3s A medium size room with a length of 10m or so. +// Large Room 1.5s A large size room suitable for live performances. +// Medium Hall 1.8s A medium size concert hall. +// Large Hall 1.8s A large size concert hall suitable for a full orchestra. +// Plate 1.3s A plate reverb simulation. +// + +#define I3DL2_ENVIRONMENT_PRESET_SMALLROOM -1000, -600, 0.0f, 1.10f, 0.83f, -400, 0.005f, 500, 0.010f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_MEDIUMROOM -1000, -600, 0.0f, 1.30f, 0.83f, -1000, 0.010f, -200, 0.020f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_LARGEROOM -1000, -600, 0.0f, 1.50f, 0.83f, -1600, 0.020f, -1000, 0.040f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_MEDIUMHALL -1000, -600, 0.0f, 1.80f, 0.70f, -1300, 0.015f, -800, 0.030f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_LARGEHALL -1000, -600, 0.0f, 1.80f, 0.70f, -2000, 0.030f, -1400, 0.060f, 100.0f, 100.0f, 5000.0f +#define I3DL2_ENVIRONMENT_PRESET_PLATE -1000, -200, 0.0f, 1.30f, 0.90f, 0, 0.002f, 0, 0.010f, 100.0f, 75.0f, 5000.0f + +// +// DirectSound3D Algorithms +// + +// Default DirectSound3D algorithm {00000000-0000-0000-0000-000000000000} +#define DS3DALG_DEFAULT GUID_NULL + +// No virtualization (Pan3D) {C241333F-1C1B-11d2-94F5-00C04FC28ACA} +DEFINE_GUID(DS3DALG_NO_VIRTUALIZATION, 0xc241333f, 0x1c1b, 0x11d2, 0x94, 0xf5, 0x0, 0xc0, 0x4f, 0xc2, 0x8a, 0xca); + +// High-quality HRTF algorithm {C2413340-1C1B-11d2-94F5-00C04FC28ACA} +DEFINE_GUID(DS3DALG_HRTF_FULL, 0xc2413340, 0x1c1b, 0x11d2, 0x94, 0xf5, 0x0, 0xc0, 0x4f, 0xc2, 0x8a, 0xca); + +// Lower-quality HRTF algorithm {C2413342-1C1B-11d2-94F5-00C04FC28ACA} +DEFINE_GUID(DS3DALG_HRTF_LIGHT, 0xc2413342, 0x1c1b, 0x11d2, 0x94, 0xf5, 0x0, 0xc0, 0x4f, 0xc2, 0x8a, 0xca); + + +#if DIRECTSOUND_VERSION >= 0x0800 + +// +// DirectSound Internal Effect Algorithms +// + + +// Gargle {DAFD8210-5711-4B91-9FE3-F75B7AE279BF} +DEFINE_GUID(GUID_DSFX_STANDARD_GARGLE, 0xdafd8210, 0x5711, 0x4b91, 0x9f, 0xe3, 0xf7, 0x5b, 0x7a, 0xe2, 0x79, 0xbf); + +// Chorus {EFE6629C-81F7-4281-BD91-C9D604A95AF6} +DEFINE_GUID(GUID_DSFX_STANDARD_CHORUS, 0xefe6629c, 0x81f7, 0x4281, 0xbd, 0x91, 0xc9, 0xd6, 0x04, 0xa9, 0x5a, 0xf6); + +// Flanger {EFCA3D92-DFD8-4672-A603-7420894BAD98} +DEFINE_GUID(GUID_DSFX_STANDARD_FLANGER, 0xefca3d92, 0xdfd8, 0x4672, 0xa6, 0x03, 0x74, 0x20, 0x89, 0x4b, 0xad, 0x98); + +// Echo/Delay {EF3E932C-D40B-4F51-8CCF-3F98F1B29D5D} +DEFINE_GUID(GUID_DSFX_STANDARD_ECHO, 0xef3e932c, 0xd40b, 0x4f51, 0x8c, 0xcf, 0x3f, 0x98, 0xf1, 0xb2, 0x9d, 0x5d); + +// Distortion {EF114C90-CD1D-484E-96E5-09CFAF912A21} +DEFINE_GUID(GUID_DSFX_STANDARD_DISTORTION, 0xef114c90, 0xcd1d, 0x484e, 0x96, 0xe5, 0x09, 0xcf, 0xaf, 0x91, 0x2a, 0x21); + +// Compressor/Limiter {EF011F79-4000-406D-87AF-BFFB3FC39D57} +DEFINE_GUID(GUID_DSFX_STANDARD_COMPRESSOR, 0xef011f79, 0x4000, 0x406d, 0x87, 0xaf, 0xbf, 0xfb, 0x3f, 0xc3, 0x9d, 0x57); + +// Parametric Equalization {120CED89-3BF4-4173-A132-3CB406CF3231} +DEFINE_GUID(GUID_DSFX_STANDARD_PARAMEQ, 0x120ced89, 0x3bf4, 0x4173, 0xa1, 0x32, 0x3c, 0xb4, 0x06, 0xcf, 0x32, 0x31); + +// I3DL2 Environmental Reverberation: Reverb (Listener) Effect {EF985E71-D5C7-42D4-BA4D-2D073E2E96F4} +DEFINE_GUID(GUID_DSFX_STANDARD_I3DL2REVERB, 0xef985e71, 0xd5c7, 0x42d4, 0xba, 0x4d, 0x2d, 0x07, 0x3e, 0x2e, 0x96, 0xf4); + +// Waves Reverberation {87FC0268-9A55-4360-95AA-004A1D9DE26C} +DEFINE_GUID(GUID_DSFX_WAVES_REVERB, 0x87fc0268, 0x9a55, 0x4360, 0x95, 0xaa, 0x00, 0x4a, 0x1d, 0x9d, 0xe2, 0x6c); + +// +// DirectSound Capture Effect Algorithms +// + + +// Acoustic Echo Canceller {BF963D80-C559-11D0-8A2B-00A0C9255AC1} +// Matches KSNODETYPE_ACOUSTIC_ECHO_CANCEL in ksmedia.h +DEFINE_GUID(GUID_DSCFX_CLASS_AEC, 0xBF963D80L, 0xC559, 0x11D0, 0x8A, 0x2B, 0x00, 0xA0, 0xC9, 0x25, 0x5A, 0xC1); + +// Microsoft AEC {CDEBB919-379A-488a-8765-F53CFD36DE40} +DEFINE_GUID(GUID_DSCFX_MS_AEC, 0xcdebb919, 0x379a, 0x488a, 0x87, 0x65, 0xf5, 0x3c, 0xfd, 0x36, 0xde, 0x40); + +// System AEC {1C22C56D-9879-4f5b-A389-27996DDC2810} +DEFINE_GUID(GUID_DSCFX_SYSTEM_AEC, 0x1c22c56d, 0x9879, 0x4f5b, 0xa3, 0x89, 0x27, 0x99, 0x6d, 0xdc, 0x28, 0x10); + +// Noise Supression {E07F903F-62FD-4e60-8CDD-DEA7236665B5} +// Matches KSNODETYPE_NOISE_SUPPRESS in post Windows ME DDK's ksmedia.h +DEFINE_GUID(GUID_DSCFX_CLASS_NS, 0xe07f903f, 0x62fd, 0x4e60, 0x8c, 0xdd, 0xde, 0xa7, 0x23, 0x66, 0x65, 0xb5); + +// Microsoft Noise Suppresion {11C5C73B-66E9-4ba1-A0BA-E814C6EED92D} +DEFINE_GUID(GUID_DSCFX_MS_NS, 0x11c5c73b, 0x66e9, 0x4ba1, 0xa0, 0xba, 0xe8, 0x14, 0xc6, 0xee, 0xd9, 0x2d); + +// System Noise Suppresion {5AB0882E-7274-4516-877D-4EEE99BA4FD0} +DEFINE_GUID(GUID_DSCFX_SYSTEM_NS, 0x5ab0882e, 0x7274, 0x4516, 0x87, 0x7d, 0x4e, 0xee, 0x99, 0xba, 0x4f, 0xd0); + +#endif // DIRECTSOUND_VERSION >= 0x0800 + +#endif // __DSOUND_INCLUDED__ + + + +#ifdef __cplusplus +}; +#endif // __cplusplus + diff --git a/rtaudio/include/ginclude.h b/rtaudio/include/ginclude.h new file mode 100644 index 0000000..b627dc2 --- /dev/null +++ b/rtaudio/include/ginclude.h @@ -0,0 +1,38 @@ +#ifndef __gInclude__ +#define __gInclude__ + +#if SGI + #undef BEOS + #undef MAC + #undef WINDOWS + // + #define ASIO_BIG_ENDIAN 1 + #define ASIO_CPU_MIPS 1 +#elif defined WIN32 + #undef BEOS + #undef MAC + #undef SGI + #define WINDOWS 1 + #define ASIO_LITTLE_ENDIAN 1 + #define ASIO_CPU_X86 1 +#elif BEOS + #undef MAC + #undef SGI + #undef WINDOWS + #define ASIO_LITTLE_ENDIAN 1 + #define ASIO_CPU_X86 1 + // +#else + #define MAC 1 + #undef BEOS + #undef WINDOWS + #undef SGI + #define ASIO_BIG_ENDIAN 1 + #define ASIO_CPU_PPC 1 +#endif + +// always +#define NATIVE_INT64 0 +#define IEEE754_64FLOAT 1 + +#endif // __gInclude__ diff --git a/rtaudio/include/iasiodrv.h b/rtaudio/include/iasiodrv.h new file mode 100644 index 0000000..64d2dbb --- /dev/null +++ b/rtaudio/include/iasiodrv.h @@ -0,0 +1,37 @@ +#include "asiosys.h" +#include "asio.h" + +/* Forward Declarations */ + +#ifndef __ASIODRIVER_FWD_DEFINED__ +#define __ASIODRIVER_FWD_DEFINED__ +typedef interface IASIO IASIO; +#endif /* __ASIODRIVER_FWD_DEFINED__ */ + +interface IASIO : public IUnknown +{ + + virtual ASIOBool init(void *sysHandle) = 0; + virtual void getDriverName(char *name) = 0; + virtual long getDriverVersion() = 0; + virtual void getErrorMessage(char *string) = 0; + virtual ASIOError start() = 0; + virtual ASIOError stop() = 0; + virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels) = 0; + virtual ASIOError getLatencies(long *inputLatency, long *outputLatency) = 0; + virtual ASIOError getBufferSize(long *minSize, long *maxSize, + long *preferredSize, long *granularity) = 0; + virtual ASIOError canSampleRate(ASIOSampleRate sampleRate) = 0; + virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate) = 0; + virtual ASIOError setSampleRate(ASIOSampleRate sampleRate) = 0; + virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources) = 0; + virtual ASIOError setClockSource(long reference) = 0; + virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0; + virtual ASIOError getChannelInfo(ASIOChannelInfo *info) = 0; + virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels, + long bufferSize, ASIOCallbacks *callbacks) = 0; + virtual ASIOError disposeBuffers() = 0; + virtual ASIOError controlPanel() = 0; + virtual ASIOError future(long selector,void *opt) = 0; + virtual ASIOError outputReady() = 0; +}; diff --git a/rtaudio/include/iasiothiscallresolver.cpp b/rtaudio/include/iasiothiscallresolver.cpp new file mode 100644 index 0000000..38c39d2 --- /dev/null +++ b/rtaudio/include/iasiothiscallresolver.cpp @@ -0,0 +1,563 @@ +/* + IASIOThiscallResolver.cpp see the comments in iasiothiscallresolver.h for + the top level description - this comment describes the technical details of + the implementation. + + The latest version of this file is available from: + http://www.audiomulch.com/~rossb/code/calliasio + + please email comments to Ross Bencina + + BACKGROUND + + The IASIO interface declared in the Steinberg ASIO 2 SDK declares + functions with no explicit calling convention. This causes MSVC++ to default + to using the thiscall convention, which is a proprietary convention not + implemented by some non-microsoft compilers - notably borland BCC, + C++Builder, and gcc. MSVC++ is the defacto standard compiler used by + Steinberg. As a result of this situation, the ASIO sdk will compile with + any compiler, however attempting to execute the compiled code will cause a + crash due to different default calling conventions on non-Microsoft + compilers. + + IASIOThiscallResolver solves the problem by providing an adapter class that + delegates to the IASIO interface using the correct calling convention + (thiscall). Due to the lack of support for thiscall in the Borland and GCC + compilers, the calls have been implemented in assembly language. + + A number of macros are defined for thiscall function calls with different + numbers of parameters, with and without return values - it may be possible + to modify the format of these macros to make them work with other inline + assemblers. + + + THISCALL DEFINITION + + A number of definitions of the thiscall calling convention are floating + around the internet. The following definition has been validated against + output from the MSVC++ compiler: + + For non-vararg functions, thiscall works as follows: the object (this) + pointer is passed in ECX. All arguments are passed on the stack in + right to left order. The return value is placed in EAX. The callee + clears the passed arguments from the stack. + + + FINDING FUNCTION POINTERS FROM AN IASIO POINTER + + The first field of a COM object is a pointer to its vtble. Thus a pointer + to an object implementing the IASIO interface also points to a pointer to + that object's vtbl. The vtble is a table of function pointers for all of + the virtual functions exposed by the implemented interfaces. + + If we consider a variable declared as a pointer to IASO: + + IASIO *theAsioDriver + + theAsioDriver points to: + + object implementing IASIO + { + IASIOvtbl *vtbl + other data + } + + in other words, theAsioDriver points to a pointer to an IASIOvtbl + + vtbl points to a table of function pointers: + + IASIOvtbl ( interface IASIO : public IUnknown ) + { + (IUnknown functions) + 0 virtual HRESULT STDMETHODCALLTYPE (*QueryInterface)(REFIID riid, void **ppv) = 0; + 4 virtual ULONG STDMETHODCALLTYPE (*AddRef)() = 0; + 8 virtual ULONG STDMETHODCALLTYPE (*Release)() = 0; + + (IASIO functions) + 12 virtual ASIOBool (*init)(void *sysHandle) = 0; + 16 virtual void (*getDriverName)(char *name) = 0; + 20 virtual long (*getDriverVersion)() = 0; + 24 virtual void (*getErrorMessage)(char *string) = 0; + 28 virtual ASIOError (*start)() = 0; + 32 virtual ASIOError (*stop)() = 0; + 36 virtual ASIOError (*getChannels)(long *numInputChannels, long *numOutputChannels) = 0; + 40 virtual ASIOError (*getLatencies)(long *inputLatency, long *outputLatency) = 0; + 44 virtual ASIOError (*getBufferSize)(long *minSize, long *maxSize, + long *preferredSize, long *granularity) = 0; + 48 virtual ASIOError (*canSampleRate)(ASIOSampleRate sampleRate) = 0; + 52 virtual ASIOError (*getSampleRate)(ASIOSampleRate *sampleRate) = 0; + 56 virtual ASIOError (*setSampleRate)(ASIOSampleRate sampleRate) = 0; + 60 virtual ASIOError (*getClockSources)(ASIOClockSource *clocks, long *numSources) = 0; + 64 virtual ASIOError (*setClockSource)(long reference) = 0; + 68 virtual ASIOError (*getSamplePosition)(ASIOSamples *sPos, ASIOTimeStamp *tStamp) = 0; + 72 virtual ASIOError (*getChannelInfo)(ASIOChannelInfo *info) = 0; + 76 virtual ASIOError (*createBuffers)(ASIOBufferInfo *bufferInfos, long numChannels, + long bufferSize, ASIOCallbacks *callbacks) = 0; + 80 virtual ASIOError (*disposeBuffers)() = 0; + 84 virtual ASIOError (*controlPanel)() = 0; + 88 virtual ASIOError (*future)(long selector,void *opt) = 0; + 92 virtual ASIOError (*outputReady)() = 0; + }; + + The numbers in the left column show the byte offset of each function ptr + from the beginning of the vtbl. These numbers are used in the code below + to select different functions. + + In order to find the address of a particular function, theAsioDriver + must first be dereferenced to find the value of the vtbl pointer: + + mov eax, theAsioDriver + mov edx, [theAsioDriver] // edx now points to vtbl[0] + + Then an offset must be added to the vtbl pointer to select a + particular function, for example vtbl+44 points to the slot containing + a pointer to the getBufferSize function. + + Finally vtbl+x must be dereferenced to obtain the value of the function + pointer stored in that address: + + call [edx+44] // call the function pointed to by + // the value in the getBufferSize field of the vtbl + + + SEE ALSO + + Martin Fay's OpenASIO DLL at http://www.martinfay.com solves the same + problem by providing a new COM interface which wraps IASIO with an + interface that uses portable calling conventions. OpenASIO must be compiled + with MSVC, and requires that you ship the OpenASIO DLL with your + application. + + + ACKNOWLEDGEMENTS + + Ross Bencina: worked out the thiscall details above, wrote the original + Borland asm macros, and a patch for asio.cpp (which is no longer needed). + Thanks to Martin Fay for introducing me to the issues discussed here, + and to Rene G. Ceballos for assisting with asm dumps from MSVC++. + + Antti Silvast: converted the original calliasio to work with gcc and NASM + by implementing the asm code in a separate file. + + Fraser Adams: modified the original calliasio containing the Borland inline + asm to add inline asm for gcc i.e. Intel syntax for Borland and AT&T syntax + for gcc. This seems a neater approach for gcc than to have a separate .asm + file and it means that we only need one version of the thiscall patch. + + Fraser Adams: rewrote the original calliasio patch in the form of the + IASIOThiscallResolver class in order to avoid modifications to files from + the Steinberg SDK, which may have had potential licence issues. + + Andrew Baldwin: contributed fixes for compatibility problems with more + recent versions of the gcc assembler. +*/ + + +// We only need IASIOThiscallResolver at all if we are on Win32. For other +// platforms we simply bypass the IASIOThiscallResolver definition to allow us +// to be safely #include'd whatever the platform to keep client code portable +#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__) + + +// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver +// is not used. +#if !defined(_MSC_VER) + + +#include +#include + +// We have a mechanism in iasiothiscallresolver.h to ensure that asio.h is +// #include'd before it in client code, we do NOT want to do this test here. +#define iasiothiscallresolver_sourcefile 1 +#include "iasiothiscallresolver.h" +#undef iasiothiscallresolver_sourcefile + +// iasiothiscallresolver.h redefines ASIOInit for clients, but we don't want +// this macro defined in this translation unit. +#undef ASIOInit + + +// theAsioDriver is a global pointer to the current IASIO instance which the +// ASIO SDK uses to perform all actions on the IASIO interface. We substitute +// our own forwarding interface into this pointer. +extern IASIO* theAsioDriver; + + +// The following macros define the inline assembler for BORLAND first then gcc + +#if defined(__BCPLUSPLUS__) || defined(__BORLANDC__) + + +#define CALL_THISCALL_0( resultName, thisPtr, funcOffset )\ + void *this_ = (thisPtr); \ + __asm { \ + mov ecx, this_ ; \ + mov eax, [ecx] ; \ + call [eax+funcOffset] ; \ + mov resultName, eax ; \ + } + + +#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 )\ + void *this_ = (thisPtr); \ + __asm { \ + mov eax, param1 ; \ + push eax ; \ + mov ecx, this_ ; \ + mov eax, [ecx] ; \ + call [eax+funcOffset] ; \ + } + + +#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 )\ + void *this_ = (thisPtr); \ + __asm { \ + mov eax, param1 ; \ + push eax ; \ + mov ecx, this_ ; \ + mov eax, [ecx] ; \ + call [eax+funcOffset] ; \ + mov resultName, eax ; \ + } + + +#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 )\ + void *this_ = (thisPtr); \ + void *doubleParamPtr_ (¶m1); \ + __asm { \ + mov eax, doubleParamPtr_ ; \ + push [eax+4] ; \ + push [eax] ; \ + mov ecx, this_ ; \ + mov eax, [ecx] ; \ + call [eax+funcOffset] ; \ + mov resultName, eax ; \ + } + + +#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 )\ + void *this_ = (thisPtr); \ + __asm { \ + mov eax, param2 ; \ + push eax ; \ + mov eax, param1 ; \ + push eax ; \ + mov ecx, this_ ; \ + mov eax, [ecx] ; \ + call [eax+funcOffset] ; \ + mov resultName, eax ; \ + } + + +#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\ + void *this_ = (thisPtr); \ + __asm { \ + mov eax, param4 ; \ + push eax ; \ + mov eax, param3 ; \ + push eax ; \ + mov eax, param2 ; \ + push eax ; \ + mov eax, param1 ; \ + push eax ; \ + mov ecx, this_ ; \ + mov eax, [ecx] ; \ + call [eax+funcOffset] ; \ + mov resultName, eax ; \ + } + + +#elif defined(__GNUC__) + + +#define CALL_THISCALL_0( resultName, thisPtr, funcOffset ) \ + __asm__ __volatile__ ("movl (%1), %%edx\n\t" \ + "call *"#funcOffset"(%%edx)\n\t" \ + :"=a"(resultName) /* Output Operands */ \ + :"c"(thisPtr) /* Input Operands */ \ + ); \ + + +#define CALL_VOID_THISCALL_1( thisPtr, funcOffset, param1 ) \ + __asm__ __volatile__ ("pushl %0\n\t" \ + "movl (%1), %%edx\n\t" \ + "call *"#funcOffset"(%%edx)\n\t" \ + : /* Output Operands */ \ + :"r"(param1), /* Input Operands */ \ + "c"(thisPtr) \ + ); \ + + +#define CALL_THISCALL_1( resultName, thisPtr, funcOffset, param1 ) \ + __asm__ __volatile__ ("pushl %1\n\t" \ + "movl (%2), %%edx\n\t" \ + "call *"#funcOffset"(%%edx)\n\t" \ + :"=a"(resultName) /* Output Operands */ \ + :"r"(param1), /* Input Operands */ \ + "c"(thisPtr) \ + ); \ + + +#define CALL_THISCALL_1_DOUBLE( resultName, thisPtr, funcOffset, param1 ) \ + __asm__ __volatile__ ("pushl 4(%1)\n\t" \ + "pushl (%1)\n\t" \ + "movl (%2), %%edx\n\t" \ + "call *"#funcOffset"(%%edx);\n\t" \ + :"=a"(resultName) /* Output Operands */ \ + :"a"(¶m1), /* Input Operands */ \ + /* Note: Using "r" above instead of "a" fails */ \ + /* when using GCC 3.3.3, and maybe later versions*/\ + "c"(thisPtr) \ + ); \ + + +#define CALL_THISCALL_2( resultName, thisPtr, funcOffset, param1, param2 ) \ + __asm__ __volatile__ ("pushl %1\n\t" \ + "pushl %2\n\t" \ + "movl (%3), %%edx\n\t" \ + "call *"#funcOffset"(%%edx)\n\t" \ + :"=a"(resultName) /* Output Operands */ \ + :"r"(param2), /* Input Operands */ \ + "r"(param1), \ + "c"(thisPtr) \ + ); \ + + +#define CALL_THISCALL_4( resultName, thisPtr, funcOffset, param1, param2, param3, param4 )\ + __asm__ __volatile__ ("pushl %1\n\t" \ + "pushl %2\n\t" \ + "pushl %3\n\t" \ + "pushl %4\n\t" \ + "movl (%5), %%edx\n\t" \ + "call *"#funcOffset"(%%edx)\n\t" \ + :"=a"(resultName) /* Output Operands */ \ + :"r"(param4), /* Input Operands */ \ + "r"(param3), \ + "r"(param2), \ + "r"(param1), \ + "c"(thisPtr) \ + ); \ + +#endif + + + +// Our static singleton instance. +IASIOThiscallResolver IASIOThiscallResolver::instance; + +// Constructor called to initialize static Singleton instance above. Note that +// it is important not to clear that_ incase it has already been set by the call +// to placement new in ASIOInit(). +IASIOThiscallResolver::IASIOThiscallResolver() +{ +} + +// Constructor called from ASIOInit() below +IASIOThiscallResolver::IASIOThiscallResolver(IASIO* that) +: that_( that ) +{ +} + +// Implement IUnknown methods as assert(false). IASIOThiscallResolver is not +// really a COM object, just a wrapper which will work with the ASIO SDK. +// If you wanted to use ASIO without the SDK you might want to implement COM +// aggregation in these methods. +HRESULT STDMETHODCALLTYPE IASIOThiscallResolver::QueryInterface(REFIID riid, void **ppv) +{ + (void)riid; // suppress unused variable warning + + assert( false ); // this function should never be called by the ASIO SDK. + + *ppv = NULL; + return E_NOINTERFACE; +} + +ULONG STDMETHODCALLTYPE IASIOThiscallResolver::AddRef() +{ + assert( false ); // this function should never be called by the ASIO SDK. + + return 1; +} + +ULONG STDMETHODCALLTYPE IASIOThiscallResolver::Release() +{ + assert( false ); // this function should never be called by the ASIO SDK. + + return 1; +} + + +// Implement the IASIO interface methods by performing the vptr manipulation +// described above then delegating to the real implementation. +ASIOBool IASIOThiscallResolver::init(void *sysHandle) +{ + ASIOBool result; + CALL_THISCALL_1( result, that_, 12, sysHandle ); + return result; +} + +void IASIOThiscallResolver::getDriverName(char *name) +{ + CALL_VOID_THISCALL_1( that_, 16, name ); +} + +long IASIOThiscallResolver::getDriverVersion() +{ + ASIOBool result; + CALL_THISCALL_0( result, that_, 20 ); + return result; +} + +void IASIOThiscallResolver::getErrorMessage(char *string) +{ + CALL_VOID_THISCALL_1( that_, 24, string ); +} + +ASIOError IASIOThiscallResolver::start() +{ + ASIOBool result; + CALL_THISCALL_0( result, that_, 28 ); + return result; +} + +ASIOError IASIOThiscallResolver::stop() +{ + ASIOBool result; + CALL_THISCALL_0( result, that_, 32 ); + return result; +} + +ASIOError IASIOThiscallResolver::getChannels(long *numInputChannels, long *numOutputChannels) +{ + ASIOBool result; + CALL_THISCALL_2( result, that_, 36, numInputChannels, numOutputChannels ); + return result; +} + +ASIOError IASIOThiscallResolver::getLatencies(long *inputLatency, long *outputLatency) +{ + ASIOBool result; + CALL_THISCALL_2( result, that_, 40, inputLatency, outputLatency ); + return result; +} + +ASIOError IASIOThiscallResolver::getBufferSize(long *minSize, long *maxSize, + long *preferredSize, long *granularity) +{ + ASIOBool result; + CALL_THISCALL_4( result, that_, 44, minSize, maxSize, preferredSize, granularity ); + return result; +} + +ASIOError IASIOThiscallResolver::canSampleRate(ASIOSampleRate sampleRate) +{ + ASIOBool result; + CALL_THISCALL_1_DOUBLE( result, that_, 48, sampleRate ); + return result; +} + +ASIOError IASIOThiscallResolver::getSampleRate(ASIOSampleRate *sampleRate) +{ + ASIOBool result; + CALL_THISCALL_1( result, that_, 52, sampleRate ); + return result; +} + +ASIOError IASIOThiscallResolver::setSampleRate(ASIOSampleRate sampleRate) +{ + ASIOBool result; + CALL_THISCALL_1_DOUBLE( result, that_, 56, sampleRate ); + return result; +} + +ASIOError IASIOThiscallResolver::getClockSources(ASIOClockSource *clocks, long *numSources) +{ + ASIOBool result; + CALL_THISCALL_2( result, that_, 60, clocks, numSources ); + return result; +} + +ASIOError IASIOThiscallResolver::setClockSource(long reference) +{ + ASIOBool result; + CALL_THISCALL_1( result, that_, 64, reference ); + return result; +} + +ASIOError IASIOThiscallResolver::getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp) +{ + ASIOBool result; + CALL_THISCALL_2( result, that_, 68, sPos, tStamp ); + return result; +} + +ASIOError IASIOThiscallResolver::getChannelInfo(ASIOChannelInfo *info) +{ + ASIOBool result; + CALL_THISCALL_1( result, that_, 72, info ); + return result; +} + +ASIOError IASIOThiscallResolver::createBuffers(ASIOBufferInfo *bufferInfos, + long numChannels, long bufferSize, ASIOCallbacks *callbacks) +{ + ASIOBool result; + CALL_THISCALL_4( result, that_, 76, bufferInfos, numChannels, bufferSize, callbacks ); + return result; +} + +ASIOError IASIOThiscallResolver::disposeBuffers() +{ + ASIOBool result; + CALL_THISCALL_0( result, that_, 80 ); + return result; +} + +ASIOError IASIOThiscallResolver::controlPanel() +{ + ASIOBool result; + CALL_THISCALL_0( result, that_, 84 ); + return result; +} + +ASIOError IASIOThiscallResolver::future(long selector,void *opt) +{ + ASIOBool result; + CALL_THISCALL_2( result, that_, 88, selector, opt ); + return result; +} + +ASIOError IASIOThiscallResolver::outputReady() +{ + ASIOBool result; + CALL_THISCALL_0( result, that_, 92 ); + return result; +} + + +// Implement our substitute ASIOInit() method +ASIOError IASIOThiscallResolver::ASIOInit(ASIODriverInfo *info) +{ + // To ensure that our instance's vptr is correctly constructed, even if + // ASIOInit is called prior to main(), we explicitly call its constructor + // (potentially over the top of an existing instance). Note that this is + // pretty ugly, and is only safe because IASIOThiscallResolver has no + // destructor and contains no objects with destructors. + new((void*)&instance) IASIOThiscallResolver( theAsioDriver ); + + // Interpose between ASIO client code and the real driver. + theAsioDriver = &instance; + + // Note that we never need to switch theAsioDriver back to point to the + // real driver because theAsioDriver is reset to zero in ASIOExit(). + + // Delegate to the real ASIOInit + return ::ASIOInit(info); +} + + +#endif /* !defined(_MSC_VER) */ + +#endif /* Win32 */ + diff --git a/rtaudio/include/iasiothiscallresolver.h b/rtaudio/include/iasiothiscallresolver.h new file mode 100644 index 0000000..b59d910 --- /dev/null +++ b/rtaudio/include/iasiothiscallresolver.h @@ -0,0 +1,201 @@ +// **************************************************************************** +// +// Changed: I have modified this file slightly (includes) to work with +// RtAudio. RtAudio.cpp must include this file after asio.h. +// +// File: IASIOThiscallResolver.h +// Description: The IASIOThiscallResolver class implements the IASIO +// interface and acts as a proxy to the real IASIO interface by +// calling through its vptr table using the thiscall calling +// convention. To put it another way, we interpose +// IASIOThiscallResolver between ASIO SDK code and the driver. +// This is necessary because most non-Microsoft compilers don't +// implement the thiscall calling convention used by IASIO. +// +// iasiothiscallresolver.cpp contains the background of this +// problem plus a technical description of the vptr +// manipulations. +// +// In order to use this mechanism one simply has to add +// iasiothiscallresolver.cpp to the list of files to compile +// and #include +// +// Note that this #include must come after the other ASIO SDK +// #includes, for example: +// +// #include +// #include +// #include +// #include +// #include +// +// Actually the important thing is to #include +// after . We have +// incorporated a test to enforce this ordering. +// +// The code transparently takes care of the interposition by +// using macro substitution to intercept calls to ASIOInit() +// and ASIOExit(). We save the original ASIO global +// "theAsioDriver" in our "that" variable, and then set +// "theAsioDriver" to equal our IASIOThiscallResolver instance. +// +// Whilst this method of resolving the thiscall problem requires +// the addition of #include to client +// code it has the advantage that it does not break the terms +// of the ASIO licence by publishing it. We are NOT modifying +// any Steinberg code here, we are merely implementing the IASIO +// interface in the same way that we would need to do if we +// wished to provide an open source ASIO driver. +// +// For compilation with MinGW -lole32 needs to be added to the +// linker options. For BORLAND, linking with Import32.lib is +// sufficient. +// +// The dependencies are with: CoInitialize, CoUninitialize, +// CoCreateInstance, CLSIDFromString - used by asiolist.cpp +// and are required on Windows whether ThiscallResolver is used +// or not. +// +// Searching for the above strings in the root library path +// of your compiler should enable the correct libraries to be +// identified if they aren't immediately obvious. +// +// Note that the current implementation of IASIOThiscallResolver +// is not COM compliant - it does not correctly implement the +// IUnknown interface. Implementing it is not necessary because +// it is not called by parts of the ASIO SDK which call through +// theAsioDriver ptr. The IUnknown methods are implemented as +// assert(false) to ensure that the code fails if they are +// ever called. +// Restrictions: None. Public Domain & Open Source distribute freely +// You may use IASIOThiscallResolver commercially as well as +// privately. +// You the user assume the responsibility for the use of the +// files, binary or text, and there is no guarantee or warranty, +// expressed or implied, including but not limited to the +// implied warranties of merchantability and fitness for a +// particular purpose. You assume all responsibility and agree +// to hold no entity, copyright holder or distributors liable +// for any loss of data or inaccurate representations of data +// as a result of using IASIOThiscallResolver. +// Version: 1.4 Added separate macro CALL_THISCALL_1_DOUBLE from +// Andrew Baldwin, and volatile for whole gcc asm blocks, +// both for compatibility with newer gcc versions. Cleaned up +// Borland asm to use one less register. +// 1.3 Switched to including assert.h for better compatibility. +// Wrapped entire .h and .cpp contents with a check for +// _MSC_VER to provide better compatibility with MS compilers. +// Changed Singleton implementation to use static instance +// instead of freestore allocated instance. Removed ASIOExit +// macro as it is no longer needed. +// 1.2 Removed semicolons from ASIOInit and ASIOExit macros to +// allow them to be embedded in expressions (if statements). +// Cleaned up some comments. Removed combase.c dependency (it +// doesn't compile with BCB anyway) by stubbing IUnknown. +// 1.1 Incorporated comments from Ross Bencina including things +// such as changing name from ThiscallResolver to +// IASIOThiscallResolver, tidying up the constructor, fixing +// a bug in IASIOThiscallResolver::ASIOExit() and improving +// portability through the use of conditional compilation +// 1.0 Initial working version. +// Created: 6/09/2003 +// Authors: Fraser Adams +// Ross Bencina +// Rene G. Ceballos +// Martin Fay +// Antti Silvast +// Andrew Baldwin +// +// **************************************************************************** + + +#ifndef included_iasiothiscallresolver_h +#define included_iasiothiscallresolver_h + +// We only need IASIOThiscallResolver at all if we are on Win32. For other +// platforms we simply bypass the IASIOThiscallResolver definition to allow us +// to be safely #include'd whatever the platform to keep client code portable +#if defined(WIN32) || defined(_WIN32) || defined(__WIN32__) + + +// If microsoft compiler we can call IASIO directly so IASIOThiscallResolver +// is not used. +#if !defined(_MSC_VER) + + +// The following is in order to ensure that this header is only included after +// the other ASIO headers (except for the case of iasiothiscallresolver.cpp). +// We need to do this because IASIOThiscallResolver works by eclipsing the +// original definition of ASIOInit() with a macro (see below). +#if !defined(iasiothiscallresolver_sourcefile) + #if !defined(__ASIO_H) + #error iasiothiscallresolver.h must be included AFTER asio.h + #endif +#endif + +#include +#include "iasiodrv.h" /* From ASIO SDK */ + + +class IASIOThiscallResolver : public IASIO { +private: + IASIO* that_; // Points to the real IASIO + + static IASIOThiscallResolver instance; // Singleton instance + + // Constructors - declared private so construction is limited to + // our Singleton instance + IASIOThiscallResolver(); + IASIOThiscallResolver(IASIO* that); +public: + + // Methods from the IUnknown interface. We don't fully implement IUnknown + // because the ASIO SDK never calls these methods through theAsioDriver ptr. + // These methods are implemented as assert(false). + virtual HRESULT STDMETHODCALLTYPE QueryInterface(REFIID riid, void **ppv); + virtual ULONG STDMETHODCALLTYPE AddRef(); + virtual ULONG STDMETHODCALLTYPE Release(); + + // Methods from the IASIO interface, implemented as forwarning calls to that. + virtual ASIOBool init(void *sysHandle); + virtual void getDriverName(char *name); + virtual long getDriverVersion(); + virtual void getErrorMessage(char *string); + virtual ASIOError start(); + virtual ASIOError stop(); + virtual ASIOError getChannels(long *numInputChannels, long *numOutputChannels); + virtual ASIOError getLatencies(long *inputLatency, long *outputLatency); + virtual ASIOError getBufferSize(long *minSize, long *maxSize, long *preferredSize, long *granularity); + virtual ASIOError canSampleRate(ASIOSampleRate sampleRate); + virtual ASIOError getSampleRate(ASIOSampleRate *sampleRate); + virtual ASIOError setSampleRate(ASIOSampleRate sampleRate); + virtual ASIOError getClockSources(ASIOClockSource *clocks, long *numSources); + virtual ASIOError setClockSource(long reference); + virtual ASIOError getSamplePosition(ASIOSamples *sPos, ASIOTimeStamp *tStamp); + virtual ASIOError getChannelInfo(ASIOChannelInfo *info); + virtual ASIOError createBuffers(ASIOBufferInfo *bufferInfos, long numChannels, long bufferSize, ASIOCallbacks *callbacks); + virtual ASIOError disposeBuffers(); + virtual ASIOError controlPanel(); + virtual ASIOError future(long selector,void *opt); + virtual ASIOError outputReady(); + + // Class method, see ASIOInit() macro below. + static ASIOError ASIOInit(ASIODriverInfo *info); // Delegates to ::ASIOInit +}; + + +// Replace calls to ASIOInit with our interposing version. +// This macro enables us to perform thiscall resolution simply by #including +// after the asio #includes (this file _must_ be +// included _after_ the asio #includes) + +#define ASIOInit(name) IASIOThiscallResolver::ASIOInit((name)) + + +#endif /* !defined(_MSC_VER) */ + +#endif /* Win32 */ + +#endif /* included_iasiothiscallresolver_h */ + + diff --git a/rtaudio/include/soundcard.h b/rtaudio/include/soundcard.h new file mode 100644 index 0000000..e8fc9f6 --- /dev/null +++ b/rtaudio/include/soundcard.h @@ -0,0 +1,2061 @@ +#ifndef SOUNDCARD_H +#define SOUNDCARD_H +/* + **************************************************************************** + * Copyright by 4Front Technologies 1993-2006 + * + ****************************************************************************** + * Modifications to this file are NOT allowed. This header file controls the + * OSS API. For compatibility reasons only 4Front Technologies can make changes + * to the definitions. If you have any questions, please contact + * hannu@opensound.com. + ****************************************************************************** + * + * Redistribution and use in source and binary forms, without + * modification, are permitted provided that the following conditions are + * met: 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. 2. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY + * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED + * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + * DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR + * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL + * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER + * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT + * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + **************************************************************************** + */ +/* + * Purpose: The C/C++ header file that defines the OSS API. + * Description: + * This header file contains all the declarations required to compile OSS + * programs. The latest version is always installed together with OSS + * use of the latest version is strongly recommended. + * + * {!notice This header file contains many obsolete definitions + * (for compatibility with older applications that still ned them). + * Do not use this file as a reference manual of OSS. + * Please check the OSS Programmer's guide for descriptions + * of the supported API details (http://www.opensound.com/pguide).} + */ + +#if defined(__cplusplus) +#define EXTERNC extern "C" +#else +#define EXTERNC extern +#endif /* EXTERN_C_WRAPPERS */ + +#define OSS_VERSION 0x040002 + +#define SOUND_VERSION OSS_VERSION +#define OPEN_SOUND_SYSTEM + +#if defined(__hpux) && !defined(_HPUX_SOURCE) +# error "-D_HPUX_SOURCE must be used when compiling OSS applications" +#endif + +#ifdef __hpux +#include +#endif + +#ifdef linux +/* In Linux we need to be prepared for cross compiling */ +#include +#else +# ifdef __FreeBSD__ +# include +# else +# include +# endif +#endif + +#ifndef __SIOWR +#if defined(__hpux) || (defined(_IOWR) && (defined(_AIX) || (!defined(sun) && !defined(sparc) && !defined(__INCioctlh) && !defined(__Lynx__)))) + +/* + * Make sure the ioctl macros are compatible with the ones already used + * by this operating system. + */ +#define SIOCPARM_MASK IOCPARM_MASK +#define SIOC_VOID IOC_VOID +#define SIOC_OUT IOC_OUT +#define SIOC_IN IOC_IN +#define SIOC_INOUT IOC_INOUT +#define __SIOC_SIZE _IOC_SIZE +#define __SIOC_DIR _IOC_DIR +#define __SIOC_NONE _IOC_NONE +#define __SIOC_READ _IOC_READ +#define __SIOC_WRITE _IOC_WRITE +#define __SIO _IO +#define __SIOR _IOR +#define __SIOW _IOW +#define __SIOWR _IOWR +#else + +/* #define SIOCTYPE (0xff<<8) */ +#define SIOCPARM_MASK 0x1fff /* parameters must be < 8192 bytes */ +#define SIOC_VOID 0x00000000 /* no parameters */ +#define SIOC_OUT 0x20000000 /* copy out parameters */ +#define SIOC_IN 0x40000000 /* copy in parameters */ +#define SIOC_INOUT (SIOC_IN|SIOC_OUT) + +#define __SIO(x,y) ((int)(SIOC_VOID|(x<<8)|y)) +#define __SIOR(x,y,t) ((int)(SIOC_OUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y)) +#define __SIOW(x,y,t) ((int)(SIOC_IN|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y)) +#define __SIOWR(x,y,t) ((int)(SIOC_INOUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y)) +#define __SIOC_SIZE(x) ((x>>16)&SIOCPARM_MASK) +#define __SIOC_DIR(x) (x & 0xf0000000) +#define __SIOC_NONE SIOC_VOID +#define __SIOC_READ SIOC_OUT +#define __SIOC_WRITE SIOC_IN +# endif /* _IOWR */ +#endif /* !__SIOWR */ + +#define OSS_LONGNAME_SIZE 64 +#define OSS_LABEL_SIZE 16 +#define OSS_DEVNODE_SIZE 32 +typedef char oss_longname_t[OSS_LONGNAME_SIZE]; +typedef char oss_label_t[OSS_LABEL_SIZE]; +typedef char oss_devnode_t[OSS_DEVNODE_SIZE]; + +#ifndef DISABLE_SEQUENCER +/* + **************************************************************************** + * IOCTL Commands for /dev/sequencer and /dev/music (AKA /dev/sequencer2) + * + * Note that this interface is obsolete and no longer developed. New + * applications should use /dev/midi instead. + ****************************************************************************/ +#define SNDCTL_SEQ_RESET __SIO ('Q', 0) +#define SNDCTL_SEQ_SYNC __SIO ('Q', 1) +#define SNDCTL_SYNTH_INFO __SIOWR('Q', 2, struct synth_info) +#define SNDCTL_SEQ_CTRLRATE __SIOWR('Q', 3, int) /* Set/get timer resolution (HZ) */ +#define SNDCTL_SEQ_GETOUTCOUNT __SIOR ('Q', 4, int) +#define SNDCTL_SEQ_GETINCOUNT __SIOR ('Q', 5, int) +#define SNDCTL_SEQ_PERCMODE __SIOW ('Q', 6, int) +#define SNDCTL_FM_LOAD_INSTR __SIOW ('Q', 7, struct sbi_instrument) /* Obsolete. Don't use!!!!!! */ +#define SNDCTL_SEQ_TESTMIDI __SIOW ('Q', 8, int) +#define SNDCTL_SEQ_RESETSAMPLES __SIOW ('Q', 9, int) +#define SNDCTL_SEQ_NRSYNTHS __SIOR ('Q',10, int) +#define SNDCTL_SEQ_NRMIDIS __SIOR ('Q',11, int) +#define SNDCTL_MIDI_INFO __SIOWR('Q',12, struct midi_info) /* OBSOLETE - use SNDCTL_MIDIINFO instead */ +#define SNDCTL_SEQ_THRESHOLD __SIOW ('Q',13, int) +#define SNDCTL_SYNTH_MEMAVL __SIOWR('Q',14, int) /* in=dev#, out=memsize */ +#define SNDCTL_FM_4OP_ENABLE __SIOW ('Q',15, int) /* in=dev# */ +#define SNDCTL_SEQ_PANIC __SIO ('Q',17) +#define SNDCTL_SEQ_OUTOFBAND __SIOW ('Q',18, struct seq_event_rec) +#define SNDCTL_SEQ_GETTIME __SIOR ('Q',19, int) +#define SNDCTL_SYNTH_ID __SIOWR('Q',20, struct synth_info) +#define SNDCTL_SYNTH_CONTROL __SIOWR('Q',21, struct synth_control) +#define SNDCTL_SYNTH_REMOVESAMPLE __SIOWR('Q',22, struct remove_sample) /* Reserved for future use */ +#define SNDCTL_SEQ_TIMING_ENABLE __SIO ('Q', 23) /* Enable incoming MIDI timing messages */ +#define SNDCTL_SEQ_ACTSENSE_ENABLE __SIO ('Q', 24) /* Enable incoming active sensing messages */ +#define SNDCTL_SEQ_RT_ENABLE __SIO ('Q', 25) /* Enable other incoming realtime messages */ + +typedef struct synth_control +{ + int devno; /* Synthesizer # */ + char data[4000]; /* Device spesific command/data record */ +} synth_control; + +typedef struct remove_sample +{ + int devno; /* Synthesizer # */ + int bankno; /* MIDI bank # (0=General MIDI) */ + int instrno; /* MIDI instrument number */ +} remove_sample; + +typedef struct seq_event_rec +{ + unsigned char arr[8]; +} seq_event_rec; + +#define SNDCTL_TMR_TIMEBASE __SIOWR('T', 1, int) +#define SNDCTL_TMR_START __SIO ('T', 2) +#define SNDCTL_TMR_STOP __SIO ('T', 3) +#define SNDCTL_TMR_CONTINUE __SIO ('T', 4) +#define SNDCTL_TMR_TEMPO __SIOWR('T', 5, int) +#define SNDCTL_TMR_SOURCE __SIOWR('T', 6, int) +# define TMR_INTERNAL 0x00000001 +# define TMR_EXTERNAL 0x00000002 +# define TMR_MODE_MIDI 0x00000010 +# define TMR_MODE_FSK 0x00000020 +# define TMR_MODE_CLS 0x00000040 +# define TMR_MODE_SMPTE 0x00000080 +#define SNDCTL_TMR_METRONOME __SIOW ('T', 7, int) +#define SNDCTL_TMR_SELECT __SIOW ('T', 8, int) + +/* + * Sample loading mechanism for internal synthesizers (/dev/sequencer) + * (for the .PAT format). + */ + +struct patch_info +{ + unsigned short key; /* Use WAVE_PATCH here */ +#define WAVE_PATCH _PATCHKEY(0x04) +#define GUS_PATCH WAVE_PATCH +#define WAVEFRONT_PATCH _PATCHKEY(0x06) + + short device_no; /* Synthesizer number */ + short instr_no; /* Midi pgm# */ + + unsigned int mode; +/* + * The least significant byte has the same format than the GUS .PAT + * files + */ +#define WAVE_16_BITS 0x01 /* bit 0 = 8 or 16 bit wave data. */ +#define WAVE_UNSIGNED 0x02 /* bit 1 = Signed - Unsigned data. */ +#define WAVE_LOOPING 0x04 /* bit 2 = looping enabled-1. */ +#define WAVE_BIDIR_LOOP 0x08 /* bit 3 = Set is bidirectional looping. */ +#define WAVE_LOOP_BACK 0x10 /* bit 4 = Set is looping backward. */ +#define WAVE_SUSTAIN_ON 0x20 /* bit 5 = Turn sustaining on. (Env. pts. 3) */ +#define WAVE_ENVELOPES 0x40 /* bit 6 = Enable envelopes - 1 */ +#define WAVE_FAST_RELEASE 0x80 /* bit 7 = Shut off immediately after note off */ + /* (use the env_rate/env_offs fields). */ +/* Linux specific bits */ +#define WAVE_VIBRATO 0x00010000 /* The vibrato info is valid */ +#define WAVE_TREMOLO 0x00020000 /* The tremolo info is valid */ +#define WAVE_SCALE 0x00040000 /* The scaling info is valid */ +#define WAVE_FRACTIONS 0x00080000 /* Fraction information is valid */ +/* Reserved bits */ +#define WAVE_ROM 0x40000000 /* For future use */ +#define WAVE_MULAW 0x20000000 /* For future use */ +/* Other bits must be zeroed */ + + int len; /* Size of the wave data in bytes */ + int loop_start, loop_end; /* Byte offsets from the beginning */ + +/* + * The base_freq and base_note fields are used when computing the + * playback speed for a note. The base_note defines the tone frequency + * which is heard if the sample is played using the base_freq as the + * playback speed. + * + * The low_note and high_note fields define the minimum and maximum note + * frequencies for which this sample is valid. It is possible to define + * more than one samples for an instrument number at the same time. The + * low_note and high_note fields are used to select the most suitable one. + * + * The fields base_note, high_note and low_note should contain + * the note frequency multiplied by 1000. For example value for the + * middle A is 440*1000. + */ + + unsigned int base_freq; + unsigned int base_note; + unsigned int high_note; + unsigned int low_note; + int panning; /* -128=left, 127=right */ + int detuning; + + /* Envelope. Enabled by mode bit WAVE_ENVELOPES */ + unsigned char env_rate[6]; /* GUS HW ramping rate */ + unsigned char env_offset[6]; /* 255 == 100% */ + + /* + * The tremolo, vibrato and scale info are not supported yet. + * Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or + * WAVE_SCALE + */ + + unsigned char tremolo_sweep; + unsigned char tremolo_rate; + unsigned char tremolo_depth; + + unsigned char vibrato_sweep; + unsigned char vibrato_rate; + unsigned char vibrato_depth; + + int scale_frequency; + unsigned int scale_factor; /* from 0 to 2048 or 0 to 2 */ + + int volume; + int fractions; + int reserved1; + int spare[2]; + char data[1]; /* The waveform data starts here */ +}; + +struct sysex_info +{ + short key; /* Use SYSEX_PATCH or MAUI_PATCH here */ +#define SYSEX_PATCH _PATCHKEY(0x05) +#define MAUI_PATCH _PATCHKEY(0x06) + short device_no; /* Synthesizer number */ + int len; /* Size of the sysex data in bytes */ + unsigned char data[1]; /* Sysex data starts here */ +}; + +/* + * /dev/sequencer input events. + * + * The data written to the /dev/sequencer is a stream of events. Events + * are records of 4 or 8 bytes. The first byte defines the size. + * Any number of events can be written with a write call. There + * is a set of macros for sending these events. Use these macros if you + * want to maximize portability of your program. + * + * Events SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO. Are also input events. + * (All input events are currently 4 bytes long. Be prepared to support + * 8 byte events also. If you receive any event having first byte >= 128, + * it's a 8 byte event. + * + * The events are documented at the end of this file. + * + * Normal events (4 bytes) + * There is also a 8 byte version of most of the 4 byte events. The + * 8 byte one is recommended. + * + * NOTE! All 4 byte events are now obsolete. Applications should not write + * them. However 4 byte events are still used as inputs from + * /dev/sequencer (/dev/music uses only 8 byte ones). + */ +#define SEQ_NOTEOFF 0 +#define SEQ_FMNOTEOFF SEQ_NOTEOFF /* Just old name */ +#define SEQ_NOTEON 1 +#define SEQ_FMNOTEON SEQ_NOTEON +#define SEQ_WAIT TMR_WAIT_ABS +#define SEQ_PGMCHANGE 3 +#define SEQ_FMPGMCHANGE SEQ_PGMCHANGE +#define SEQ_SYNCTIMER TMR_START +#define SEQ_MIDIPUTC 5 +#define SEQ_DRUMON 6 /*** OBSOLETE ***/ +#define SEQ_DRUMOFF 7 /*** OBSOLETE ***/ +#define SEQ_ECHO TMR_ECHO /* For synching programs with output */ +#define SEQ_AFTERTOUCH 9 +#define SEQ_CONTROLLER 10 +#define SEQ_BALANCE 11 +#define SEQ_VOLMODE 12 + +/************************************ + * Midi controller numbers * + ************************************/ +/* + * Controllers 0 to 31 (0x00 to 0x1f) and + * 32 to 63 (0x20 to 0x3f) are continuous + * controllers. + * In the MIDI 1.0 these controllers are sent using + * two messages. Controller numbers 0 to 31 are used + * to send the MSB and the controller numbers 32 to 63 + * are for the LSB. Note that just 7 bits are used in MIDI bytes. + */ + +#define CTL_BANK_SELECT 0x00 +#define CTL_MODWHEEL 0x01 +#define CTL_BREATH 0x02 +/* undefined 0x03 */ +#define CTL_FOOT 0x04 +#define CTL_PORTAMENTO_TIME 0x05 +#define CTL_DATA_ENTRY 0x06 +#define CTL_MAIN_VOLUME 0x07 +#define CTL_BALANCE 0x08 +/* undefined 0x09 */ +#define CTL_PAN 0x0a +#define CTL_EXPRESSION 0x0b +/* undefined 0x0c */ +/* undefined 0x0d */ +/* undefined 0x0e */ +/* undefined 0x0f */ +#define CTL_GENERAL_PURPOSE1 0x10 +#define CTL_GENERAL_PURPOSE2 0x11 +#define CTL_GENERAL_PURPOSE3 0x12 +#define CTL_GENERAL_PURPOSE4 0x13 +/* undefined 0x14 - 0x1f */ + +/* undefined 0x20 */ +/* The controller numbers 0x21 to 0x3f are reserved for the */ +/* least significant bytes of the controllers 0x00 to 0x1f. */ +/* These controllers are not recognised by the driver. */ + +/* Controllers 64 to 69 (0x40 to 0x45) are on/off switches. */ +/* 0=OFF and 127=ON (intermediate values are possible) */ +#define CTL_DAMPER_PEDAL 0x40 +#define CTL_SUSTAIN 0x40 /* Alias */ +#define CTL_HOLD 0x40 /* Alias */ +#define CTL_PORTAMENTO 0x41 +#define CTL_SOSTENUTO 0x42 +#define CTL_SOFT_PEDAL 0x43 +/* undefined 0x44 */ +#define CTL_HOLD2 0x45 +/* undefined 0x46 - 0x4f */ + +#define CTL_GENERAL_PURPOSE5 0x50 +#define CTL_GENERAL_PURPOSE6 0x51 +#define CTL_GENERAL_PURPOSE7 0x52 +#define CTL_GENERAL_PURPOSE8 0x53 +/* undefined 0x54 - 0x5a */ +#define CTL_EXT_EFF_DEPTH 0x5b +#define CTL_TREMOLO_DEPTH 0x5c +#define CTL_CHORUS_DEPTH 0x5d +#define CTL_DETUNE_DEPTH 0x5e +#define CTL_CELESTE_DEPTH 0x5e /* Alias for the above one */ +#define CTL_PHASER_DEPTH 0x5f +#define CTL_DATA_INCREMENT 0x60 +#define CTL_DATA_DECREMENT 0x61 +#define CTL_NONREG_PARM_NUM_LSB 0x62 +#define CTL_NONREG_PARM_NUM_MSB 0x63 +#define CTL_REGIST_PARM_NUM_LSB 0x64 +#define CTL_REGIST_PARM_NUM_MSB 0x65 +/* undefined 0x66 - 0x78 */ +/* reserved 0x79 - 0x7f */ + +/* Pseudo controllers (not midi compatible) */ +#define CTRL_PITCH_BENDER 255 +#define CTRL_PITCH_BENDER_RANGE 254 +#define CTRL_EXPRESSION 253 /* Obsolete */ +#define CTRL_MAIN_VOLUME 252 /* Obsolete */ + +/* + * Volume mode defines how volumes are used + */ + +#define VOL_METHOD_ADAGIO 1 +#define VOL_METHOD_LINEAR 2 + +/* + * Note! SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO are used also as + * input events. + */ + +/* + * Event codes 0xf0 to 0xfc are reserved for future extensions. + */ + +#define SEQ_FULLSIZE 0xfd /* Long events */ +/* + * SEQ_FULLSIZE events are used for loading patches/samples to the + * synthesizer devices. These events are passed directly to the driver + * of the associated synthesizer device. There is no limit to the size + * of the extended events. These events are not queued but executed + * immediately when the write() is called (execution can take several + * seconds of time). + * + * When a SEQ_FULLSIZE message is written to the device, it must + * be written using exactly one write() call. Other events cannot + * be mixed to the same write. + * + * For FM synths (YM3812/OPL3) use struct sbi_instrument and write it to the + * /dev/sequencer. Don't write other data together with the instrument structure + * Set the key field of the structure to FM_PATCH. The device field is used to + * route the patch to the corresponding device. + * + * For wave table use struct patch_info. Initialize the key field + * to WAVE_PATCH. + */ +#define SEQ_PRIVATE 0xfe /* Low level HW dependent events (8 bytes) */ +#define SEQ_EXTENDED 0xff /* Extended events (8 bytes) OBSOLETE */ + +/* + * Record for FM patches + */ + +typedef unsigned char sbi_instr_data[32]; + +struct sbi_instrument +{ + unsigned short key; /* FM_PATCH or OPL3_PATCH */ +#define FM_PATCH _PATCHKEY(0x01) +#define OPL3_PATCH _PATCHKEY(0x03) + short device; /* Synth# (0-4) */ + int channel; /* Program# to be initialized */ + sbi_instr_data operators; /* Register settings for operator cells (.SBI format) */ +}; + +struct synth_info +{ /* Read only */ + char name[30]; + int device; /* 0-N. INITIALIZE BEFORE CALLING */ + int synth_type; +#define SYNTH_TYPE_FM 0 +#define SYNTH_TYPE_SAMPLE 1 +#define SYNTH_TYPE_MIDI 2 /* Midi interface */ + + int synth_subtype; +#define FM_TYPE_ADLIB 0x00 +#define FM_TYPE_OPL3 0x01 +#define MIDI_TYPE_MPU401 0x401 + +#define SAMPLE_TYPE_BASIC 0x10 +#define SAMPLE_TYPE_GUS SAMPLE_TYPE_BASIC +#define SAMPLE_TYPE_WAVEFRONT 0x11 + + int perc_mode; /* No longer supported */ + int nr_voices; + int nr_drums; /* Obsolete field */ + int instr_bank_size; + unsigned int capabilities; +#define SYNTH_CAP_PERCMODE 0x00000001 /* No longer used */ +#define SYNTH_CAP_OPL3 0x00000002 /* Set if OPL3 supported */ +#define SYNTH_CAP_INPUT 0x00000004 /* Input (MIDI) device */ + int dummies[19]; /* Reserve space */ +}; + +struct sound_timer_info +{ + char name[32]; + int caps; +}; + +struct midi_info /* OBSOLETE */ +{ + char name[30]; + int device; /* 0-N. INITIALIZE BEFORE CALLING */ + unsigned int capabilities; /* To be defined later */ + int dev_type; + int dummies[18]; /* Reserve space */ +}; + +/* + * Level 2 event types for /dev/sequencer + */ + +/* + * The 4 most significant bits of byte 0 specify the class of + * the event: + * + * 0x8X = system level events, + * 0x9X = device/port specific events, event[1] = device/port, + * The last 4 bits give the subtype: + * 0x02 = Channel event (event[3] = chn). + * 0x01 = note event (event[4] = note). + * (0x01 is not used alone but always with bit 0x02). + * event[2] = MIDI message code (0x80=note off etc.) + * + */ + +#define EV_SEQ_LOCAL 0x80 +#define EV_TIMING 0x81 +#define EV_CHN_COMMON 0x92 +#define EV_CHN_VOICE 0x93 +#define EV_SYSEX 0x94 +#define EV_SYSTEM 0x95 /* MIDI system and real time messages (input only) */ +/* + * Event types 200 to 220 are reserved for application use. + * These numbers will not be used by the driver. + */ + +/* + * Events for event type EV_CHN_VOICE + */ + +#define MIDI_NOTEOFF 0x80 +#define MIDI_NOTEON 0x90 +#define MIDI_KEY_PRESSURE 0xA0 + +/* + * Events for event type EV_CHN_COMMON + */ + +#define MIDI_CTL_CHANGE 0xB0 +#define MIDI_PGM_CHANGE 0xC0 +#define MIDI_CHN_PRESSURE 0xD0 +#define MIDI_PITCH_BEND 0xE0 + +#define MIDI_SYSTEM_PREFIX 0xF0 + +/* + * Timer event types + */ +#define TMR_WAIT_REL 1 /* Time relative to the prev time */ +#define TMR_WAIT_ABS 2 /* Absolute time since TMR_START */ +#define TMR_STOP 3 +#define TMR_START 4 +#define TMR_CONTINUE 5 +#define TMR_TEMPO 6 +#define TMR_ECHO 8 +#define TMR_CLOCK 9 /* MIDI clock */ +#define TMR_SPP 10 /* Song position pointer */ +#define TMR_TIMESIG 11 /* Time signature */ + +/* + * Local event types + */ +#define LOCL_STARTAUDIO 1 +#define LOCL_STARTAUDIO2 2 +#define LOCL_STARTAUDIO3 3 +#define LOCL_STARTAUDIO4 4 + +#if (!defined(__KERNEL__) && !defined(KERNEL) && !defined(INKERNEL) && !defined(_KERNEL)) || defined(USE_SEQ_MACROS) +/* + * Some convenience macros to simplify programming of the + * /dev/sequencer interface + * + * These macros define the API which should be used when possible. + */ +#define SEQ_DECLAREBUF() SEQ_USE_EXTBUF() + +void seqbuf_dump (void); /* This function must be provided by programs */ + +EXTERNC int OSS_init (int seqfd, int buflen); +EXTERNC void OSS_seqbuf_dump (int fd, unsigned char *buf, int buflen); +EXTERNC void OSS_seq_advbuf (int len, int fd, unsigned char *buf, int buflen); +EXTERNC void OSS_seq_needbuf (int len, int fd, unsigned char *buf, + int buflen); +EXTERNC void OSS_patch_caching (int dev, int chn, int patch, int fd, + unsigned char *buf, int buflen); +EXTERNC void OSS_drum_caching (int dev, int chn, int patch, int fd, + unsigned char *buf, int buflen); +EXTERNC void OSS_write_patch (int fd, unsigned char *buf, int len); +EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len); + +#define SEQ_PM_DEFINES int __foo_bar___ +#ifdef OSSLIB +# define SEQ_USE_EXTBUF() \ + EXTERNC unsigned char *_seqbuf; \ + EXTERNC int _seqbuflen;EXTERNC int _seqbufptr +# define SEQ_DEFINEBUF(len) SEQ_USE_EXTBUF();static int _requested_seqbuflen=len +# define _SEQ_ADVBUF(len) OSS_seq_advbuf(len, seqfd, _seqbuf, _seqbuflen) +# define _SEQ_NEEDBUF(len) OSS_seq_needbuf(len, seqfd, _seqbuf, _seqbuflen) +# define SEQ_DUMPBUF() OSS_seqbuf_dump(seqfd, _seqbuf, _seqbuflen) + +# define SEQ_LOAD_GMINSTR(dev, instr) \ + OSS_patch_caching(dev, -1, instr, seqfd, _seqbuf, _seqbuflen) +# define SEQ_LOAD_GMDRUM(dev, drum) \ + OSS_drum_caching(dev, -1, drum, seqfd, _seqbuf, _seqbuflen) +#else /* !OSSLIB */ + +# define SEQ_LOAD_GMINSTR(dev, instr) +# define SEQ_LOAD_GMDRUM(dev, drum) + +# define SEQ_USE_EXTBUF() \ + EXTERNC unsigned char _seqbuf[]; \ + EXTERNC int _seqbuflen;EXTERNC int _seqbufptr + +#ifndef USE_SIMPLE_MACROS +/* Sample seqbuf_dump() implementation: + * + * SEQ_DEFINEBUF (2048); -- Defines a buffer for 2048 bytes + * + * int seqfd; -- The file descriptor for /dev/sequencer. + * + * void + * seqbuf_dump () + * { + * if (_seqbufptr) + * if (write (seqfd, _seqbuf, _seqbufptr) == -1) + * { + * perror ("write /dev/sequencer"); + * exit (-1); + * } + * _seqbufptr = 0; + * } + */ + +#define SEQ_DEFINEBUF(len) \ + unsigned char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0 +#define _SEQ_NEEDBUF(len) \ + if ((_seqbufptr+(len)) > _seqbuflen) seqbuf_dump() +#define _SEQ_ADVBUF(len) _seqbufptr += len +#define SEQ_DUMPBUF seqbuf_dump +#else +/* + * This variation of the sequencer macros is used just to format one event + * using fixed buffer. + * + * The program using the macro library must define the following macros before + * using this library. + * + * #define _seqbuf name of the buffer (unsigned char[]) + * #define _SEQ_ADVBUF(len) If the applic needs to know the exact + * size of the event, this macro can be used. + * Otherwise this must be defined as empty. + * #define _seqbufptr Define the name of index variable or 0 if + * not required. + */ +#define _SEQ_NEEDBUF(len) /* empty */ +#endif +#endif /* !OSSLIB */ + +#define SEQ_VOLUME_MODE(dev, mode) \ + {_SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = SEQ_EXTENDED;\ + _seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\ + _seqbuf[_seqbufptr+2] = (dev);\ + _seqbuf[_seqbufptr+3] = (mode);\ + _seqbuf[_seqbufptr+4] = 0;\ + _seqbuf[_seqbufptr+5] = 0;\ + _seqbuf[_seqbufptr+6] = 0;\ + _seqbuf[_seqbufptr+7] = 0;\ + _SEQ_ADVBUF(8);} + +/* + * Midi voice messages + */ + +#define _CHN_VOICE(dev, event, chn, note, parm) \ + {_SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = EV_CHN_VOICE;\ + _seqbuf[_seqbufptr+1] = (dev);\ + _seqbuf[_seqbufptr+2] = (event);\ + _seqbuf[_seqbufptr+3] = (chn);\ + _seqbuf[_seqbufptr+4] = (note);\ + _seqbuf[_seqbufptr+5] = (parm);\ + _seqbuf[_seqbufptr+6] = (0);\ + _seqbuf[_seqbufptr+7] = 0;\ + _SEQ_ADVBUF(8);} + +#define SEQ_START_NOTE(dev, chn, note, vol) \ + _CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol) + +#define SEQ_STOP_NOTE(dev, chn, note, vol) \ + _CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol) + +#define SEQ_KEY_PRESSURE(dev, chn, note, pressure) \ + _CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure) + +/* + * Midi channel messages + */ + +#define _CHN_COMMON(dev, event, chn, p1, p2, w14) \ + {_SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = EV_CHN_COMMON;\ + _seqbuf[_seqbufptr+1] = (dev);\ + _seqbuf[_seqbufptr+2] = (event);\ + _seqbuf[_seqbufptr+3] = (chn);\ + _seqbuf[_seqbufptr+4] = (p1);\ + _seqbuf[_seqbufptr+5] = (p2);\ + *(short *)&_seqbuf[_seqbufptr+6] = (w14);\ + _SEQ_ADVBUF(8);} +/* + * SEQ_SYSEX permits sending of sysex messages. (It may look that it permits + * sending any MIDI bytes but it's absolutely not possible. Trying to do + * so _will_ cause problems with MPU401 intelligent mode). + * + * Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be + * sent by calling SEQ_SYSEX() several times (there must be no other events + * between them). First sysex fragment must have 0xf0 in the first byte + * and the last byte (buf[len-1] of the last fragment must be 0xf7. No byte + * between these sysex start and end markers cannot be larger than 0x7f. Also + * lengths of each fragments (except the last one) must be 6. + * + * Breaking the above rules may work with some MIDI ports but is likely to + * cause fatal problems with some other devices (such as MPU401). + */ +#define SEQ_SYSEX(dev, buf, len) \ + {int ii, ll=(len); \ + unsigned char *bufp=buf;\ + if (ll>6)ll=6;\ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = EV_SYSEX;\ + _seqbuf[_seqbufptr+1] = (dev);\ + for(ii=0;ii>8)&0xff);\ + _seqbuf[_seqbufptr+7] = 0;\ + _SEQ_ADVBUF(8);} +/* + * The following 5 macros are incorrectly implemented and obsolete. + * Use SEQ_BENDER and SEQ_CONTROL (with proper controller) instead. + */ +#define SEQ_PITCHBEND(dev, voice, value) \ + SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER, value) +#define SEQ_BENDER_RANGE(dev, voice, value) \ + SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER_RANGE, value) +#define SEQ_EXPRESSION(dev, voice, value) \ + SEQ_CONTROL(dev, voice, CTL_EXPRESSION, value*128) +#define SEQ_MAIN_VOLUME(dev, voice, value) \ + SEQ_CONTROL(dev, voice, CTL_MAIN_VOLUME, (value*16383)/100) +#define SEQ_PANNING(dev, voice, pos) \ + SEQ_CONTROL(dev, voice, CTL_PAN, (pos+128) / 2) + +/* + * Timing and syncronization macros + */ + +#define _TIMER_EVENT(ev, parm) {_SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr+0] = EV_TIMING; \ + _seqbuf[_seqbufptr+1] = (ev); \ + _seqbuf[_seqbufptr+2] = 0;\ + _seqbuf[_seqbufptr+3] = 0;\ + *(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \ + _SEQ_ADVBUF(8);} + +#define SEQ_START_TIMER() _TIMER_EVENT(TMR_START, 0) +#define SEQ_STOP_TIMER() _TIMER_EVENT(TMR_STOP, 0) +#define SEQ_CONTINUE_TIMER() _TIMER_EVENT(TMR_CONTINUE, 0) +#define SEQ_WAIT_TIME(ticks) _TIMER_EVENT(TMR_WAIT_ABS, ticks) +#define SEQ_DELTA_TIME(ticks) _TIMER_EVENT(TMR_WAIT_REL, ticks) +#define SEQ_ECHO_BACK(key) _TIMER_EVENT(TMR_ECHO, key) +#define SEQ_SET_TEMPO(value) _TIMER_EVENT(TMR_TEMPO, value) +#define SEQ_SONGPOS(pos) _TIMER_EVENT(TMR_SPP, pos) +#define SEQ_TIME_SIGNATURE(sig) _TIMER_EVENT(TMR_TIMESIG, sig) + +/* + * Local control events + */ + +#define _LOCAL_EVENT(ev, parm) {_SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \ + _seqbuf[_seqbufptr+1] = (ev); \ + _seqbuf[_seqbufptr+2] = 0;\ + _seqbuf[_seqbufptr+3] = 0;\ + *(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \ + _SEQ_ADVBUF(8);} + +#define SEQ_PLAYAUDIO(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO, devmask) +#define SEQ_PLAYAUDIO2(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO2, devmask) +#define SEQ_PLAYAUDIO3(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO3, devmask) +#define SEQ_PLAYAUDIO4(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO4, devmask) +/* + * Events for the level 1 interface only + */ + +#define SEQ_MIDIOUT(device, byte) {_SEQ_NEEDBUF(4);\ + _seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\ + _seqbuf[_seqbufptr+1] = (byte);\ + _seqbuf[_seqbufptr+2] = (device);\ + _seqbuf[_seqbufptr+3] = 0;\ + _SEQ_ADVBUF(4);} + +/* + * Patch loading. + */ +#ifdef OSSLIB +# define SEQ_WRPATCH(patchx, len) \ + OSS_write_patch(seqfd, (char*)(patchx), len) +# define SEQ_WRPATCH2(patchx, len) \ + OSS_write_patch2(seqfd, (char*)(patchx), len) +#else +# define SEQ_WRPATCH(patchx, len) \ + {if (_seqbufptr) SEQ_DUMPBUF();\ + if (write(seqfd, (char*)(patchx), len)==-1) \ + perror("Write patch: /dev/sequencer");} +# define SEQ_WRPATCH2(patchx, len) \ + (SEQ_DUMPBUF(), write(seqfd, (char*)(patchx), len)) +#endif + +#endif +#endif /* ifndef DISABLE_SEQUENCER */ + +/* + **************************************************************************** + * ioctl commands for the /dev/midi## + ****************************************************************************/ +#define SNDCTL_MIDI_PRETIME __SIOWR('m', 0, int) + +#if 0 +/* + * The SNDCTL_MIDI_MPUMODE and SNDCTL_MIDI_MPUCMD calls + * are completely obsolete. The hardware device (MPU-401 "intelligent mode" + * and compatibles) has disappeared from the market 10 years ago so there + * is no need for this stuff. The MPU-401 "UART" mode devices don't support + * this stuff. + */ +typedef struct +{ + unsigned char cmd; + char nr_args, nr_returns; + unsigned char data[30]; +} mpu_command_rec; + +#define SNDCTL_MIDI_MPUMODE __SIOWR('m', 1, int) +#define SNDCTL_MIDI_MPUCMD __SIOWR('m', 2, mpu_command_rec) +#endif + +/* + * SNDCTL_MIDI_MTCINPUT turns on a mode where OSS automatically inserts + * MTC quarter frame messages (F1 xx) to the input. + * The argument is the MTC mode: + * + * -1 = Turn MTC messages OFF (default) + * 24 = 24 FPS + * 25 = 25 FPS + * 29 = 30 FPS drop frame + * 30 = 30 FPS + * + * Note that 25 FPS mode is probably the only mode that is supported. Other + * modes may be supported in the future versions of OSS, 25 FPS is handy + * because it generates 25*4=100 quarter frame messages per second which + * matches the usual 100 HZ system timer rate). + * + * The quarter frame timer will be reset to 0:00:00:00.0 at the moment this + * ioctl is made. + */ +#define SNDCTL_MIDI_MTCINPUT __SIOWR('m', 3, int) + +/* + * MTC/SMPTE time code record (for future use) + */ +typedef struct +{ + unsigned char hours, minutes, seconds, frames, qframes; + char direction; +#define MTC_DIR_STOPPED 0 +#define MTC_DIR_FORWARD 1 +#define MTC_DIR_BACKWARD -1 + unsigned char time_code_type; + unsigned int flags; +} oss_mtc_data_t; + +#define SNDCTL_MIDI_SETMODE __SIOWR('m', 6, int) +# define MIDI_MODE_TRADITIONAL 0 +# define MIDI_MODE_TIMED 1 /* Input times are in MIDI ticks */ +# define MIDI_MODE_TIMED_ABS 2 /* Input times are absolute (usecs) */ + +/* + * Packet header for MIDI_MODE_TIMED and MIDI_MODE_TIMED_ABS + */ +typedef unsigned long long oss_midi_time_t; /* Variable type for MIDI time (clock ticks) */ + +typedef struct +{ + int magic; /* Initialize to MIDI_HDR_MAGIC */ +#define MIDI_HDR_MAGIC -1 + unsigned short event_type; +#define MIDI_EV_WRITE 0 /* Write or read (with payload) */ +#define MIDI_EV_TEMPO 1 +#define MIDI_EV_ECHO 2 +#define MIDI_EV_START 3 +#define MIDI_EV_STOP 4 +#define MIDI_EV_CONTINUE 5 +#define MIDI_EV_XPRESSWRITE 6 +#define MIDI_EV_TIMEBASE 7 +#define MIDI_EV_DEVCTL 8 /* Device control read/write */ + unsigned short options; +#define MIDI_OPT_NONE 0x0000 +#define MIDI_OPT_TIMED 0x0001 +#define MIDI_OPT_CONTINUATION 0x0002 +#define MIDI_OPT_USECTIME 0x0004 /* Time is absolute (in usecs) */ +#define MIDI_OPT_BUSY 0x0008 /* Reserved for internal use */ + oss_midi_time_t time; + int parm; + int filler[3]; /* Fur future expansion - init to zeros */ +} midi_packet_header_t; +/* + * MIDI_PAYLOAD_SIZE is the maximum size of one MIDI input chunk. It must be + * less (or equal) than 1024 which is the read size recommended in the + * documentation. TODO: Explain this better. + */ +#define MIDI_PAYLOAD_SIZE 1000 + +typedef struct +{ + midi_packet_header_t hdr; + unsigned char payload[MIDI_PAYLOAD_SIZE]; +} midi_packet_t; + +#define SNDCTL_MIDI_TIMEBASE __SIOWR('m', 7, int) +#define SNDCTL_MIDI_TEMPO __SIOWR('m', 8, int) +/* + * User land MIDI servers (synths) can use SNDCTL_MIDI_SET_LATENCY + * to request MIDI events to be sent to them in advance. The parameter + * (in microseconds) tells how much before the events are submitted. + * + * This feature is only valid for loopback devices and possibly some other + * types of virtual devices. + */ +#define SNDCTL_MIDI_SET_LATENCY __SIOW ('m', 9, int) +/* + **************************************************************************** + * IOCTL commands for /dev/dsp + ****************************************************************************/ + +#define SNDCTL_DSP_HALT __SIO ('P', 0) +#define SNDCTL_DSP_RESET SNDCTL_DSP_HALT /* Old name */ +#define SNDCTL_DSP_SYNC __SIO ('P', 1) +#define SNDCTL_DSP_SPEED __SIOWR('P', 2, int) + +/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */ +#define SNDCTL_DSP_STEREO __SIOWR('P', 3, int) +/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */ + +#define SNDCTL_DSP_GETBLKSIZE __SIOWR('P', 4, int) +#define SNDCTL_DSP_SAMPLESIZE SNDCTL_DSP_SETFMT +#define SNDCTL_DSP_CHANNELS __SIOWR('P', 6, int) +#define SNDCTL_DSP_POST __SIO ('P', 8) +#define SNDCTL_DSP_SUBDIVIDE __SIOWR('P', 9, int) +#define SNDCTL_DSP_SETFRAGMENT __SIOWR('P',10, int) + +/* Audio data formats (Note! U8=8 and S16_LE=16 for compatibility) */ +#define SNDCTL_DSP_GETFMTS __SIOR ('P',11, int) /* Returns a mask */ +#define SNDCTL_DSP_SETFMT __SIOWR('P',5, int) /* Selects ONE fmt */ +# define AFMT_QUERY 0x00000000 /* Return current fmt */ +# define AFMT_MU_LAW 0x00000001 +# define AFMT_A_LAW 0x00000002 +# define AFMT_IMA_ADPCM 0x00000004 +# define AFMT_U8 0x00000008 +# define AFMT_S16_LE 0x00000010 /* Little endian signed 16 */ +# define AFMT_S16_BE 0x00000020 /* Big endian signed 16 */ +# define AFMT_S8 0x00000040 +# define AFMT_U16_LE 0x00000080 /* Little endian U16 */ +# define AFMT_U16_BE 0x00000100 /* Big endian U16 */ +# define AFMT_MPEG 0x00000200 /* MPEG (2) audio */ + +/* AC3 _compressed_ bitstreams (See Programmer's Guide for details). */ +# define AFMT_AC3 0x00000400 +/* Ogg Vorbis _compressed_ bit streams */ +# define AFMT_VORBIS 0x00000800 + +/* 32 bit formats (MSB aligned) formats */ +# define AFMT_S32_LE 0x00001000 +# define AFMT_S32_BE 0x00002000 + +/* Reserved for _native_ endian double precision IEEE floating point */ +# define AFMT_FLOAT 0x00004000 + +/* 24 bit formats (LSB aligned in 32 bit word) formats */ +# define AFMT_S24_LE 0x00008000 +# define AFMT_S24_BE 0x00010000 + +/* + * S/PDIF raw format. In this format the S/PDIF frames (including all + * control and user bits) are included in the data stream. Each sample + * is stored in a 32 bit frame (see IEC-958 for more info). This format + * is supported by very few devices and it's only usable for purposes + * where full access to the control/user bits is required (real time control). + */ +# define AFMT_SPDIF_RAW 0x00020000 + +/* 24 bit packed (3 byte) little endian format (USB compatibility) */ +# define AFMT_S24_PACKED 0x00040000 + + +/* + * Some big endian/little endian handling macros (native endian and opposite + * endian formats). The usage of these macros is described in the OSS + * Programmer's Manual. + */ + +#if defined(_AIX) || defined(AIX) || defined(sparc) || defined(__hppa) || defined(PPC) || defined(__powerpc__) && !defined(i386) && !defined(__i386) && !defined(__i386__) + +/* Big endian machines */ +# define _PATCHKEY(id) (0xfd00|id) +# define AFMT_S16_NE AFMT_S16_BE +# define AFMT_U16_NE AFMT_U16_BE +# define AFMT_S32_NE AFMT_S32_BE +# define AFMT_S24_NE AFMT_S24_BE +# define AFMT_S16_OE AFMT_S16_LE +# define AFMT_S32_OE AFMT_S32_LE +# define AFMT_S24_OE AFMT_S24_LE +#else +# define _PATCHKEY(id) ((id<<8)|0xfd) +# define AFMT_S16_NE AFMT_S16_LE +# define AFMT_U16_NE AFMT_U16_LE +# define AFMT_S32_NE AFMT_S32_LE +# define AFMT_S24_NE AFMT_S24_LE +# define AFMT_S16_OE AFMT_S16_BE +# define AFMT_S32_OE AFMT_S32_BE +# define AFMT_S24_OE AFMT_S24_BE +#endif +/* + * Buffer status queries. + */ +typedef struct audio_buf_info +{ + int fragments; /* # of available fragments (partially usend ones not counted) */ + int fragstotal; /* Total # of fragments allocated */ + int fragsize; /* Size of a fragment in bytes */ + int bytes; /* Available space in bytes (includes partially used fragments) */ + /* Note! 'bytes' could be more than fragments*fragsize */ +} audio_buf_info; + +#define SNDCTL_DSP_GETOSPACE __SIOR ('P',12, audio_buf_info) +#define SNDCTL_DSP_GETISPACE __SIOR ('P',13, audio_buf_info) +#define SNDCTL_DSP_GETCAPS __SIOR ('P',15, int) +# define PCM_CAP_REVISION 0x000000ff /* Bits for revision level (0 to 255) */ +# define PCM_CAP_DUPLEX 0x00000100 /* Full duplex record/playback */ +# define PCM_CAP_REALTIME 0x00000200 /* Not in use */ +# define PCM_CAP_BATCH 0x00000400 /* Device has some kind of */ + /* internal buffers which may */ + /* cause some delays and */ + /* decrease precision of timing */ +# define PCM_CAP_COPROC 0x00000800 /* Has a coprocessor */ + /* Sometimes it's a DSP */ + /* but usually not */ +# define PCM_CAP_TRIGGER 0x00001000 /* Supports SETTRIGGER */ +# define PCM_CAP_MMAP 0x00002000 /* Supports mmap() */ +# define PCM_CAP_MULTI 0x00004000 /* Supports multiple open */ +# define PCM_CAP_BIND 0x00008000 /* Supports binding to front/rear/center/lfe */ +# define PCM_CAP_INPUT 0x00010000 /* Supports recording */ +# define PCM_CAP_OUTPUT 0x00020000 /* Supports playback */ +# define PCM_CAP_VIRTUAL 0x00040000 /* Virtuial device */ +/* 0x00040000 and 0x00080000 reserved for future use */ + +/* Analog/digital control capabilities */ +# define PCM_CAP_ANALOGOUT 0x00100000 +# define PCM_CAP_ANALOGIN 0x00200000 +# define PCM_CAP_DIGITALOUT 0x00400000 +# define PCM_CAP_DIGITALIN 0x00800000 +# define PCM_CAP_ADMASK 0x00f00000 +/* + * NOTE! (capabilities & PCM_CAP_ADMASK)==0 means just that the + * digital/analog interface control features are not supported by the + * device/driver. However the device still supports analog, digital or + * both inputs/outputs (depending on the device). See the OSS Programmer's + * Guide for full details. + */ +# define PCM_CAP_SHADOW 0x01000000 /* "Shadow" device */ + +/* + * Preferred channel usage. These bits can be used to + * give recommendations to the application. Used by few drivers. + * For example if ((caps & DSP_CH_MASK) == DSP_CH_MONO) means that + * the device works best in mono mode. However it doesn't necessarily mean + * that the device cannot be used in stereo. These bits should only be used + * special applications such as multi track hard disk recorders to find out + * the initial setup. However the user should be able to override this + * selection. + * + * To find out which modes are actually supported the application should + * try to select them using SNDCTL_DSP_CHANNELS. + */ +# define DSP_CH_MASK 0x06000000 /* Mask */ +# define DSP_CH_ANY 0x00000000 /* No preferred mode */ +# define DSP_CH_MONO 0x02000000 +# define DSP_CH_STEREO 0x04000000 +# define DSP_CH_MULTI 0x06000000 /* More than two channels */ + +# define PCM_CAP_HIDDEN 0x08000000 /* Hidden device */ +# define PCM_CAP_FREERATE 0x10000000 +# define PCM_CAP_MODEM 0x20000000 /* Modem device */ +# define PCM_CAP_DEFAULT 0x40000000 /* "Default" device */ + +/* + * The PCM_CAP_* capability names were known as DSP_CAP_* prior OSS 4.0 + * so it's necessary to define the older names too. + */ +#define DSP_CAP_ADMASK PCM_CAP_ADMASK +#define DSP_CAP_ANALOGIN PCM_CAP_ANALOGIN +#define DSP_CAP_ANALOGOUT PCM_CAP_ANALOGOUT +#define DSP_CAP_BATCH PCM_CAP_BATCH +#define DSP_CAP_BIND PCM_CAP_BIND +#define DSP_CAP_COPROC PCM_CAP_COPROC +#define DSP_CAP_DEFAULT PCM_CAP_DEFAULT +#define DSP_CAP_DIGITALIN PCM_CAP_DIGITALIN +#define DSP_CAP_DIGITALOUT PCM_CAP_DIGITALOUT +#define DSP_CAP_DUPLEX PCM_CAP_DUPLEX +#define DSP_CAP_FREERATE PCM_CAP_FREERATE +#define DSP_CAP_HIDDEN PCM_CAP_HIDDEN +#define DSP_CAP_INPUT PCM_CAP_INPUT +#define DSP_CAP_MMAP PCM_CAP_MMAP +#define DSP_CAP_MODEM PCM_CAP_MODEM +#define DSP_CAP_MULTI PCM_CAP_MULTI +#define DSP_CAP_OUTPUT PCM_CAP_OUTPUT +#define DSP_CAP_REALTIME PCM_CAP_REALTIME +#define DSP_CAP_REVISION PCM_CAP_REVISION +#define DSP_CAP_SHADOW PCM_CAP_SHADOW +#define DSP_CAP_TRIGGER PCM_CAP_TRIGGER +#define DSP_CAP_VIRTUAL PCM_CAP_VIRTUAL + +#define SNDCTL_DSP_GETTRIGGER __SIOR ('P',16, int) +#define SNDCTL_DSP_SETTRIGGER __SIOW ('P',16, int) +# define PCM_ENABLE_INPUT 0x00000001 +# define PCM_ENABLE_OUTPUT 0x00000002 + +typedef struct count_info +{ + unsigned int bytes; /* Total # of bytes processed */ + int blocks; /* # of fragment transitions since last time */ + int ptr; /* Current DMA pointer value */ +} count_info; + +#define SNDCTL_DSP_GETIPTR __SIOR ('P',17, count_info) +#define SNDCTL_DSP_GETOPTR __SIOR ('P',18, count_info) + +typedef struct buffmem_desc +{ + unsigned *buffer; + int size; +} buffmem_desc; +#define SNDCTL_DSP_SETSYNCRO __SIO ('P', 21) +#define SNDCTL_DSP_SETDUPLEX __SIO ('P', 22) + +#define SNDCTL_DSP_PROFILE __SIOW ('P', 23, int) /* OBSOLETE */ +#define APF_NORMAL 0 /* Normal applications */ +#define APF_NETWORK 1 /* Underruns probably caused by an "external" delay */ +#define APF_CPUINTENS 2 /* Underruns probably caused by "overheating" the CPU */ + +#define SNDCTL_DSP_GETODELAY __SIOR ('P', 23, int) + +typedef struct audio_errinfo +{ + int play_underruns; + int rec_overruns; + unsigned int play_ptradjust; + unsigned int rec_ptradjust; + int play_errorcount; + int rec_errorcount; + int play_lasterror; + int rec_lasterror; + int play_errorparm; + int rec_errorparm; + int filler[16]; +} audio_errinfo; + +#define SNDCTL_DSP_GETPLAYVOL __SIOR ('P', 24, int) +#define SNDCTL_DSP_SETPLAYVOL __SIOWR('P', 24, int) +#define SNDCTL_DSP_GETERROR __SIOR ('P', 25, audio_errinfo) +/* + **************************************************************************** + * Digital interface (S/PDIF) control interface + */ + +typedef struct oss_digital_control +{ + unsigned int caps; +#define DIG_CBITIN_NONE 0x00000000 +#define DIG_CBITIN_LIMITED 0x00000001 +#define DIG_CBITIN_DATA 0x00000002 +#define DIG_CBITIN_BYTE0 0x00000004 +#define DIG_CBITIN_FULL 0x00000008 +#define DIG_CBITIN_MASK 0x0000000f +#define DIG_CBITOUT_NONE 0x00000000 +#define DIG_CBITOUT_LIMITED 0x00000010 +#define DIG_CBITOUT_BYTE0 0x00000020 +#define DIG_CBITOUT_FULL 0x00000040 +#define DIG_CBITOUT_DATA 0x00000080 +#define DIG_CBITOUT_MASK 0x000000f0 +#define DIG_UBITIN 0x00000100 +#define DIG_UBITOUT 0x00000200 +#define DIG_VBITOUT 0x00000400 +#define DIG_OUTRATE 0x00000800 +#define DIG_INRATE 0x00001000 +#define DIG_INBITS 0x00002000 +#define DIG_OUTBITS 0x00004000 +#define DIG_EXACT 0x00010000 +#define DIG_PRO 0x00020000 +#define DIG_CONSUMER 0x00040000 +#define DIG_PASSTHROUGH 0x00080000 +#define DIG_OUTSEL 0x00100000 + + unsigned int valid; +#define VAL_CBITIN 0x00000001 +#define VAL_UBITIN 0x00000002 +#define VAL_CBITOUT 0x00000004 +#define VAL_UBITOUT 0x00000008 +#define VAL_ISTATUS 0x00000010 +#define VAL_IRATE 0x00000020 +#define VAL_ORATE 0x00000040 +#define VAL_INBITS 0x00000080 +#define VAL_OUTBITS 0x00000100 +#define VAL_REQUEST 0x00000200 +#define VAL_OUTSEL 0x00000400 + +#define VAL_OUTMASK (VAL_CBITOUT|VAL_UBITOUT|VAL_ORATE|VAL_OUTBITS|VAL_OUTSEL) + + unsigned int request, param; +#define SPD_RQ_PASSTHROUGH 1 + + unsigned char cbitin[24]; + unsigned char ubitin[24]; + unsigned char cbitout[24]; + unsigned char ubitout[24]; + + unsigned int outsel; +#define OUTSEL_DIGITAL 1 +#define OUTSEL_ANALOG 2 +#define OUTSEL_BOTH (OUTSEL_DIGITAL|OUTSEL_ANALOG) + + int in_data; /* Audio/data if autodetectable by the receiver */ +#define IND_UNKNOWN 0 +#define IND_AUDIO 1 +#define IND_DATA 2 + + int in_locked; /* Receiver locked */ +#define LOCK_NOT_INDICATED 0 +#define LOCK_UNLOCKED 1 +#define LOCK_LOCKED 2 + + int in_quality; /* Input signal quality */ +#define IN_QUAL_NOT_INDICATED 0 +#define IN_QUAL_POOR 1 +#define IN_QUAL_GOOD 2 + + int in_vbit, out_vbit; /* V bits */ +#define VBIT_NOT_INDICATED 0 +#define VBIT_OFF 1 +#define VBIT_ON 2 + + unsigned int in_errors; /* Various input errro conditions */ +#define INERR_CRC 0x0001 +#define INERR_QCODE_CRC 0x0002 +#define INERR_PARITY 0x0004 +#define INERR_BIPHASE 0x0008 + + int srate_in, srate_out; + int bits_in, bits_out; + + int filler[32]; +} oss_digital_control; + +#define SNDCTL_DSP_READCTL __SIOWR('P', 26, oss_digital_control) +#define SNDCTL_DSP_WRITECTL __SIOWR('P', 27, oss_digital_control) + +/* + **************************************************************************** + * Sync groups for audio devices + */ +typedef struct oss_syncgroup +{ + int id; + int mode; + int filler[16]; +} oss_syncgroup; + +#define SNDCTL_DSP_SYNCGROUP __SIOWR('P', 28, oss_syncgroup) +#define SNDCTL_DSP_SYNCSTART __SIOW ('P', 29, int) + +/* + ************************************************************************** + * "cooked" mode enables software based conversions for sample rate, sample + * format (bits) and number of channels (mono/stereo). These conversions are + * required with some devices that support only one sample rate or just stereo + * to let the applications to use other formats. The cooked mode is enabled by + * default. However it's necessary to disable this mode when mmap() is used or + * when very deterministic timing is required. SNDCTL_DSP_COOKEDMODE is an + * optional call introduced in OSS 3.9.6f. It's _error return must be ignored_ + * since normally this call will return erno=EINVAL. + * + * SNDCTL_DSP_COOKEDMODE must be called immediately after open before doing + * anything else. Otherwise the call will not have any effect. + */ +#define SNDCTL_DSP_COOKEDMODE __SIOW ('P', 30, int) + +/* + ************************************************************************** + * SNDCTL_DSP_SILENCE and SNDCTL_DSP_SKIP are new calls in OSS 3.99.0 + * that can be used to implement pause/continue during playback (no effect + * on recording). + */ +#define SNDCTL_DSP_SILENCE __SIO ('P', 31) +#define SNDCTL_DSP_SKIP __SIO ('P', 32) +/* + **************************************************************************** + * Abort transfer (reset) functions for input and output + */ +#define SNDCTL_DSP_HALT_INPUT __SIO ('P', 33) +#define SNDCTL_DSP_RESET_INPUT SNDCTL_DSP_HALT_INPUT /* Old name */ +#define SNDCTL_DSP_HALT_OUTPUT __SIO ('P', 34) +#define SNDCTL_DSP_RESET_OUTPUT SNDCTL_DSP_HALT_OUTPUT /* Old name */ +/* + **************************************************************************** + * Low water level control + */ +#define SNDCTL_DSP_LOW_WATER __SIOW ('P', 34, int) + +/* + **************************************************************************** + * 64 bit pointer support. Only available in environments that support + * the 64 bit (long long) integer type. + */ +#ifndef OSS_NO_LONG_LONG +typedef struct +{ + long long samples; + int fifo_samples; + int filler[32]; /* For future use */ +} oss_count_t; + +#define SNDCTL_DSP_CURRENT_IPTR __SIOR ('P', 35, oss_count_t) +#define SNDCTL_DSP_CURRENT_OPTR __SIOR ('P', 36, oss_count_t) +#endif + +/* + **************************************************************************** + * Interface for selecting recording sources and playback output routings. + */ +#define SNDCTL_DSP_GET_RECSRC_NAMES __SIOR ('P', 37, oss_mixer_enuminfo) +#define SNDCTL_DSP_GET_RECSRC __SIOR ('P', 38, int) +#define SNDCTL_DSP_SET_RECSRC __SIOWR('P', 38, int) + +#define SNDCTL_DSP_GET_PLAYTGT_NAMES __SIOR ('P', 39, oss_mixer_enuminfo) +#define SNDCTL_DSP_GET_PLAYTGT __SIOR ('P', 40, int) +#define SNDCTL_DSP_SET_PLAYTGT __SIOWR('P', 40, int) +#define SNDCTL_DSP_GETRECVOL __SIOR ('P', 41, int) +#define SNDCTL_DSP_SETRECVOL __SIOWR('P', 41, int) + +/* + *************************************************************************** + * Some calls for setting the channel assignment with multi channel devices + * (see the manual for details). + */ +#ifndef OSS_NO_LONG_LONG +#define SNDCTL_DSP_GET_CHNORDER __SIOR ('P', 42, unsigned long long) +#define SNDCTL_DSP_SET_CHNORDER __SIOWR('P', 42, unsigned long long) +# define CHID_UNDEF 0 +# define CHID_L 1 +# define CHID_R 2 +# define CHID_C 3 +# define CHID_LFE 4 +# define CHID_LS 5 +# define CHID_RS 6 +# define CHID_LR 7 +# define CHID_RR 8 +#define CHNORDER_UNDEF 0x0000000000000000ULL +#define CHNORDER_NORMAL 0x0000000087654321ULL +#endif + +#define MAX_PEAK_CHANNELS 128 +typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS]; +#define SNDCTL_DSP_GETIPEAKS __SIOR('P', 43, oss_peaks_t) +#define SNDCTL_DSP_GETOPEAKS __SIOR('P', 44, oss_peaks_t) + +#define SNDCTL_DSP_POLICY __SIOW('P', 45, int) /* See the manual */ + +/* + **************************************************************************** + * Few ioctl calls that are not official parts of OSS. They have been used + * by few freeware implementations of OSS. + */ +#define SNDCTL_DSP_GETCHANNELMASK __SIOWR('P', 64, int) +#define SNDCTL_DSP_BIND_CHANNEL __SIOWR('P', 65, int) +# define DSP_BIND_QUERY 0x00000000 +# define DSP_BIND_FRONT 0x00000001 +# define DSP_BIND_SURR 0x00000002 +# define DSP_BIND_CENTER_LFE 0x00000004 +# define DSP_BIND_HANDSET 0x00000008 +# define DSP_BIND_MIC 0x00000010 +# define DSP_BIND_MODEM1 0x00000020 +# define DSP_BIND_MODEM2 0x00000040 +# define DSP_BIND_I2S 0x00000080 +# define DSP_BIND_SPDIF 0x00000100 +# define DSP_BIND_REAR 0x00000200 + +#ifndef NO_LEGACY_MIXER +/* + **************************************************************************** + * IOCTL commands for the "legacy " /dev/mixer API (obsolete) + * + * Mixer controls + * + * There can be up to 20 different analog mixer channels. The + * SOUND_MIXER_NRDEVICES gives the currently supported maximum. + * The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells + * the devices supported by the particular mixer. + * + * {!notice This "legacy" mixer API is obsolete. It has been superceded + * by a new one (see below). + */ + +#define SOUND_MIXER_NRDEVICES 28 +#define SOUND_MIXER_VOLUME 0 +#define SOUND_MIXER_BASS 1 +#define SOUND_MIXER_TREBLE 2 +#define SOUND_MIXER_SYNTH 3 +#define SOUND_MIXER_PCM 4 +#define SOUND_MIXER_SPEAKER 5 +#define SOUND_MIXER_LINE 6 +#define SOUND_MIXER_MIC 7 +#define SOUND_MIXER_CD 8 +#define SOUND_MIXER_IMIX 9 /* Recording monitor */ +#define SOUND_MIXER_ALTPCM 10 +#define SOUND_MIXER_RECLEV 11 /* Recording level */ +#define SOUND_MIXER_IGAIN 12 /* Input gain */ +#define SOUND_MIXER_OGAIN 13 /* Output gain */ +/* + * Some soundcards have three line level inputs (line, aux1 and aux2). + * Since each card manufacturer has assigned different meanings to + * these inputs, it's impractical to assign specific meanings + * (eg line, cd, synth etc.) to them. + */ +#define SOUND_MIXER_LINE1 14 /* Input source 1 (aux1) */ +#define SOUND_MIXER_LINE2 15 /* Input source 2 (aux2) */ +#define SOUND_MIXER_LINE3 16 /* Input source 3 (line) */ +#define SOUND_MIXER_DIGITAL1 17 /* Digital I/O 1 */ +#define SOUND_MIXER_DIGITAL2 18 /* Digital I/O 2 */ +#define SOUND_MIXER_DIGITAL3 19 /* Digital I/O 3 */ +#define SOUND_MIXER_PHONE 20 /* Phone */ +#define SOUND_MIXER_MONO 21 /* Mono Output */ +#define SOUND_MIXER_VIDEO 22 /* Video/TV (audio) in */ +#define SOUND_MIXER_RADIO 23 /* Radio in */ +#define SOUND_MIXER_DEPTH 24 /* Surround depth */ +#define SOUND_MIXER_REARVOL 25 /* Rear/Surround speaker vol */ +#define SOUND_MIXER_CENTERVOL 26 /* Center/LFE speaker vol */ +#define SOUND_MIXER_SIDEVOL 27 /* Side-Surround (8speaker) vol */ + +/* + * Warning: SOUND_MIXER_SURRVOL is an old name of SOUND_MIXER_SIDEVOL. + * They are both assigned to the same mixer control. Don't + * use both control names in the same program/driver. + */ +#define SOUND_MIXER_SURRVOL SOUND_MIXER_SIDEVOL + +/* Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX) */ +/* Not counted to SOUND_MIXER_NRDEVICES, but use the same number space */ +#define SOUND_ONOFF_MIN 28 +#define SOUND_ONOFF_MAX 30 + +/* Note! Number 31 cannot be used since the sign bit is reserved */ +#define SOUND_MIXER_NONE 31 + +/* + * The following unsupported macros are no longer functional. + * Use SOUND_MIXER_PRIVATE# macros in future. + */ +#define SOUND_MIXER_ENHANCE SOUND_MIXER_NONE +#define SOUND_MIXER_MUTE SOUND_MIXER_NONE +#define SOUND_MIXER_LOUD SOUND_MIXER_NONE + +#define SOUND_DEVICE_LABELS \ + {"Vol ", "Bass ", "Treble", "Synth", "Pcm ", "Speaker ", "Line ", \ + "Mic ", "CD ", "Mix ", "Pcm2 ", "Rec ", "IGain", "OGain", \ + "Aux1", "Aux2", "Aux3", "Digital1", "Digital2", "Digital3", \ + "Phone", "Mono", "Video", "Radio", "Depth", \ + "Rear", "Center", "Side"} + +#define SOUND_DEVICE_NAMES \ + {"vol", "bass", "treble", "synth", "pcm", "speaker", "line", \ + "mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \ + "aux1", "aux2", "aux3", "dig1", "dig2", "dig3", \ + "phone", "mono", "video", "radio", "depth", \ + "rear", "center", "side"} + +/* Device bitmask identifiers */ + +#define SOUND_MIXER_RECSRC 0xff /* Arg contains a bit for each recording source */ +#define SOUND_MIXER_DEVMASK 0xfe /* Arg contains a bit for each supported device */ +#define SOUND_MIXER_RECMASK 0xfd /* Arg contains a bit for each supported recording source */ +#define SOUND_MIXER_CAPS 0xfc +# define SOUND_CAP_EXCL_INPUT 0x00000001 /* Only one recording source at a time */ +# define SOUND_CAP_NOLEGACY 0x00000004 /* For internal use only */ +# define SOUND_CAP_NORECSRC 0x00000008 +#define SOUND_MIXER_STEREODEVS 0xfb /* Mixer channels supporting stereo */ + +/* OSS/Free ONLY */ +#define SOUND_MIXER_OUTSRC 0xfa /* Arg contains a bit for each input source to output */ +#define SOUND_MIXER_OUTMASK 0xf9 /* Arg contains a bit for each supported input source to output */ +/* OSS/Free ONLY */ + +/* Device mask bits */ + +#define SOUND_MASK_VOLUME (1 << SOUND_MIXER_VOLUME) +#define SOUND_MASK_BASS (1 << SOUND_MIXER_BASS) +#define SOUND_MASK_TREBLE (1 << SOUND_MIXER_TREBLE) +#define SOUND_MASK_SYNTH (1 << SOUND_MIXER_SYNTH) +#define SOUND_MASK_PCM (1 << SOUND_MIXER_PCM) +#define SOUND_MASK_SPEAKER (1 << SOUND_MIXER_SPEAKER) +#define SOUND_MASK_LINE (1 << SOUND_MIXER_LINE) +#define SOUND_MASK_MIC (1 << SOUND_MIXER_MIC) +#define SOUND_MASK_CD (1 << SOUND_MIXER_CD) +#define SOUND_MASK_IMIX (1 << SOUND_MIXER_IMIX) +#define SOUND_MASK_ALTPCM (1 << SOUND_MIXER_ALTPCM) +#define SOUND_MASK_RECLEV (1 << SOUND_MIXER_RECLEV) +#define SOUND_MASK_IGAIN (1 << SOUND_MIXER_IGAIN) +#define SOUND_MASK_OGAIN (1 << SOUND_MIXER_OGAIN) +#define SOUND_MASK_LINE1 (1 << SOUND_MIXER_LINE1) +#define SOUND_MASK_LINE2 (1 << SOUND_MIXER_LINE2) +#define SOUND_MASK_LINE3 (1 << SOUND_MIXER_LINE3) +#define SOUND_MASK_DIGITAL1 (1 << SOUND_MIXER_DIGITAL1) +#define SOUND_MASK_DIGITAL2 (1 << SOUND_MIXER_DIGITAL2) +#define SOUND_MASK_DIGITAL3 (1 << SOUND_MIXER_DIGITAL3) +#define SOUND_MASK_MONO (1 << SOUND_MIXER_MONO) +#define SOUND_MASK_PHONE (1 << SOUND_MIXER_PHONE) +#define SOUND_MASK_RADIO (1 << SOUND_MIXER_RADIO) +#define SOUND_MASK_VIDEO (1 << SOUND_MIXER_VIDEO) +#define SOUND_MASK_DEPTH (1 << SOUND_MIXER_DEPTH) +#define SOUND_MASK_REARVOL (1 << SOUND_MIXER_REARVOL) +#define SOUND_MASK_CENTERVOL (1 << SOUND_MIXER_CENTERVOL) +#define SOUND_MASK_SIDEVOL (1 << SOUND_MIXER_SIDEVOL) + +/* Note! SOUND_MASK_SURRVOL is alias of SOUND_MASK_SIDEVOL */ +#define SOUND_MASK_SURRVOL (1 << SOUND_MIXER_SIDEVOL) + +/* Obsolete macros */ +#define SOUND_MASK_MUTE (1 << SOUND_MIXER_MUTE) +#define SOUND_MASK_ENHANCE (1 << SOUND_MIXER_ENHANCE) +#define SOUND_MASK_LOUD (1 << SOUND_MIXER_LOUD) + +#define MIXER_READ(dev) __SIOR('M', dev, int) +#define SOUND_MIXER_READ_VOLUME MIXER_READ(SOUND_MIXER_VOLUME) +#define SOUND_MIXER_READ_BASS MIXER_READ(SOUND_MIXER_BASS) +#define SOUND_MIXER_READ_TREBLE MIXER_READ(SOUND_MIXER_TREBLE) +#define SOUND_MIXER_READ_SYNTH MIXER_READ(SOUND_MIXER_SYNTH) +#define SOUND_MIXER_READ_PCM MIXER_READ(SOUND_MIXER_PCM) +#define SOUND_MIXER_READ_SPEAKER MIXER_READ(SOUND_MIXER_SPEAKER) +#define SOUND_MIXER_READ_LINE MIXER_READ(SOUND_MIXER_LINE) +#define SOUND_MIXER_READ_MIC MIXER_READ(SOUND_MIXER_MIC) +#define SOUND_MIXER_READ_CD MIXER_READ(SOUND_MIXER_CD) +#define SOUND_MIXER_READ_IMIX MIXER_READ(SOUND_MIXER_IMIX) +#define SOUND_MIXER_READ_ALTPCM MIXER_READ(SOUND_MIXER_ALTPCM) +#define SOUND_MIXER_READ_RECLEV MIXER_READ(SOUND_MIXER_RECLEV) +#define SOUND_MIXER_READ_IGAIN MIXER_READ(SOUND_MIXER_IGAIN) +#define SOUND_MIXER_READ_OGAIN MIXER_READ(SOUND_MIXER_OGAIN) +#define SOUND_MIXER_READ_LINE1 MIXER_READ(SOUND_MIXER_LINE1) +#define SOUND_MIXER_READ_LINE2 MIXER_READ(SOUND_MIXER_LINE2) +#define SOUND_MIXER_READ_LINE3 MIXER_READ(SOUND_MIXER_LINE3) + +/* Obsolete macros */ +#define SOUND_MIXER_READ_MUTE MIXER_READ(SOUND_MIXER_MUTE) +#define SOUND_MIXER_READ_ENHANCE MIXER_READ(SOUND_MIXER_ENHANCE) +#define SOUND_MIXER_READ_LOUD MIXER_READ(SOUND_MIXER_LOUD) + +#define SOUND_MIXER_READ_RECSRC MIXER_READ(SOUND_MIXER_RECSRC) +#define SOUND_MIXER_READ_DEVMASK MIXER_READ(SOUND_MIXER_DEVMASK) +#define SOUND_MIXER_READ_RECMASK MIXER_READ(SOUND_MIXER_RECMASK) +#define SOUND_MIXER_READ_STEREODEVS MIXER_READ(SOUND_MIXER_STEREODEVS) +#define SOUND_MIXER_READ_CAPS MIXER_READ(SOUND_MIXER_CAPS) + +#define MIXER_WRITE(dev) __SIOWR('M', dev, int) +#define SOUND_MIXER_WRITE_VOLUME MIXER_WRITE(SOUND_MIXER_VOLUME) +#define SOUND_MIXER_WRITE_BASS MIXER_WRITE(SOUND_MIXER_BASS) +#define SOUND_MIXER_WRITE_TREBLE MIXER_WRITE(SOUND_MIXER_TREBLE) +#define SOUND_MIXER_WRITE_SYNTH MIXER_WRITE(SOUND_MIXER_SYNTH) +#define SOUND_MIXER_WRITE_PCM MIXER_WRITE(SOUND_MIXER_PCM) +#define SOUND_MIXER_WRITE_SPEAKER MIXER_WRITE(SOUND_MIXER_SPEAKER) +#define SOUND_MIXER_WRITE_LINE MIXER_WRITE(SOUND_MIXER_LINE) +#define SOUND_MIXER_WRITE_MIC MIXER_WRITE(SOUND_MIXER_MIC) +#define SOUND_MIXER_WRITE_CD MIXER_WRITE(SOUND_MIXER_CD) +#define SOUND_MIXER_WRITE_IMIX MIXER_WRITE(SOUND_MIXER_IMIX) +#define SOUND_MIXER_WRITE_ALTPCM MIXER_WRITE(SOUND_MIXER_ALTPCM) +#define SOUND_MIXER_WRITE_RECLEV MIXER_WRITE(SOUND_MIXER_RECLEV) +#define SOUND_MIXER_WRITE_IGAIN MIXER_WRITE(SOUND_MIXER_IGAIN) +#define SOUND_MIXER_WRITE_OGAIN MIXER_WRITE(SOUND_MIXER_OGAIN) +#define SOUND_MIXER_WRITE_LINE1 MIXER_WRITE(SOUND_MIXER_LINE1) +#define SOUND_MIXER_WRITE_LINE2 MIXER_WRITE(SOUND_MIXER_LINE2) +#define SOUND_MIXER_WRITE_LINE3 MIXER_WRITE(SOUND_MIXER_LINE3) + +/* Obsolete macros */ +#define SOUND_MIXER_WRITE_MUTE MIXER_WRITE(SOUND_MIXER_MUTE) +#define SOUND_MIXER_WRITE_ENHANCE MIXER_WRITE(SOUND_MIXER_ENHANCE) +#define SOUND_MIXER_WRITE_LOUD MIXER_WRITE(SOUND_MIXER_LOUD) + +#define SOUND_MIXER_WRITE_RECSRC MIXER_WRITE(SOUND_MIXER_RECSRC) + +typedef struct mixer_info /* OBSOLETE */ +{ + char id[16]; + char name[32]; + int modify_counter; + int card_number; + int port_number; + char handle[32]; +} mixer_info; + +/* SOUND_MIXER_INFO is obsolete - use SNDCTL_MIXERINFO instead */ +#define SOUND_MIXER_INFO __SIOR ('M', 101, mixer_info) + +/* + * Two ioctls for special souncard function (OSS/Free only) + */ +#define SOUND_MIXER_AGC _SIOWR('M', 103, int) +#define SOUND_MIXER_3DSE _SIOWR('M', 104, int) +/* + * The SOUND_MIXER_PRIVATE# commands can be redefined by low level drivers. + * These features can be used when accessing device specific features. + */ +#define SOUND_MIXER_PRIVATE1 __SIOWR('M', 111, int) +#define SOUND_MIXER_PRIVATE2 __SIOWR('M', 112, int) +#define SOUND_MIXER_PRIVATE3 __SIOWR('M', 113, int) +#define SOUND_MIXER_PRIVATE4 __SIOWR('M', 114, int) +#define SOUND_MIXER_PRIVATE5 __SIOWR('M', 115, int) + +/* The following two controls were never implemented and they should not be used. */ +#define SOUND_MIXER_READ_MAINVOL __SIOR ('M', 116, int) +#define SOUND_MIXER_WRITE_MAINVOL __SIOWR('M', 116, int) + +/* + * SOUND_MIXER_GETLEVELS and SOUND_MIXER_SETLEVELS calls can be used + * for querying current mixer settings from the driver and for loading + * default volume settings _prior_ activating the mixer (loading + * doesn't affect current state of the mixer hardware). These calls + * are for internal use by the driver software only. + */ + +typedef struct mixer_vol_table +{ + int num; /* Index to volume table */ + char name[32]; + int levels[32]; +} mixer_vol_table; + +#define SOUND_MIXER_GETLEVELS __SIOWR('M', 116, mixer_vol_table) +#define SOUND_MIXER_SETLEVELS __SIOWR('M', 117, mixer_vol_table) + +#define OSS_GETVERSION __SIOR ('M', 118, int) + +/* + * Calls to set/get the recording gain for the currently active + * recording source. These calls automatically map to the right control. + * Note that these calls are not supported by all drivers. In this case + * the call will return -1 with errno set to EINVAL + * + * The _MONGAIN work in similar way but set/get the monitoring gain for + * the currently selected recording source. + */ +#define SOUND_MIXER_READ_RECGAIN __SIOR ('M', 119, int) +#define SOUND_MIXER_WRITE_RECGAIN __SIOWR('M', 119, int) +#define SOUND_MIXER_READ_MONGAIN __SIOR ('M', 120, int) +#define SOUND_MIXER_WRITE_MONGAIN __SIOWR('M', 120, int) + +/* The following call is for driver development time purposes. It's not + * present in any released drivers. + */ +typedef unsigned char oss_reserved_t[512]; +#define SOUND_MIXER_RESERVED __SIOWR('M', 121, oss_reserved_t) +#endif /* ifndef NO_LEGACY_MIXER */ + +/* + ************************************************************************* + * The "new" mixer API of OSS 4.0 and later. + * + * This improved mixer API makes it possible to access every possible feature + * of every possible device. However you should read the mixer programming + * section of the OSS API Developer's Manual. There is no chance that you + * could use this interface correctly just by examining this header. + */ + +typedef struct oss_sysinfo +{ + char product[32]; /* For example OSS/Free, OSS/Linux or OSS/Solaris */ + char version[32]; /* For example 4.0a */ + int versionnum; /* See OSS_GETVERSION */ + char options[128]; /* Reserved */ + + int numaudios; /* # of audio/dsp devices */ + int openedaudio[8]; /* Bit mask telling which audio devices are busy */ + + int numsynths; /* # of availavle synth devices */ + int nummidis; /* # of available MIDI ports */ + int numtimers; /* # of available timer devices */ + int nummixers; /* # of mixer devices */ + + int openedmidi[8]; /* Bit mask telling which midi devices are busy */ + int numcards; /* Number of sound cards in the system */ + int numaudioengines; /* Number of audio engines in the system */ + int filler[240]; /* For future expansion (set to -1) */ +} oss_sysinfo; + +typedef struct oss_mixext +{ + int dev; /* Mixer device number */ + int ctrl; /* Controller number */ + int type; /* Entry type */ +# define MIXT_DEVROOT 0 /* Device root entry */ +# define MIXT_GROUP 1 /* Controller group */ +# define MIXT_ONOFF 2 /* OFF (0) or ON (1) */ +# define MIXT_ENUM 3 /* Enumerated (0 to maxvalue) */ +# define MIXT_MONOSLIDER 4 /* Mono slider (0 to 255) */ +# define MIXT_STEREOSLIDER 5 /* Stereo slider (dual 0 to 255) */ +# define MIXT_MESSAGE 6 /* (Readable) textual message */ +# define MIXT_MONOVU 7 /* VU meter value (mono) */ +# define MIXT_STEREOVU 8 /* VU meter value (stereo) */ +# define MIXT_MONOPEAK 9 /* VU meter peak value (mono) */ +# define MIXT_STEREOPEAK 10 /* VU meter peak value (stereo) */ +# define MIXT_RADIOGROUP 11 /* Radio button group */ +# define MIXT_MARKER 12 /* Separator between normal and extension entries */ +# define MIXT_VALUE 13 /* Decimal value entry */ +# define MIXT_HEXVALUE 14 /* Hexadecimal value entry */ +# define MIXT_MONODB 15 /* OBSOLETE */ +# define MIXT_STEREODB 16 /* OBSOLETE */ +# define MIXT_SLIDER 17 /* Slider (mono) with full (31 bit) postitive integer range */ +# define MIXT_3D 18 + +/* + * Sliders with range expanded to 15 bits per channel (0-32767) + */ +# define MIXT_MONOSLIDER16 19 +# define MIXT_STEREOSLIDER16 20 + + /* Possible value range (minvalue to maxvalue) */ + /* Note that maxvalue may also be smaller than minvalue */ + int maxvalue; + int minvalue; + + int flags; +# define MIXF_READABLE 0x00000001 /* Has readable value */ +# define MIXF_WRITEABLE 0x00000002 /* Has writeable value */ +# define MIXF_POLL 0x00000004 /* May change itself */ +# define MIXF_HZ 0x00000008 /* Herz scale */ +# define MIXF_STRING 0x00000010 /* Use dynamic extensions for value */ +# define MIXF_DYNAMIC 0x00000010 /* Supports dynamic extensions */ +# define MIXF_OKFAIL 0x00000020 /* Interpret value as 1=OK, 0=FAIL */ +# define MIXF_FLAT 0x00000040 /* Flat vertical space requirements */ +# define MIXF_LEGACY 0x00000080 /* Legacy mixer control group */ +# define MIXF_CENTIBEL 0x00000100 /* Centibel (0.1 dB) step size */ +# define MIXF_DECIBEL 0x00000200 /* Step size of 1 dB */ +# define MIXF_MAINVOL 0x00000400 /* Main volume control */ +# define MIXF_PCMVOL 0x00000800 /* PCM output volume control */ +# define MIXF_RECVOL 0x00001000 /* PCM recording volume control */ + char id[16]; /* Mnemonic ID (mainly for internal use) */ + int parent; /* Entry# of parent (group) node (-1 if root) */ + + int dummy; /* Internal use */ + + int timestamp; + + char data[64]; /* Misc data (entry type dependent) */ + unsigned char enum_present[32]; /* Mask of allowed enum values */ + int control_no; /* SOUND_MIXER_VOLUME..SOUND_MIXER_MIDI */ + /* (-1 means not indicated) */ + +/* + * The desc field is reserved for internal purposes of OSS. It should not be + * used by applications. + */ + unsigned int desc; +#define MIXEXT_SCOPE_MASK 0x0000003f +#define MIXEXT_SCOPE_OTHER 0x00000000 +#define MIXEXT_SCOPE_INPUT 0x00000001 +#define MIXEXT_SCOPE_OUTPUT 0x00000002 +#define MIXEXT_SCOPE_MONITOR 0x00000003 +#define MIXEXT_SCOPE_RECSWITCH 0x00000004 + + char extname[32]; + int update_counter; + int filler[7]; +} oss_mixext; + +typedef struct oss_mixext_root +{ + char id[16]; + char name[48]; +} oss_mixext_root; + +typedef struct oss_mixer_value +{ + int dev; + int ctrl; + int value; + int flags; /* Reserved for future use. Initialize to 0 */ + int timestamp; /* Must be set to oss_mixext.timestamp */ + int filler[8]; /* Reserved for future use. Initialize to 0 */ +} oss_mixer_value; + +#define OSS_ENUM_MAXVALUE 255 +typedef struct oss_mixer_enuminfo +{ + int dev; + int ctrl; + int nvalues; + int version; /* Read the manual */ + short strindex[OSS_ENUM_MAXVALUE]; + char strings[3000]; +} oss_mixer_enuminfo; + +#define OPEN_READ PCM_ENABLE_INPUT +#define OPEN_WRITE PCM_ENABLE_OUTPUT +#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) + +typedef struct oss_audioinfo +{ + int dev; /* Audio device number */ + char name[64]; + int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */ + int pid; + int caps; /* PCM_CAP_INPUT, PCM_CAP_OUTPUT */ + int iformats, oformats; + int magic; /* Reserved for internal use */ + char cmd[64]; /* Command using the device (if known) */ + int card_number; + int port_number; + int mixer_dev; + int legacy_device; /* Obsolete field. Replaced by devnode */ + int enabled; /* 1=enabled, 0=device not ready at this moment */ + int flags; /* For internal use only - no practical meaning */ + int min_rate, max_rate; /* Sample rate limits */ + int min_channels, max_channels; /* Number of channels supported */ + int binding; /* DSP_BIND_FRONT, etc. 0 means undefined */ + int rate_source; + char handle[32]; +#define OSS_MAX_SAMPLE_RATES 20 /* Cannot be changed */ + unsigned int nrates, rates[OSS_MAX_SAMPLE_RATES]; /* Please read the manual before using these */ + oss_longname_t song_name; /* Song name (if given) */ + oss_label_t label; /* Device label (if given) */ + int latency; /* In usecs, -1=unknown */ + oss_devnode_t devnode; /* Device special file name (absolute path) */ + int next_play_engine; /* Read the documentation for more info */ + int next_rec_engine; /* Read the documentation for more info */ + int filler[184]; +} oss_audioinfo; + +typedef struct oss_mixerinfo +{ + int dev; + char id[16]; + char name[32]; + int modify_counter; + int card_number; + int port_number; + char handle[32]; + int magic; /* Reserved */ + int enabled; /* Reserved */ + int caps; +#define MIXER_CAP_VIRTUAL 0x00000001 +#define MIXER_CAP_LAYOUT_B 0x00000002 /* For internal use only */ +#define MIXER_CAP_NARROW 0x00000004 /* Conserve horiz space */ + int flags; /* Reserved */ + int nrext; + /* + * The priority field can be used to select the default (motherboard) + * mixer device. The mixer with the highest priority is the + * most preferred one. -2 or less means that this device cannot be used + * as the default mixer. + */ + int priority; + oss_devnode_t devnode; /* Device special file name (absolute path) */ + int legacy_device; + int filler[245]; /* Reserved */ +} oss_mixerinfo; + +typedef struct oss_midi_info +{ + int dev; /* Midi device number */ + char name[64]; + int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */ + int pid; + char cmd[64]; /* Command using the device (if known) */ + int caps; +#define MIDI_CAP_MPU401 0x00000001 /**** OBSOLETE ****/ +#define MIDI_CAP_INPUT 0x00000002 +#define MIDI_CAP_OUTPUT 0x00000004 +#define MIDI_CAP_INOUT (MIDI_CAP_INPUT|MIDI_CAP_OUTPUT) +#define MIDI_CAP_VIRTUAL 0x00000008 /* Pseudo device */ +#define MIDI_CAP_MTCINPUT 0x00000010 /* Supports SNDCTL_MIDI_MTCINPUT */ +#define MIDI_CAP_CLIENT 0x00000020 /* Virtual client side device */ +#define MIDI_CAP_SERVER 0x00000040 /* Virtual server side device */ +#define MIDI_CAP_INTERNAL 0x00000080 /* Internal (synth) device */ +#define MIDI_CAP_EXTERNAL 0x00000100 /* external (MIDI port) device */ +#define MIDI_CAP_PTOP 0x00000200 /* Point to point link to one device */ +#define MIDI_CAP_MTC 0x00000400 /* MTC/SMPTE (control) device */ + int magic; /* Reserved for internal use */ + int card_number; + int port_number; + int enabled; /* 1=enabled, 0=device not ready at this moment */ + int flags; /* For internal use only - no practical meaning */ + char handle[32]; + oss_longname_t song_name; /* Song name (if known) */ + oss_label_t label; /* Device label (if given) */ + int latency; /* In usecs, -1=unknown */ + oss_devnode_t devnode; /* Device special file name (absolute path) */ + int legacy_device; /* Legacy device mapping */ + int filler[235]; +} oss_midi_info; + +typedef struct oss_card_info +{ + int card; + char shortname[16]; + char longname[128]; + int flags; + int filler[256]; +} oss_card_info; + +#define SNDCTL_SYSINFO __SIOR ('X', 1, oss_sysinfo) +#define OSS_SYSINFO SNDCTL_SYSINFO /* Old name */ + +#define SNDCTL_MIX_NRMIX __SIOR ('X', 2, int) +#define SNDCTL_MIX_NREXT __SIOWR('X', 3, int) +#define SNDCTL_MIX_EXTINFO __SIOWR('X', 4, oss_mixext) +#define SNDCTL_MIX_READ __SIOWR('X', 5, oss_mixer_value) +#define SNDCTL_MIX_WRITE __SIOWR('X', 6, oss_mixer_value) + +#define SNDCTL_AUDIOINFO __SIOWR('X', 7, oss_audioinfo) +#define SNDCTL_MIX_ENUMINFO __SIOWR('X', 8, oss_mixer_enuminfo) +#define SNDCTL_MIDIINFO __SIOWR('X', 9, oss_midi_info) +#define SNDCTL_MIXERINFO __SIOWR('X',10, oss_mixerinfo) +#define SNDCTL_CARDINFO __SIOWR('X',11, oss_card_info) +#define SNDCTL_ENGINEINFO __SIOWR('X',12, oss_audioinfo) +#define SNDCTL_AUDIOINFO_EX __SIOWR('X',13, oss_audioinfo) + +/* ioctl codes 'X', 200-255 are reserved for internal use */ + +/* + * Few more "globally" available ioctl calls. + */ +#define SNDCTL_SETSONG __SIOW ('Y', 2, oss_longname_t) +#define SNDCTL_GETSONG __SIOR ('Y', 2, oss_longname_t) +#define SNDCTL_SETNAME __SIOW ('Y', 3, oss_longname_t) +#define SNDCTL_SETLABEL __SIOW ('Y', 4, oss_label_t) +#define SNDCTL_GETLABEL __SIOR ('Y', 4, oss_label_t) +/* + * The "new" mixer API definitions end here. + *************************************** + */ + +/* + ********************************************************* + * Few routines that are included in -lOSSlib + * + * At this moment this interface is not used. OSSlib contains just + * stubs that call the related system calls directly. + */ +#ifdef OSSLIB +extern int osslib_open (const char *path, int flags, int dummy); +extern void osslib_close (int fd); +extern int osslib_write (int fd, const void *buf, int count); +extern int osslib_read (int fd, void *buf, int count); +extern int osslib_ioctl (int fd, unsigned int request, void *arg); +#else +# define osslib_open open +# define osslib_close close +# define osslib_write write +# define osslib_read read +# define osslib_ioctl ioctl +#endif + +#if 1 +#define SNDCTL_DSP_NONBLOCK __SIO ('P',14) /* Obsolete. Not supported any more */ +#endif + +#if 1 +/* + * Some obsolete macros that are not part of Open Sound System API. + */ +#define SOUND_PCM_READ_RATE SOUND_PCM_READ_RATE_is_obsolete +#define SOUND_PCM_READ_BITS SOUND_PCM_READ_BITS_is_obsolete +#define SOUND_PCM_READ_CHANNELS SOUND_PCM_READ_CHANNELS_is_obsolete +#define SOUND_PCM_WRITE_RATE SOUND_PCM_WRITE_RATE_is_obsolet_use_SNDCTL_DSP_SPEED_instead +#define SOUND_PCM_WRITE_CHANNELS SOUND_PCM_WRITE_CHANNELS_is_obsolete_use_SNDCTL_DSP_CHANNELS_instead +#define SOUND_PCM_WRITE_BITS SOUND_PCM_WRITE_BITS_is_obsolete_use_SNDCTL_DSP_SETFMT_instead +#define SOUND_PCM_POST SOUND_PCM_POST_is_obsolete_use_SNDCTL_DSP_POST_instead +#define SOUND_PCM_RESET SOUND_PCM_RESET_is_obsolete_use_SNDCTL_DSP_HALT_instead +#define SOUND_PCM_SYNC SOUND_PCM_SYNC_is_obsolete_use_SNDCTL_DSP_SYNC_instead +#define SOUND_PCM_SUBDIVIDE SOUND_PCM_SUBDIVIDE_is_obsolete_use_SNDCTL_DSP_SUBDIVIDE_instead +#define SOUND_PCM_SETFRAGMENT SOUND_PCM_SETFRAGMENT_is_obsolete_use_SNDCTL_DSP_SETFRAGMENT_instead +#define SOUND_PCM_GETFMTS SOUND_PCM_GETFMTS_is_obsolete_use_SNDCTL_DSP_GETFMTS_instead +#define SOUND_PCM_SETFMT SOUND_PCM_SETFMT_is_obsolete_use_SNDCTL_DSP_SETFMT_instead +#define SOUND_PCM_GETOSPACE SOUND_PCM_GETOSPACE_is_obsolete_use_SNDCTL_DSP_GETOSPACE_instead +#define SOUND_PCM_GETISPACE SOUND_PCM_GETISPACE_is_obsolete_use_SNDCTL_DSP_GETISPACE_instead +#define SOUND_PCM_NONBLOCK SOUND_PCM_NONBLOCK_is_obsolete_use_SNDCTL_DSP_NONBLOCK_instead +#define SOUND_PCM_GETCAPS SOUND_PCM_GETCAPS_is_obsolete_use_SNDCTL_DSP_GETCAPS_instead +#define SOUND_PCM_GETTRIGGER SOUND_PCM_GETTRIGGER_is_obsolete_use_SNDCTL_DSP_GETTRIGGER_instead +#define SOUND_PCM_SETTRIGGER SOUND_PCM_SETTRIGGER_is_obsolete_use_SNDCTL_DSP_SETTRIGGER_instead +#define SOUND_PCM_SETSYNCRO SOUND_PCM_SETSYNCRO_is_obsolete_use_SNDCTL_DSP_SETSYNCRO_instead +#define SOUND_PCM_GETIPTR SOUND_PCM_GETIPTR_is_obsolete_use_SNDCTL_DSP_GETIPTR_instead +#define SOUND_PCM_GETOPTR SOUND_PCM_GETOPTR_is_obsolete_use_SNDCTL_DSP_GETOPTR_instead +#define SOUND_PCM_MAPINBUF SOUND_PCM_MAPINBUF_is_obsolete_use_SNDCTL_DSP_MAPINBUF_instead +#define SOUND_PCM_MAPOUTBUF SOUND_PCM_MAPOUTBUF_is_obsolete_use_SNDCTL_DSP_MAPOUTBUF_instead +#endif + +#endif diff --git a/rtaudio/readme b/rtaudio/readme new file mode 100644 index 0000000..8228cd3 --- /dev/null +++ b/rtaudio/readme @@ -0,0 +1,61 @@ +RtAudio - a set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. + +By Gary P. Scavone, 2001-2008. + +This distribution of RtAudio contains the following: + +doc: RtAudio documentation (see doc/html/index.html) +tests: example RtAudio programs +asio: header and source files necessary for ASIO compilation +tests/Windows: Visual C++ .net test program workspace and projects + +OVERVIEW: + +RtAudio is a set of C++ classes that provide a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X, SGI, and Windows (DirectSound and ASIO) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives: + + - object-oriented C++ design + - simple, common API across all supported platforms + - only one source and two header files for easy inclusion in programming projects + - allow simultaneous multi-api support + - support dynamic connection of devices + - provide extensive audio device parameter control + - allow audio device capability probing + - automatic internal conversion for data format, channel number compensation, (de)interleaving, and byte-swapping + +RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Where applicable, multiple API support can be compiled and a particular API specified when creating an RtAudio instance. See the \ref apinotes section for information specific to each of the supported audio APIs. + +FURTHER READING: + +For complete documentation on RtAudio, see the doc directory of the distribution or surf to http://www.music.mcgill.ca/~gary/rtaudio/. + + +LEGAL AND ETHICAL: + +The RtAudio license is similar to the MIT License. + + RtAudio: a set of realtime audio i/o C++ classes + Copyright (c) 2001-2008 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. diff --git a/sample.ogg b/sample.ogg new file mode 100644 index 0000000..33c18ea Binary files /dev/null and b/sample.ogg differ diff --git a/scale.lua b/scale.lua new file mode 100644 index 0000000..969dc1a --- /dev/null +++ b/scale.lua @@ -0,0 +1,33 @@ +-- function creating a sine wave sample: +function sampleSine(freq, duration, sampleRate) + local data = { } + for i = 1,duration*sampleRate do + data[i] = math.sin( (i*freq/sampleRate)*math.pi*2) + end + return proAudio.sampleFromMemory(data, sampleRate) +end + +-- plays a sample shifted by a number of halftones for a definable period of time +function playNote(sample, pitch, duration, volumeL, volumeR, disparity) + local scale = 2^(pitch/12) + local sound = proAudio.soundLoop(sample, volumeL, volumeR, disparity, scale) + proAudio.sleep(duration) + proAudio.soundStop(sound) +end + + +-- create an audio device using default parameters and exit in case of errors +require("proAudioRt") +if not proAudio.create() then os.exit(1) end + +-- generate a sample: +local sample = sampleSine(440, 0.5, 88200) + +-- play scale (a major): +local duration = 0.5 +for i,note in ipairs({ 0, 2, 4, 5, 7, 9, 11, 12 }) do + playNote(sample, note, duration) +end + +-- cleanup +proAudio.destroy() diff --git a/stb_vorbis.c b/stb_vorbis.c new file mode 100644 index 0000000..f5b35c3 --- /dev/null +++ b/stb_vorbis.c @@ -0,0 +1,5349 @@ +// Ogg Vorbis I audio decoder -- version 0.99996 +// +// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools. +// +// Placed in the public domain April 2007 by the author: no copyright is +// claimed, and you may use it for any purpose you like. +// +// No warranty for any purpose is expressed or implied by the author (nor +// by RAD Game Tools). Report bugs and send enhancements to the author. +// +// Get the latest version and other information at: +// http://nothings.org/stb_vorbis/ + + +// Todo: +// +// - seeking (note you can seek yourself using the pushdata API) +// +// Limitations: +// +// - floor 0 not supported (used in old ogg vorbis files) +// - lossless sample-truncation at beginning ignored +// - cannot concatenate multiple vorbis streams +// - sample positions are 32-bit, limiting seekable 192Khz +// files to around 6 hours (Ogg supports 64-bit) +// +// All of these limitations may be removed in future versions. + + +////////////////////////////////////////////////////////////////////////////// +// +// HEADER BEGINS HERE +// + +#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H +#define STB_VORBIS_INCLUDE_STB_VORBIS_H + +#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) +#define STB_VORBIS_NO_STDIO 1 +#endif + +#ifndef STB_VORBIS_NO_STDIO +#include +#endif + +#ifdef __cplusplus +extern "C" { +#endif + +/////////// THREAD SAFETY + +// Individual stb_vorbis* handles are not thread-safe; you cannot decode from +// them from multiple threads at the same time. However, you can have multiple +// stb_vorbis* handles and decode from them independently in multiple thrads. + + +/////////// MEMORY ALLOCATION + +// normally stb_vorbis uses malloc() to allocate memory at startup, +// and alloca() to allocate temporary memory during a frame on the +// stack. (Memory consumption will depend on the amount of setup +// data in the file and how you set the compile flags for speed +// vs. size. In my test files the maximal-size usage is ~150KB.) +// +// You can modify the wrapper functions in the source (setup_malloc, +// setup_temp_malloc, temp_malloc) to change this behavior, or you +// can use a simpler allocation model: you pass in a buffer from +// which stb_vorbis will allocate _all_ its memory (including the +// temp memory). "open" may fail with a VORBIS_outofmem if you +// do not pass in enough data; there is no way to determine how +// much you do need except to succeed (at which point you can +// query get_info to find the exact amount required. yes I know +// this is lame). +// +// If you pass in a non-NULL buffer of the type below, allocation +// will occur from it as described above. Otherwise just pass NULL +// to use malloc()/alloca() + +typedef struct +{ + char *alloc_buffer; + int alloc_buffer_length_in_bytes; +} stb_vorbis_alloc; + + +/////////// FUNCTIONS USEABLE WITH ALL INPUT MODES + +typedef struct stb_vorbis stb_vorbis; + +typedef struct +{ + unsigned int sample_rate; + int channels; + + unsigned int setup_memory_required; + unsigned int setup_temp_memory_required; + unsigned int temp_memory_required; + + int max_frame_size; +} stb_vorbis_info; + +// get general information about the file +extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f); + +// get the last error detected (clears it, too) +extern int stb_vorbis_get_error(stb_vorbis *f); + +// close an ogg vorbis file and free all memory in use +extern void stb_vorbis_close(stb_vorbis *f); + +// this function returns the offset (in samples) from the beginning of the +// file that will be returned by the next decode, if it is known, or -1 +// otherwise. after a flush_pushdata() call, this may take a while before +// it becomes valid again. +// NOT WORKING YET after a seek with PULLDATA API +extern int stb_vorbis_get_sample_offset(stb_vorbis *f); + +// returns the current seek point within the file, or offset from the beginning +// of the memory buffer. In pushdata mode it returns 0. +extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f); + +/////////// PUSHDATA API + +#ifndef STB_VORBIS_NO_PUSHDATA_API + +// this API allows you to get blocks of data from any source and hand +// them to stb_vorbis. you have to buffer them; stb_vorbis will tell +// you how much it used, and you have to give it the rest next time; +// and stb_vorbis may not have enough data to work with and you will +// need to give it the same data again PLUS more. Note that the Vorbis +// specification does not bound the size of an individual frame. + +extern stb_vorbis *stb_vorbis_open_pushdata( + unsigned char *datablock, int datablock_length_in_bytes, + int *datablock_memory_consumed_in_bytes, + int *error, + stb_vorbis_alloc *alloc_buffer); +// create a vorbis decoder by passing in the initial data block containing +// the ogg&vorbis headers (you don't need to do parse them, just provide +// the first N bytes of the file--you're told if it's not enough, see below) +// on success, returns an stb_vorbis *, does not set error, returns the amount of +// data parsed/consumed on this call in *datablock_memory_consumed_in_bytes; +// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed +// if returns NULL and *error is VORBIS_need_more_data, then the input block was +// incomplete and you need to pass in a larger block from the start of the file + +extern int stb_vorbis_decode_frame_pushdata( + stb_vorbis *f, unsigned char *datablock, int datablock_length_in_bytes, + int *channels, // place to write number of float * buffers + float ***output, // place to write float ** array of float * buffers + int *samples // place to write number of output samples + ); +// decode a frame of audio sample data if possible from the passed-in data block +// +// return value: number of bytes we used from datablock +// possible cases: +// 0 bytes used, 0 samples output (need more data) +// N bytes used, 0 samples output (resynching the stream, keep going) +// N bytes used, M samples output (one frame of data) +// note that after opening a file, you will ALWAYS get one N-bytes,0-sample +// frame, because Vorbis always "discards" the first frame. +// +// Note that on resynch, stb_vorbis will rarely consume all of the buffer, +// instead only datablock_length_in_bytes-3 or less. This is because it wants +// to avoid missing parts of a page header if they cross a datablock boundary, +// without writing state-machiney code to record a partial detection. +// +// The number of channels returned are stored in *channels (which can be +// NULL--it is always the same as the number of channels reported by +// get_info). *output will contain an array of float* buffers, one per +// channel. In other words, (*output)[0][0] contains the first sample from +// the first channel, and (*output)[1][0] contains the first sample from +// the second channel. + +extern void stb_vorbis_flush_pushdata(stb_vorbis *f); +// inform stb_vorbis that your next datablock will not be contiguous with +// previous ones (e.g. you've seeked in the data); future attempts to decode +// frames will cause stb_vorbis to resynchronize (as noted above), and +// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it +// will begin decoding the _next_ frame. +// +// if you want to seek using pushdata, you need to seek in your file, then +// call stb_vorbis_flush_pushdata(), then start calling decoding, then once +// decoding is returning you data, call stb_vorbis_get_sample_offset, and +// if you don't like the result, seek your file again and repeat. +#endif + + +////////// PULLING INPUT API + +#ifndef STB_VORBIS_NO_PULLDATA_API +// This API assumes stb_vorbis is allowed to pull data from a source-- +// either a block of memory containing the _entire_ vorbis stream, or a +// FILE * that you or it create, or possibly some other reading mechanism +// if you go modify the source to replace the FILE * case with some kind +// of callback to your code. (But if you don't support seeking, you may +// just want to go ahead and use pushdata.) + +#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION) +extern int stb_vorbis_decode_filename(char *filename, int *channels, int* sample_rate, short **output); +#endif +extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, int* sample_rate, short **output); +// decode an entire file and output the data interleaved into a malloc()ed +// buffer stored in *output. The return value is the number of samples +// decoded, or -1 if the file could not be opened or was not an ogg vorbis file. +// When you're done with it, just free() the pointer returned in *output. + +extern stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len, + int *error, stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from an ogg vorbis stream in memory (note +// this must be the entire stream!). on failure, returns NULL and sets *error + +#ifndef STB_VORBIS_NO_STDIO +extern stb_vorbis * stb_vorbis_open_filename(char *filename, + int *error, stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from a filename via fopen(). on failure, +// returns NULL and sets *error (possibly to VORBIS_file_open_failure). + +extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close, + int *error, stb_vorbis_alloc *alloc_buffer); +// create an ogg vorbis decoder from an open FILE *, looking for a stream at +// the _current_ seek point (ftell). on failure, returns NULL and sets *error. +// note that stb_vorbis must "own" this stream; if you seek it in between +// calls to stb_vorbis, it will become confused. Morever, if you attempt to +// perform stb_vorbis_seek_*() operations on this file, it will assume it +// owns the _entire_ rest of the file after the start point. Use the next +// function, stb_vorbis_open_file_section(), to limit it. + +extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close, + int *error, stb_vorbis_alloc *alloc_buffer, unsigned int len); +// create an ogg vorbis decoder from an open FILE *, looking for a stream at +// the _current_ seek point (ftell); the stream will be of length 'len' bytes. +// on failure, returns NULL and sets *error. note that stb_vorbis must "own" +// this stream; if you seek it in between calls to stb_vorbis, it will become +// confused. +#endif + +extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number); +extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number); +// NOT WORKING YET +// these functions seek in the Vorbis file to (approximately) 'sample_number'. +// after calling seek_frame(), the next call to get_frame_*() will include +// the specified sample. after calling stb_vorbis_seek(), the next call to +// stb_vorbis_get_samples_* will start with the specified sample. If you +// do not need to seek to EXACTLY the target sample when using get_samples_*, +// you can also use seek_frame(). + +extern void stb_vorbis_seek_start(stb_vorbis *f); +// this function is equivalent to stb_vorbis_seek(f,0), but it +// actually works + +extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f); +extern float stb_vorbis_stream_length_in_seconds(stb_vorbis *f); +// these functions return the total length of the vorbis stream + +extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output); +// decode the next frame and return the number of samples. the number of +// channels returned are stored in *channels (which can be NULL--it is always +// the same as the number of channels reported by get_info). *output will +// contain an array of float* buffers, one per channel. These outputs will +// be overwritten on the next call to stb_vorbis_get_frame_*. +// +// You generally should not intermix calls to stb_vorbis_get_frame_*() +// and stb_vorbis_get_samples_*(), since the latter calls the former. + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts); +extern int stb_vorbis_get_frame_short (stb_vorbis *f, int num_c, short **buffer, int num_samples); +#endif +// decode the next frame and return the number of samples per channel. the +// data is coerced to the number of channels you request according to the +// channel coercion rules (see below). You must pass in the size of your +// buffer(s) so that stb_vorbis will not overwrite the end of the buffer. +// The maximum buffer size needed can be gotten from get_info(); however, +// the Vorbis I specification implies an absolute maximum of 4096 samples +// per channel. Note that for interleaved data, you pass in the number of +// shorts (the size of your array), but the return value is the number of +// samples per channel, not the total number of samples. + +// Channel coercion rules: +// Let M be the number of channels requested, and N the number of channels present, +// and Cn be the nth channel; let stereo L be the sum of all L and center channels, +// and stereo R be the sum of all R and center channels (channel assignment from the +// vorbis spec). +// M N output +// 1 k sum(Ck) for all k +// 2 * stereo L, stereo R +// k l k > l, the first l channels, then 0s +// k l k <= l, the first k channels +// Note that this is not _good_ surround etc. mixing at all! It's just so +// you get something useful. + +extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats); +extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples); +// gets num_samples samples, not necessarily on a frame boundary--this requires +// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES. +// Returns the number of samples stored per channel; it may be less than requested +// at the end of the file. If there are no more samples in the file, returns 0. + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts); +extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples); +#endif +// gets num_samples samples, not necessarily on a frame boundary--this requires +// buffering so you have to supply the buffers. Applies the coercion rules above +// to produce 'channels' channels. Returns the number of samples stored per channel; +// it may be less than requested at the end of the file. If there are no more +// samples in the file, returns 0. + +#endif + +//////// ERROR CODES + +enum STBVorbisError +{ + VORBIS__no_error, + + VORBIS_need_more_data=1, // not a real error + + VORBIS_invalid_api_mixing, // can't mix API modes + VORBIS_outofmem, // not enough memory + VORBIS_feature_not_supported, // uses floor 0 + VORBIS_too_many_channels, // STB_VORBIS_MAX_CHANNELS is too small + VORBIS_file_open_failure, // fopen() failed + VORBIS_seek_without_length, // can't seek in unknown-length file + + VORBIS_unexpected_eof=10, // file is truncated? + VORBIS_seek_invalid, // seek past EOF + + // decoding errors (corrupt/invalid stream) -- you probably + // don't care about the exact details of these + + // vorbis errors: + VORBIS_invalid_setup=20, + VORBIS_invalid_stream, + + // ogg errors: + VORBIS_missing_capture_pattern=30, + VORBIS_invalid_stream_structure_version, + VORBIS_continued_packet_flag_invalid, + VORBIS_incorrect_stream_serial_number, + VORBIS_invalid_first_page, + VORBIS_bad_packet_type, + VORBIS_cant_find_last_page, + VORBIS_seek_failed, +}; + + +#ifdef __cplusplus +} +#endif + +#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H +// +// HEADER ENDS HERE +// +////////////////////////////////////////////////////////////////////////////// + +#ifndef STB_VORBIS_HEADER_ONLY + +// global configuration settings (e.g. set these in the project/makefile), +// or just set them in this file at the top (although ideally the first few +// should be visible when the header file is compiled too, although it's not +// crucial) + +// STB_VORBIS_NO_PUSHDATA_API +// does not compile the code for the various stb_vorbis_*_pushdata() +// functions +// #define STB_VORBIS_NO_PUSHDATA_API + +// STB_VORBIS_NO_PULLDATA_API +// does not compile the code for the non-pushdata APIs +// #define STB_VORBIS_NO_PULLDATA_API + +// STB_VORBIS_NO_STDIO +// does not compile the code for the APIs that use FILE *s internally +// or externally (implied by STB_VORBIS_NO_PULLDATA_API) +// #define STB_VORBIS_NO_STDIO + +// STB_VORBIS_NO_INTEGER_CONVERSION +// does not compile the code for converting audio sample data from +// float to integer (implied by STB_VORBIS_NO_PULLDATA_API) +// #define STB_VORBIS_NO_INTEGER_CONVERSION + +// STB_VORBIS_NO_FAST_SCALED_FLOAT +// does not use a fast float-to-int trick to accelerate float-to-int on +// most platforms which requires endianness be defined correctly. +//#define STB_VORBIS_NO_FAST_SCALED_FLOAT + + +// STB_VORBIS_MAX_CHANNELS [number] +// globally define this to the maximum number of channels you need. +// The spec does not put a restriction on channels except that +// the count is stored in a byte, so 255 is the hard limit. +// Reducing this saves about 16 bytes per value, so using 16 saves +// (255-16)*16 or around 4KB. Plus anything other memory usage +// I forgot to account for. Can probably go as low as 8 (7.1 audio), +// 6 (5.1 audio), or 2 (stereo only). +#ifndef STB_VORBIS_MAX_CHANNELS +#define STB_VORBIS_MAX_CHANNELS 16 // enough for anyone? +#endif + +// STB_VORBIS_PUSHDATA_CRC_COUNT [number] +// after a flush_pushdata(), stb_vorbis begins scanning for the +// next valid page, without backtracking. when it finds something +// that looks like a page, it streams through it and verifies its +// CRC32. Should that validation fail, it keeps scanning. But it's +// possible that _while_ streaming through to check the CRC32 of +// one candidate page, it sees another candidate page. This #define +// determines how many "overlapping" candidate pages it can search +// at once. Note that "real" pages are typically ~4KB to ~8KB, whereas +// garbage pages could be as big as 64KB, but probably average ~16KB. +// So don't hose ourselves by scanning an apparent 64KB page and +// missing a ton of real ones in the interim; so minimum of 2 +#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT +#define STB_VORBIS_PUSHDATA_CRC_COUNT 4 +#endif + +// STB_VORBIS_FAST_HUFFMAN_LENGTH [number] +// sets the log size of the huffman-acceleration table. Maximum +// supported value is 24. with larger numbers, more decodings are O(1), +// but the table size is larger so worse cache missing, so you'll have +// to probe (and try multiple ogg vorbis files) to find the sweet spot. +#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH +#define STB_VORBIS_FAST_HUFFMAN_LENGTH 10 +#endif + +// STB_VORBIS_FAST_BINARY_LENGTH [number] +// sets the log size of the binary-search acceleration table. this +// is used in similar fashion to the fast-huffman size to set initial +// parameters for the binary search + +// STB_VORBIS_FAST_HUFFMAN_INT +// The fast huffman tables are much more efficient if they can be +// stored as 16-bit results instead of 32-bit results. This restricts +// the codebooks to having only 65535 possible outcomes, though. +// (At least, accelerated by the huffman table.) +#ifndef STB_VORBIS_FAST_HUFFMAN_INT +#define STB_VORBIS_FAST_HUFFMAN_SHORT +#endif + +// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH +// If the 'fast huffman' search doesn't succeed, then stb_vorbis falls +// back on binary searching for the correct one. This requires storing +// extra tables with the huffman codes in sorted order. Defining this +// symbol trades off space for speed by forcing a linear search in the +// non-fast case, except for "sparse" codebooks. +// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH + +// STB_VORBIS_DIVIDES_IN_RESIDUE +// stb_vorbis precomputes the result of the scalar residue decoding +// that would otherwise require a divide per chunk. you can trade off +// space for time by defining this symbol. +// #define STB_VORBIS_DIVIDES_IN_RESIDUE + +// STB_VORBIS_DIVIDES_IN_CODEBOOK +// vorbis VQ codebooks can be encoded two ways: with every case explicitly +// stored, or with all elements being chosen from a small range of values, +// and all values possible in all elements. By default, stb_vorbis expands +// this latter kind out to look like the former kind for ease of decoding, +// because otherwise an integer divide-per-vector-element is required to +// unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can +// trade off storage for speed. +//#define STB_VORBIS_DIVIDES_IN_CODEBOOK + +// STB_VORBIS_CODEBOOK_SHORTS +// The vorbis file format encodes VQ codebook floats as ax+b where a and +// b are floating point per-codebook constants, and x is a 16-bit int. +// Normally, stb_vorbis decodes them to floats rather than leaving them +// as 16-bit ints and computing ax+b while decoding. This is a speed/space +// tradeoff; you can save space by defining this flag. +#ifndef STB_VORBIS_CODEBOOK_SHORTS +#define STB_VORBIS_CODEBOOK_FLOATS +#endif + +// STB_VORBIS_DIVIDE_TABLE +// this replaces small integer divides in the floor decode loop with +// table lookups. made less than 1% difference, so disabled by default. + +// STB_VORBIS_NO_INLINE_DECODE +// disables the inlining of the scalar codebook fast-huffman decode. +// might save a little codespace; useful for debugging +// #define STB_VORBIS_NO_INLINE_DECODE + +// STB_VORBIS_NO_DEFER_FLOOR +// Normally we only decode the floor without synthesizing the actual +// full curve. We can instead synthesize the curve immediately. This +// requires more memory and is very likely slower, so I don't think +// you'd ever want to do it except for debugging. +// #define STB_VORBIS_NO_DEFER_FLOOR + + + + +////////////////////////////////////////////////////////////////////////////// + +#ifdef STB_VORBIS_NO_PULLDATA_API + #define STB_VORBIS_NO_INTEGER_CONVERSION + #define STB_VORBIS_NO_STDIO +#endif + +#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) + #define STB_VORBIS_NO_STDIO 1 +#endif + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT + + // only need endianness for fast-float-to-int, which we don't + // use for pushdata + + #ifndef STB_VORBIS_BIG_ENDIAN + #define STB_VORBIS_ENDIAN 0 + #else + #define STB_VORBIS_ENDIAN 1 + #endif + +#endif +#endif + + +#ifndef STB_VORBIS_NO_STDIO +#include +#endif + +#ifndef STB_VORBIS_NO_CRT +#include +#include +#include +#include +#if !(defined(__APPLE__) || defined(MACOSX) || defined(macintosh) || defined(Macintosh)) +#include +#endif +#else +#define NULL 0 +#endif + +#ifndef _MSC_VER + #if __GNUC__ + #define __forceinline inline + #else + #define __forceinline + #endif +#endif + +#if STB_VORBIS_MAX_CHANNELS > 256 +#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range" +#endif + +#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24 +#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range" +#endif + + +#define MAX_BLOCKSIZE_LOG 13 // from specification +#define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG) + + +typedef unsigned char uint8; +typedef signed char int8; +typedef unsigned short uint16; +typedef signed short int16; +typedef unsigned int uint32; +typedef signed int int32; + +#ifndef TRUE +#define TRUE 1 +#define FALSE 0 +#endif + +#ifdef STB_VORBIS_CODEBOOK_FLOATS +typedef float codetype; +#else +typedef uint16 codetype; +#endif + +// @NOTE +// +// Some arrays below are tagged "//varies", which means it's actually +// a variable-sized piece of data, but rather than malloc I assume it's +// small enough it's better to just allocate it all together with the +// main thing +// +// Most of the variables are specified with the smallest size I could pack +// them into. It might give better performance to make them all full-sized +// integers. It should be safe to freely rearrange the structures or change +// the sizes larger--nothing relies on silently truncating etc., nor the +// order of variables. + +#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) +#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) + +typedef struct +{ + int dimensions, entries; + uint8 *codeword_lengths; + float minimum_value; + float delta_value; + uint8 value_bits; + uint8 lookup_type; + uint8 sequence_p; + uint8 sparse; + uint32 lookup_values; + codetype *multiplicands; + uint32 *codewords; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #else + int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; + #endif + uint32 *sorted_codewords; + int *sorted_values; + int sorted_entries; +} Codebook; + +typedef struct +{ + uint8 order; + uint16 rate; + uint16 bark_map_size; + uint8 amplitude_bits; + uint8 amplitude_offset; + uint8 number_of_books; + uint8 book_list[16]; // varies +} Floor0; + +typedef struct +{ + uint8 partitions; + uint8 partition_class_list[32]; // varies + uint8 class_dimensions[16]; // varies + uint8 class_subclasses[16]; // varies + uint8 class_masterbooks[16]; // varies + int16 subclass_books[16][8]; // varies + uint16 Xlist[31*8+2]; // varies + uint8 sorted_order[31*8+2]; + uint8 neighbors[31*8+2][2]; + uint8 floor1_multiplier; + uint8 rangebits; + int values; +} Floor1; + +typedef union +{ + Floor0 floor0; + Floor1 floor1; +} Floor; + +typedef struct +{ + uint32 begin, end; + uint32 part_size; + uint8 classifications; + uint8 classbook; + uint8 **classdata; + int16 (*residue_books)[8]; +} Residue; + +typedef struct +{ + uint8 magnitude; + uint8 angle; + uint8 mux; +} MappingChannel; + +typedef struct +{ + uint16 coupling_steps; + MappingChannel *chan; + uint8 submaps; + uint8 submap_floor[15]; // varies + uint8 submap_residue[15]; // varies +} Mapping; + +typedef struct +{ + uint8 blockflag; + uint8 mapping; + uint16 windowtype; + uint16 transformtype; +} Mode; + +typedef struct +{ + uint32 goal_crc; // expected crc if match + int bytes_left; // bytes left in packet + uint32 crc_so_far; // running crc + int bytes_done; // bytes processed in _current_ chunk + uint32 sample_loc; // granule pos encoded in page +} CRCscan; + +typedef struct +{ + uint32 page_start, page_end; + uint32 after_previous_page_start; + uint32 first_decoded_sample; + uint32 last_decoded_sample; +} ProbedPage; + +struct stb_vorbis +{ + // user-accessible info + unsigned int sample_rate; + int channels; + + unsigned int setup_memory_required; + unsigned int temp_memory_required; + unsigned int setup_temp_memory_required; + + // input config +#ifndef STB_VORBIS_NO_STDIO + FILE *f; + uint32 f_start; + int close_on_free; +#endif + + uint8 *stream; + uint8 *stream_start; + uint8 *stream_end; + + uint32 stream_len; + + uint8 push_mode; + + uint32 first_audio_page_offset; + + ProbedPage p_first, p_last; + + // memory management + stb_vorbis_alloc alloc; + int setup_offset; + int temp_offset; + + // run-time results + int eof; + enum STBVorbisError error; + + // user-useful data + + // header info + int blocksize[2]; + int blocksize_0, blocksize_1; + int codebook_count; + Codebook *codebooks; + int floor_count; + uint16 floor_types[64]; // varies + Floor *floor_config; + int residue_count; + uint16 residue_types[64]; // varies + Residue *residue_config; + int mapping_count; + Mapping *mapping; + int mode_count; + Mode mode_config[64]; // varies + + uint32 total_samples; + + // decode buffer + float *channel_buffers[STB_VORBIS_MAX_CHANNELS]; + float *outputs [STB_VORBIS_MAX_CHANNELS]; + + float *previous_window[STB_VORBIS_MAX_CHANNELS]; + int previous_length; + + #ifndef STB_VORBIS_NO_DEFER_FLOOR + int16 *finalY[STB_VORBIS_MAX_CHANNELS]; + #else + float *floor_buffers[STB_VORBIS_MAX_CHANNELS]; + #endif + + uint32 current_loc; // sample location of next frame to decode + int current_loc_valid; + + // per-blocksize precomputed data + + // twiddle factors + float *A[2],*B[2],*C[2]; + float *window[2]; + uint16 *bit_reverse[2]; + + // current page/packet/segment streaming info + uint32 serial; // stream serial number for verification + int last_page; + int segment_count; + uint8 segments[255]; + uint8 page_flag; + uint8 bytes_in_seg; + uint8 first_decode; + int next_seg; + int last_seg; // flag that we're on the last segment + int last_seg_which; // what was the segment number of the last seg? + uint32 acc; + int valid_bits; + int packet_bytes; + int end_seg_with_known_loc; + uint32 known_loc_for_packet; + int discard_samples_deferred; + uint32 samples_output; + + // push mode scanning + int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching +#ifndef STB_VORBIS_NO_PUSHDATA_API + CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT]; +#endif + + // sample-access + int channel_buffer_start; + int channel_buffer_end; +}; + +#if defined(STB_VORBIS_NO_PUSHDATA_API) + #define IS_PUSH_MODE(f) FALSE +#elif defined(STB_VORBIS_NO_PULLDATA_API) + #define IS_PUSH_MODE(f) TRUE +#else + #define IS_PUSH_MODE(f) ((f)->push_mode) +#endif + +typedef struct stb_vorbis vorb; + +static int error(vorb *f, enum STBVorbisError e) +{ + f->error = e; + if (!f->eof && e != VORBIS_need_more_data) { + f->error=e; // breakpoint for debugging + } + return 0; +} + + +// these functions are used for allocating temporary memory +// while decoding. if you can afford the stack space, use +// alloca(); otherwise, provide a temp buffer and it will +// allocate out of those. + +#define array_size_required(count,size) (count*(sizeof(void *)+(size))) + +#define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size)) +#ifdef dealloca +#define temp_free(f,p) (f->alloc.alloc_buffer ? 0 : dealloca(size)) +#else +#define temp_free(f,p) 0 +#endif +#define temp_alloc_save(f) ((f)->temp_offset) +#define temp_alloc_restore(f,p) ((f)->temp_offset = (p)) + +#define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size) + +// given a sufficiently large block of memory, make an array of pointers to subblocks of it +static void *make_block_array(void *mem, int count, int size) +{ + int i; + void ** p = (void **) mem; + char *q = (char *) (p + count); + for (i=0; i < count; ++i) { + p[i] = q; + q += size; + } + return p; +} + +static void *setup_malloc(vorb *f, int sz) +{ + sz = (sz+3) & ~3; + f->setup_memory_required += sz; + if (f->alloc.alloc_buffer) { + void *p = (char *) f->alloc.alloc_buffer + f->setup_offset; + if (f->setup_offset + sz > f->temp_offset) return NULL; + f->setup_offset += sz; + return p; + } + return sz ? malloc(sz) : NULL; +} + +static void setup_free(vorb *f, void *p) +{ + if (f->alloc.alloc_buffer) return; // do nothing; setup mem is not a stack + free(p); +} + +static void *setup_temp_malloc(vorb *f, int sz) +{ + sz = (sz+3) & ~3; + if (f->alloc.alloc_buffer) { + if (f->temp_offset - sz < f->setup_offset) return NULL; + f->temp_offset -= sz; + return (char *) f->alloc.alloc_buffer + f->temp_offset; + } + return malloc(sz); +} + +static void setup_temp_free(vorb *f, void *p, size_t sz) +{ + if (f->alloc.alloc_buffer) { + f->temp_offset += (sz+3)&~3; + return; + } + free(p); +} + +#define CRC32_POLY 0x04c11db7 // from spec + +static uint32 crc_table[256]; +static void crc32_init(void) +{ + int i,j; + uint32 s; + for(i=0; i < 256; i++) { + for (s=i<<24, j=0; j < 8; ++j) + s = (s << 1) ^ (s >= (1<<31) ? CRC32_POLY : 0); + crc_table[i] = s; + } +} + +static __forceinline uint32 crc32_update(uint32 crc, uint8 byte) +{ + return (crc << 8) ^ crc_table[byte ^ (crc >> 24)]; +} + + +// used in setup, and for huffman that doesn't go fast path +static unsigned int bit_reverse(unsigned int n) +{ + n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1); + n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2); + n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4); + n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8); + return (n >> 16) | (n << 16); +} + +static float square(float x) +{ + return x*x; +} + +// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3 +// as required by the specification. fast(?) implementation from stb.h +// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup +static int ilog(int32 n) +{ + static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 }; + + // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29) + if (n < (1U << 14)) + if (n < (1U << 4)) return 0 + log2_4[n ]; + else if (n < (1U << 9)) return 5 + log2_4[n >> 5]; + else return 10 + log2_4[n >> 10]; + else if (n < (1U << 24)) + if (n < (1U << 19)) return 15 + log2_4[n >> 15]; + else return 20 + log2_4[n >> 20]; + else if (n < (1U << 29)) return 25 + log2_4[n >> 25]; + else if (n < (1U << 31)) return 30 + log2_4[n >> 30]; + else return 0; // signed n returns 0 +} + +#ifndef M_PI + #define M_PI 3.14159265358979323846264f // from CRC +#endif + +// code length assigned to a value with no huffman encoding +#define NO_CODE 255 + +/////////////////////// LEAF SETUP FUNCTIONS ////////////////////////// +// +// these functions are only called at setup, and only a few times +// per file + +static float float32_unpack(uint32 x) +{ + // from the specification + uint32 mantissa = x & 0x1fffff; + uint32 sign = x & 0x80000000; + uint32 exp = (x & 0x7fe00000) >> 21; + double res = sign ? -(double)mantissa : (double)mantissa; + return (float) ldexp((float)res, exp-788); +} + + +// zlib & jpeg huffman tables assume that the output symbols +// can either be arbitrarily arranged, or have monotonically +// increasing frequencies--they rely on the lengths being sorted; +// this makes for a very simple generation algorithm. +// vorbis allows a huffman table with non-sorted lengths. This +// requires a more sophisticated construction, since symbols in +// order do not map to huffman codes "in order". +static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values) +{ + if (!c->sparse) { + c->codewords [symbol] = huff_code; + } else { + c->codewords [count] = huff_code; + c->codeword_lengths[count] = len; + values [count] = symbol; + } +} + +static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values) +{ + int i,k,m=0; + uint32 available[32]; + + memset(available, 0, sizeof(available)); + // find the first entry + for (k=0; k < n; ++k) if (len[k] < NO_CODE) break; + if (k == n) { assert(c->sorted_entries == 0); return TRUE; } + // add to the list + add_entry(c, 0, k, m++, len[k], values); + // add all available leaves + for (i=1; i <= len[k]; ++i) + available[i] = 1 << (32-i); + // note that the above code treats the first case specially, + // but it's really the same as the following code, so they + // could probably be combined (except the initial code is 0, + // and I use 0 in available[] to mean 'empty') + for (i=k+1; i < n; ++i) { + uint32 res; + int z = len[i], y; + if (z == NO_CODE) continue; + // find lowest available leaf (should always be earliest, + // which is what the specification calls for) + // note that this property, and the fact we can never have + // more than one free leaf at a given level, isn't totally + // trivial to prove, but it seems true and the assert never + // fires, so! + while (z > 0 && !available[z]) --z; + if (z == 0) { assert(0); return FALSE; } + res = available[z]; + available[z] = 0; + add_entry(c, bit_reverse(res), i, m++, len[i], values); + // propogate availability up the tree + if (z != len[i]) { + for (y=len[i]; y > z; --y) { + assert(available[y] == 0); + available[y] = res + (1 << (32-y)); + } + } + } + return TRUE; +} + +// accelerated huffman table allows fast O(1) match of all symbols +// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH +static void compute_accelerated_huffman(Codebook *c) +{ + int i, len; + for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) + c->fast_huffman[i] = -1; + + len = c->sparse ? c->sorted_entries : c->entries; + #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT + if (len > 32767) len = 32767; // largest possible value we can encode! + #endif + for (i=0; i < len; ++i) { + if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { + uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; + // set table entries for all bit combinations in the higher bits + while (z < FAST_HUFFMAN_TABLE_SIZE) { + c->fast_huffman[z] = i; + z += 1 << c->codeword_lengths[i]; + } + } + } +} + +static int uint32_compare(const void *p, const void *q) +{ + uint32 x = * (uint32 *) p; + uint32 y = * (uint32 *) q; + return x < y ? -1 : x > y; +} + +static int include_in_sort(Codebook *c, uint8 len) +{ + if (c->sparse) { assert(len != NO_CODE); return TRUE; } + if (len == NO_CODE) return FALSE; + if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; + return FALSE; +} + +// if the fast table above doesn't work, we want to binary +// search them... need to reverse the bits +static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) +{ + int i, len; + // build a list of all the entries + // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. + // this is kind of a frivolous optimization--I don't see any performance improvement, + // but it's like 4 extra lines of code, so. + if (!c->sparse) { + int k = 0; + for (i=0; i < c->entries; ++i) + if (include_in_sort(c, lengths[i])) + c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); + assert(k == c->sorted_entries); + } else { + for (i=0; i < c->sorted_entries; ++i) + c->sorted_codewords[i] = bit_reverse(c->codewords[i]); + } + + qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); + c->sorted_codewords[c->sorted_entries] = 0xffffffff; + + len = c->sparse ? c->sorted_entries : c->entries; + // now we need to indicate how they correspond; we could either + // #1: sort a different data structure that says who they correspond to + // #2: for each sorted entry, search the original list to find who corresponds + // #3: for each original entry, find the sorted entry + // #1 requires extra storage, #2 is slow, #3 can use binary search! + for (i=0; i < len; ++i) { + int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; + if (include_in_sort(c,huff_len)) { + uint32 code = bit_reverse(c->codewords[i]); + int x=0, n=c->sorted_entries; + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + assert(c->sorted_codewords[x] == code); + if (c->sparse) { + c->sorted_values[x] = values[i]; + c->codeword_lengths[x] = huff_len; + } else { + c->sorted_values[x] = i; + } + } + } +} + +// only run while parsing the header (3 times) +static int vorbis_validate(uint8 *data) +{ + static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; + return memcmp(data, vorbis, 6) == 0; +} + +// called from setup only, once per code book +// (formula implied by specification) +static int lookup1_values(int entries, int dim) +{ + int r = (int) floor(exp((float) log((float) entries) / dim)); + if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; + ++r; // floor() to avoid _ftol() when non-CRT + assert(pow((float) r+1, dim) > entries); + assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above + return r; +} + +// called twice per file +static void compute_twiddle_factors(int n, float *A, float *B, float *C) +{ + int n4 = n >> 2, n8 = n >> 3; + int k,k2; + + for (k=k2=0; k < n4; ++k,k2+=2) { + A[k2 ] = (float) cos(4*k*M_PI/n); + A[k2+1] = (float) -sin(4*k*M_PI/n); + B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f; + B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f; + } + for (k=k2=0; k < n8; ++k,k2+=2) { + C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); + C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); + } +} + +static void compute_window(int n, float *window) +{ + int n2 = n >> 1, i; + for (i=0; i < n2; ++i) + window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI))); +} + +static void compute_bitreverse(int n, uint16 *rev) +{ + int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + int i, n8 = n >> 3; + for (i=0; i < n8; ++i) + rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2; +} + +static int init_blocksize(vorb *f, int b, int n) +{ + int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3; + f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2); + f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2); + f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4); + if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem); + compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]); + f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2); + if (!f->window[b]) return error(f, VORBIS_outofmem); + compute_window(n, f->window[b]); + f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8); + if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem); + compute_bitreverse(n, f->bit_reverse[b]); + return TRUE; +} + +static void neighbors(uint16 *x, int n, int *plow, int *phigh) +{ + int low = -1; + int high = 65536; + int i; + for (i=0; i < n; ++i) { + if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; } + if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; } + } +} + +// this has been repurposed so y is now the original index instead of y +typedef struct +{ + uint16 x,y; +} Point; + +int point_compare(const void *p, const void *q) +{ + Point *a = (Point *) p; + Point *b = (Point *) q; + return a->x < b->x ? -1 : a->x > b->x; +} + +// +/////////////////////// END LEAF SETUP FUNCTIONS ////////////////////////// + + +#if defined(STB_VORBIS_NO_STDIO) + #define USE_MEMORY(z) TRUE +#else + #define USE_MEMORY(z) ((z)->stream) +#endif + +static uint8 get8(vorb *z) +{ + if (USE_MEMORY(z)) { + if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; } + return *z->stream++; + } + + #ifndef STB_VORBIS_NO_STDIO + { + int c = fgetc(z->f); + if (c == EOF) { z->eof = TRUE; return 0; } + return c; + } + #endif +} + +static uint32 get32(vorb *f) +{ + uint32 x; + x = get8(f); + x += get8(f) << 8; + x += get8(f) << 16; + x += get8(f) << 24; + return x; +} + +static int getn(vorb *z, uint8 *data, int n) +{ + if (USE_MEMORY(z)) { + if (z->stream+n > z->stream_end) { z->eof = 1; return 0; } + memcpy(data, z->stream, n); + z->stream += n; + return 1; + } + + #ifndef STB_VORBIS_NO_STDIO + if (fread(data, n, 1, z->f) == 1) + return 1; + else { + z->eof = 1; + return 0; + } + #endif +} + +static void skip(vorb *z, int n) +{ + if (USE_MEMORY(z)) { + z->stream += n; + if (z->stream >= z->stream_end) z->eof = 1; + return; + } + #ifndef STB_VORBIS_NO_STDIO + { + long x = ftell(z->f); + fseek(z->f, x+n, SEEK_SET); + } + #endif +} + +static int set_file_offset(stb_vorbis *f, unsigned int loc) +{ + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (f->push_mode) return 0; + #endif + f->eof = 0; + if (USE_MEMORY(f)) { + if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) { + f->stream = f->stream_end; + f->eof = 1; + return 0; + } else { + f->stream = f->stream_start + loc; + return 1; + } + } + #ifndef STB_VORBIS_NO_STDIO + if (loc + f->f_start < loc || loc >= 0x80000000) { + loc = 0x7fffffff; + f->eof = 1; + } else { + loc += f->f_start; + } + if (!fseek(f->f, loc, SEEK_SET)) + return 1; + f->eof = 1; + fseek(f->f, f->f_start, SEEK_END); + return 0; + #endif +} + + +static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 }; + +static int capture_pattern(vorb *f) +{ + if (0x4f != get8(f)) return FALSE; + if (0x67 != get8(f)) return FALSE; + if (0x67 != get8(f)) return FALSE; + if (0x53 != get8(f)) return FALSE; + return TRUE; +} + +#define PAGEFLAG_continued_packet 1 +#define PAGEFLAG_first_page 2 +#define PAGEFLAG_last_page 4 + +static int start_page_no_capturepattern(vorb *f) +{ + uint32 loc0,loc1,n,i; + // stream structure version + if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version); + // header flag + f->page_flag = get8(f); + // absolute granule position + loc0 = get32(f); + loc1 = get32(f); + // @TODO: validate loc0,loc1 as valid positions? + // stream serial number -- vorbis doesn't interleave, so discard + get32(f); + //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number); + // page sequence number + n = get32(f); + f->last_page = n; + // CRC32 + get32(f); + // page_segments + f->segment_count = get8(f); + if (!getn(f, f->segments, f->segment_count)) + return error(f, VORBIS_unexpected_eof); + // assume we _don't_ know any the sample position of any segments + f->end_seg_with_known_loc = -2; + if (loc0 != ~0 || loc1 != ~0) { + // determine which packet is the last one that will complete + for (i=f->segment_count-1; i >= 0; --i) + if (f->segments[i] < 255) + break; + // 'i' is now the index of the _last_ segment of a packet that ends + if (i >= 0) { + f->end_seg_with_known_loc = i; + f->known_loc_for_packet = loc0; + } + } + if (f->first_decode) { + int i,len; + ProbedPage p; + len = 0; + for (i=0; i < f->segment_count; ++i) + len += f->segments[i]; + len += 27 + f->segment_count; + p.page_start = f->first_audio_page_offset; + p.page_end = p.page_start + len; + p.after_previous_page_start = p.page_start; + p.first_decoded_sample = 0; + p.last_decoded_sample = loc0; + f->p_first = p; + } + f->next_seg = 0; + return TRUE; +} + +static int start_page(vorb *f) +{ + if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern); + return start_page_no_capturepattern(f); +} + +static int start_packet(vorb *f) +{ + while (f->next_seg == -1) { + if (!start_page(f)) return FALSE; + if (f->page_flag & PAGEFLAG_continued_packet) + return error(f, VORBIS_continued_packet_flag_invalid); + } + f->last_seg = FALSE; + f->valid_bits = 0; + f->packet_bytes = 0; + f->bytes_in_seg = 0; + // f->next_seg is now valid + return TRUE; +} + +static int maybe_start_packet(vorb *f) +{ + if (f->next_seg == -1) { + int x = get8(f); + if (f->eof) return FALSE; // EOF at page boundary is not an error! + if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern); + if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern); + if (!start_page_no_capturepattern(f)) return FALSE; + if (f->page_flag & PAGEFLAG_continued_packet) { + // set up enough state that we can read this packet if we want, + // e.g. during recovery + f->last_seg = FALSE; + f->bytes_in_seg = 0; + return error(f, VORBIS_continued_packet_flag_invalid); + } + } + return start_packet(f); +} + +static int next_segment(vorb *f) +{ + int len; + if (f->last_seg) return 0; + if (f->next_seg == -1) { + f->last_seg_which = f->segment_count-1; // in case start_page fails + if (!start_page(f)) { f->last_seg = 1; return 0; } + if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid); + } + len = f->segments[f->next_seg++]; + if (len < 255) { + f->last_seg = TRUE; + f->last_seg_which = f->next_seg-1; + } + if (f->next_seg >= f->segment_count) + f->next_seg = -1; + assert(f->bytes_in_seg == 0); + f->bytes_in_seg = len; + return len; +} + +#define EOP (-1) +#define INVALID_BITS (-1) + +static int get8_packet_raw(vorb *f) +{ + if (!f->bytes_in_seg) { + if (f->last_seg) return EOP; + else if (!next_segment(f)) return EOP; + } + assert(f->bytes_in_seg > 0); + --f->bytes_in_seg; + ++f->packet_bytes; + return get8(f); +} + +static int get8_packet(vorb *f) +{ + int x = get8_packet_raw(f); + f->valid_bits = 0; + return x; +} + +static void flush_packet(vorb *f) +{ + while (get8_packet_raw(f) != EOP); +} + +// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important +// as the huffman decoder? +static uint32 get_bits(vorb *f, int n) +{ + uint32 z; + + if (f->valid_bits < 0) return 0; + if (f->valid_bits < n) { + if (n > 24) { + // the accumulator technique below would not work correctly in this case + z = get_bits(f, 24); + z += get_bits(f, n-24) << 24; + return z; + } + if (f->valid_bits == 0) f->acc = 0; + while (f->valid_bits < n) { + int z = get8_packet_raw(f); + if (z == EOP) { + f->valid_bits = INVALID_BITS; + return 0; + } + f->acc += z << f->valid_bits; + f->valid_bits += 8; + } + } + if (f->valid_bits < 0) return 0; + z = f->acc & ((1 << n)-1); + f->acc >>= n; + f->valid_bits -= n; + return z; +} + +/* +static int32 get_bits_signed(vorb *f, int n) { + uint32 z = get_bits(f, n); + if (z & (1 << (n-1))) + z += ~((1 << n) - 1); + return (int32) z; +} +*/ + +// @OPTIMIZE: primary accumulator for huffman +// expand the buffer to as many bits as possible without reading off end of packet +// it might be nice to allow f->valid_bits and f->acc to be stored in registers, +// e.g. cache them locally and decode locally +static __forceinline void prep_huffman(vorb *f) +{ + if (f->valid_bits <= 24) { + if (f->valid_bits == 0) f->acc = 0; + do { + int z; + if (f->last_seg && !f->bytes_in_seg) return; + z = get8_packet_raw(f); + if (z == EOP) return; + f->acc += z << f->valid_bits; + f->valid_bits += 8; + } while (f->valid_bits <= 24); + } +} + +enum +{ + VORBIS_packet_id = 1, + VORBIS_packet_comment = 3, + VORBIS_packet_setup = 5, +}; + +static int codebook_decode_scalar_raw(vorb *f, Codebook *c) +{ + int i; + prep_huffman(f); + + assert(c->sorted_codewords || c->codewords); + // cases to use binary search: sorted_codewords && !c->codewords + // sorted_codewords && c->entries > 8 + if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) { + // binary search + uint32 code = bit_reverse(f->acc); + int x=0, n=c->sorted_entries, len; + + while (n > 1) { + // invariant: sc[x] <= code < sc[x+n] + int m = x + (n >> 1); + if (c->sorted_codewords[m] <= code) { + x = m; + n -= (n>>1); + } else { + n >>= 1; + } + } + // x is now the sorted index + if (!c->sparse) x = c->sorted_values[x]; + // x is now sorted index if sparse, or symbol otherwise + len = c->codeword_lengths[x]; + if (f->valid_bits >= len) { + f->acc >>= len; + f->valid_bits -= len; + return x; + } + + f->valid_bits = 0; + return -1; + } + + // if small, linear search + assert(!c->sparse); + for (i=0; i < c->entries; ++i) { + if (c->codeword_lengths[i] == NO_CODE) continue; + if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) { + if (f->valid_bits >= c->codeword_lengths[i]) { + f->acc >>= c->codeword_lengths[i]; + f->valid_bits -= c->codeword_lengths[i]; + return i; + } + f->valid_bits = 0; + return -1; + } + } + + error(f, VORBIS_invalid_stream); + f->valid_bits = 0; + return -1; +} + +/* +static int codebook_decode_scalar(vorb *f, Codebook *c) { + int i; + if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) + prep_huffman(f); + // fast huffman table lookup + i = f->acc & FAST_HUFFMAN_TABLE_MASK; + i = c->fast_huffman[i]; + if (i >= 0) { + f->acc >>= c->codeword_lengths[i]; + f->valid_bits -= c->codeword_lengths[i]; + if (f->valid_bits < 0) { f->valid_bits = 0; return -1; } + return i; + } + return codebook_decode_scalar_raw(f,c); +} +*/ + +#ifndef STB_VORBIS_NO_INLINE_DECODE + +#define DECODE_RAW(var, f,c) \ + if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \ + prep_huffman(f); \ + var = f->acc & FAST_HUFFMAN_TABLE_MASK; \ + var = c->fast_huffman[var]; \ + if (var >= 0) { \ + int n = c->codeword_lengths[var]; \ + f->acc >>= n; \ + f->valid_bits -= n; \ + if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \ + } else { \ + var = codebook_decode_scalar_raw(f,c); \ + } + +#else + +#define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c); + +#endif + +#define DECODE(var,f,c) \ + DECODE_RAW(var,f,c) \ + if (c->sparse) var = c->sorted_values[var]; + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + #define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c) +#else + #define DECODE_VQ(var,f,c) DECODE(var,f,c) +#endif + + + + + + +// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case +// where we avoid one addition +#ifndef STB_VORBIS_CODEBOOK_FLOATS + #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) + #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) + #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) +#else + #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) + #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) + #define CODEBOOK_ELEMENT_BASE(c) (0) +#endif + +static int codebook_decode_start(vorb *f, Codebook *c, int len) +{ + int z = -1; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) + error(f, VORBIS_invalid_stream); + else { + DECODE_VQ(z,f,c); + if (c->sparse) assert(z < c->sorted_entries); + if (z < 0) { // check for EOP + if (!f->bytes_in_seg) + if (f->last_seg) + return z; + error(f, VORBIS_invalid_stream); + } + } + return z; +} + +static int codebook_decode(vorb *f, Codebook *c, float *output, int len) +{ + int i,z = codebook_decode_start(f,c,len); + if (z < 0) return FALSE; + if (len > c->dimensions) len = c->dimensions; + +#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + float last = CODEBOOK_ELEMENT_BASE(c); + int div = 1; + for (i=0; i < len; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + output[i] += val; + if (c->sequence_p) last = val + c->minimum_value; + div *= c->lookup_values; + } + return TRUE; + } +#endif + + z *= c->dimensions; + if (c->sequence_p) { + float last = CODEBOOK_ELEMENT_BASE(c); + for (i=0; i < len; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + output[i] += val; + last = val + c->minimum_value; + } + } else { + float last = CODEBOOK_ELEMENT_BASE(c); + for (i=0; i < len; ++i) { + output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; + } + } + + return TRUE; +} + +static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step) +{ + int i,z = codebook_decode_start(f,c,len); + float last = CODEBOOK_ELEMENT_BASE(c); + if (z < 0) return FALSE; + if (len > c->dimensions) len = c->dimensions; + +#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int div = 1; + for (i=0; i < len; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + output[i*step] += val; + if (c->sequence_p) last = val; + div *= c->lookup_values; + } + return TRUE; + } +#endif + + z *= c->dimensions; + for (i=0; i < len; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + output[i*step] += val; + if (c->sequence_p) last = val; + } + + return TRUE; +} + +static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode) +{ + int c_inter = *c_inter_p; + int p_inter = *p_inter_p; + int i,z, effective = c->dimensions; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); + + while (total_decode > 0) { + float last = CODEBOOK_ELEMENT_BASE(c); + DECODE_VQ(z,f,c); + #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + assert(!c->sparse || z < c->sorted_entries); + #endif + if (z < 0) { + if (!f->bytes_in_seg) + if (f->last_seg) return FALSE; + return error(f, VORBIS_invalid_stream); + } + + // if this will take us off the end of the buffers, stop short! + // we check by computing the length of the virtual interleaved + // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), + // and the length we'll be using (effective) + if (c_inter + p_inter*ch + effective > len * ch) { + effective = len*ch - (p_inter*ch - c_inter); + } + + #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int div = 1; + for (i=0; i < effective; ++i) { + int off = (z / div) % c->lookup_values; + float val = CODEBOOK_ELEMENT_FAST(c,off) + last; + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + if (c->sequence_p) last = val; + div *= c->lookup_values; + } + } else + #endif + { + z *= c->dimensions; + if (c->sequence_p) { + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + last = val; + } + } else { + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + outputs[c_inter][p_inter] += val; + if (++c_inter == ch) { c_inter = 0; ++p_inter; } + } + } + } + + total_decode -= effective; + } + *c_inter_p = c_inter; + *p_inter_p = p_inter; + return TRUE; +} + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK +static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, int *c_inter_p, int *p_inter_p, int len, int total_decode) +{ + int c_inter = *c_inter_p; + int p_inter = *p_inter_p; + int i,z, effective = c->dimensions; + + // type 0 is only legal in a scalar context + if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); + + while (total_decode > 0) { + float last = CODEBOOK_ELEMENT_BASE(c); + DECODE_VQ(z,f,c); + + if (z < 0) { + if (!f->bytes_in_seg) + if (f->last_seg) return FALSE; + return error(f, VORBIS_invalid_stream); + } + + // if this will take us off the end of the buffers, stop short! + // we check by computing the length of the virtual interleaved + // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), + // and the length we'll be using (effective) + if (c_inter + p_inter*2 + effective > len * 2) { + effective = len*2 - (p_inter*2 - c_inter); + } + + { + z *= c->dimensions; + if (c->sequence_p) { + // haven't optimized this case because I don't have any examples + for (i=0; i < effective; ++i) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + outputs[c_inter][p_inter] += val; + if (++c_inter == 2) { c_inter = 0; ++p_inter; } + last = val; + } + } else { + i=0; + if (c_inter == 1) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + outputs[c_inter][p_inter] += val; + c_inter = 0; ++p_inter; + ++i; + } + { + float *z0 = outputs[0]; + float *z1 = outputs[1]; + for (; i+1 < effective;) { + z0[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; + z1[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; + ++p_inter; + i += 2; + } + } + if (i < effective) { + float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; + outputs[c_inter][p_inter] += val; + if (++c_inter == 2) { c_inter = 0; ++p_inter; } + } + } + } + + total_decode -= effective; + } + *c_inter_p = c_inter; + *p_inter_p = p_inter; + return TRUE; +} +#endif + +static int predict_point(int x, int x0, int x1, int y0, int y1) +{ + int dy = y1 - y0; + int adx = x1 - x0; + // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86? + int err = abs(dy) * (x - x0); + int off = err / adx; + return dy < 0 ? y0 - off : y0 + off; +} + +// the following table is block-copied from the specification +static float inverse_db_table[256] = +{ + 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, + 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, + 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, + 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, + 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, + 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, + 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, + 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, + 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, + 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, + 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, + 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, + 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, + 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, + 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, + 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, + 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, + 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, + 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, + 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, + 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, + 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, + 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, + 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, + 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, + 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, + 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, + 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, + 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, + 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, + 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, + 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, + 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, + 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, + 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, + 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, + 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f, + 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f, + 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f, + 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f, + 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f, + 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f, + 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f, + 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f, + 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f, + 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f, + 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f, + 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f, + 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f, + 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f, + 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f, + 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f, + 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f, + 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f, + 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f, + 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f, + 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f, + 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f, + 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f, + 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f, + 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f, + 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f, + 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, + 0.82788260f, 0.88168307f, 0.9389798f, 1.0f +}; + + +// @OPTIMIZE: if you want to replace this bresenham line-drawing routine, +// note that you must produce bit-identical output to decode correctly; +// this specific sequence of operations is specified in the spec (it's +// drawing integer-quantized frequency-space lines that the encoder +// expects to be exactly the same) +// ... also, isn't the whole point of Bresenham's algorithm to NOT +// have to divide in the setup? sigh. +#ifndef STB_VORBIS_NO_DEFER_FLOOR +#define LINE_OP(a,b) a *= b +#else +#define LINE_OP(a,b) a = b +#endif + +#ifdef STB_VORBIS_DIVIDE_TABLE +#define DIVTAB_NUMER 32 +#define DIVTAB_DENOM 64 +int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB +#endif + +static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n) +{ + int dy = y1 - y0; + int adx = x1 - x0; + int ady = abs(dy); + int base; + int x=x0,y=y0; + int err = 0; + int sy; + +#ifdef STB_VORBIS_DIVIDE_TABLE + if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) { + if (dy < 0) { + base = -integer_divide_table[ady][adx]; + sy = base-1; + } else { + base = integer_divide_table[ady][adx]; + sy = base+1; + } + } else { + base = dy / adx; + if (dy < 0) + sy = base - 1; + else + sy = base+1; + } +#else + base = dy / adx; + if (dy < 0) + sy = base - 1; + else + sy = base+1; +#endif + ady -= abs(base) * adx; + if (x1 > n) x1 = n; + LINE_OP(output[x], inverse_db_table[y]); + for (++x; x < x1; ++x) { + err += ady; + if (err >= adx) { + err -= adx; + y += sy; + } else + y += base; + LINE_OP(output[x], inverse_db_table[y]); + } +} + +static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype) +{ + int k; + if (rtype == 0) { + int step = n / book->dimensions; + for (k=0; k < step; ++k) + if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step)) + return FALSE; + } else { + for (k=0; k < n; ) { + if (!codebook_decode(f, book, target+offset, n-k)) + return FALSE; + k += book->dimensions; + offset += book->dimensions; + } + } + return TRUE; +} + +static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode) +{ + int i,j,pass; + Residue *r = f->residue_config + rn; + int rtype = f->residue_types[rn]; + int c = r->classbook; + int classwords = f->codebooks[c].dimensions; + int n_read = r->end - r->begin; + int part_read = n_read / r->part_size; + int temp_alloc_point = temp_alloc_save(f); + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata)); + #else + int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications)); + #endif + + for (i=0; i < ch; ++i) + if (!do_not_decode[i]) + memset(residue_buffers[i], 0, sizeof(float) * n); + + if (rtype == 2 && ch != 1) { + //int len = ch * n; + for (j=0; j < ch; ++j) + if (!do_not_decode[j]) + break; + if (j == ch) + goto done; + + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set = 0; + if (ch == 2) { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = (z & 1), p_inter = z>>1; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + #else + // saves 1% + if (!codebook_decode_deinterleave_repeat_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) + goto done; + #endif + } else { + z += r->part_size; + c_inter = z & 1; + p_inter = z >> 1; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } else if (ch == 1) { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = 0, p_inter = z; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + } else { + z += r->part_size; + c_inter = 0; + p_inter = z; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } else { + while (pcount < part_read) { + int z = r->begin + pcount*r->part_size; + int c_inter = z % ch, p_inter = z/ch; + if (pass == 0) { + Codebook *c = f->codebooks+r->classbook; + int q; + DECODE(q,f,c); + if (q == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[0][class_set] = r->classdata[q]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[0][i+pcount] = q % r->classifications; + q /= r->classifications; + } + #endif + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + int z = r->begin + pcount*r->part_size; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[0][class_set][i]; + #else + int c = classifications[0][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + Codebook *book = f->codebooks + b; + if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) + goto done; + } else { + z += r->part_size; + c_inter = z % ch; + p_inter = z / ch; + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + } + goto done; + } + + for (pass=0; pass < 8; ++pass) { + int pcount = 0, class_set=0; + while (pcount < part_read) { + if (pass == 0) { + for (j=0; j < ch; ++j) { + if (!do_not_decode[j]) { + Codebook *c = f->codebooks+r->classbook; + int temp; + DECODE(temp,f,c); + if (temp == EOP) goto done; + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + part_classdata[j][class_set] = r->classdata[temp]; + #else + for (i=classwords-1; i >= 0; --i) { + classifications[j][i+pcount] = temp % r->classifications; + temp /= r->classifications; + } + #endif + } + } + } + for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { + for (j=0; j < ch; ++j) { + if (!do_not_decode[j]) { + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + int c = part_classdata[j][class_set][i]; + #else + int c = classifications[j][pcount]; + #endif + int b = r->residue_books[c][pass]; + if (b >= 0) { + float *target = residue_buffers[j]; + int offset = r->begin + pcount * r->part_size; + int n = r->part_size; + Codebook *book = f->codebooks + b; + if (!residue_decode(f, book, target, offset, n, rtype)) + goto done; + } + } + } + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + ++class_set; + #endif + } + } + done: + temp_alloc_restore(f,temp_alloc_point); +} + + +#if 0 +// slow way for debugging +void inverse_mdct_slow(float *buffer, int n) +{ + int i,j; + int n2 = n >> 1; + float *x = (float *) malloc(sizeof(*x) * n2); + memcpy(x, buffer, sizeof(*x) * n2); + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n2; ++j) + // formula from paper: + //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); + // formula from wikipedia + //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); + // these are equivalent, except the formula from the paper inverts the multiplier! + // however, what actually works is NO MULTIPLIER!?! + //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); + acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); + buffer[i] = acc; + } + free(x); +} +#elif 0 +// same as above, but just barely able to run in real time on modern machines +void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) +{ + float mcos[16384]; + int i,j; + int n2 = n >> 1, nmask = (n << 2) -1; + float *x = (float *) malloc(sizeof(*x) * n2); + memcpy(x, buffer, sizeof(*x) * n2); + for (i=0; i < 4*n; ++i) + mcos[i] = (float) cos(M_PI / 2 * i / n); + + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n2; ++j) + acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask]; + buffer[i] = acc; + } + free(x); +} +#else +// transform to use a slow dct-iv; this is STILL basically trivial, +// but only requires half as many ops +void dct_iv_slow(float *buffer, int n) +{ + float mcos[16384]; + float x[2048]; + int i,j; + //int n2 = n >> 1; + int nmask = (n << 3) - 1; + memcpy(x, buffer, sizeof(*x) * n); + for (i=0; i < 8*n; ++i) + mcos[i] = (float) cos(M_PI / 4 * i / n); + for (i=0; i < n; ++i) { + float acc = 0; + for (j=0; j < n; ++j) + acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask]; + //acc += x[j] * cos(M_PI / n * (i + 0.5) * (j + 0.5)); + buffer[i] = acc; + } + free(x); +} + +void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) +{ + int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4; + float temp[4096]; + + memcpy(temp, buffer, n2 * sizeof(float)); + dct_iv_slow(temp, n2); // returns -c'-d, a-b' + + for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b' + for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d' + for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d +} +#endif + +#ifndef LIBVORBIS_MDCT +#define LIBVORBIS_MDCT 0 +#endif + +#if LIBVORBIS_MDCT +// directly call the vorbis MDCT using an interface documented +// by Jeff Roberts... useful for performance comparison +typedef struct +{ + int n; + int log2n; + + float *trig; + int *bitrev; + + float scale; +} mdct_lookup; + +extern void mdct_init(mdct_lookup *lookup, int n); +extern void mdct_clear(mdct_lookup *l); +extern void mdct_backward(mdct_lookup *init, float *in, float *out); + +mdct_lookup M1,M2; + +void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) +{ + mdct_lookup *M; + if (M1.n == n) M = &M1; + else if (M2.n == n) M = &M2; + else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; } + else { + if (M2.n) __asm int 3; + mdct_init(&M2, n); + M = &M2; + } + + mdct_backward(M, buffer, buffer); +} +#endif + + +// the following were split out into separate functions while optimizing; +// they could be pushed back up but eh. __forceinline showed no change; +// they're probably already being inlined. +static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A) +{ + float *ee0 = e + i_off; + float *ee2 = ee0 + k_off; + int i; + + assert((n & 3) == 0); + for (i=(n>>2); i > 0; --i) { + float k00_20, k01_21; + k00_20 = ee0[ 0] - ee2[ 0]; + k01_21 = ee0[-1] - ee2[-1]; + ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0]; + ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1]; + ee2[ 0] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-1] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-2] - ee2[-2]; + k01_21 = ee0[-3] - ee2[-3]; + ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2]; + ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3]; + ee2[-2] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-3] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-4] - ee2[-4]; + k01_21 = ee0[-5] - ee2[-5]; + ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4]; + ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5]; + ee2[-4] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-5] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + + k00_20 = ee0[-6] - ee2[-6]; + k01_21 = ee0[-7] - ee2[-7]; + ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6]; + ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7]; + ee2[-6] = k00_20 * A[0] - k01_21 * A[1]; + ee2[-7] = k01_21 * A[0] + k00_20 * A[1]; + A += 8; + ee0 -= 8; + ee2 -= 8; + } +} + +static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1) +{ + int i; + float k00_20, k01_21; + + float *e0 = e + d0; + float *e2 = e0 + k_off; + + for (i=lim >> 2; i > 0; --i) { + k00_20 = e0[-0] - e2[-0]; + k01_21 = e0[-1] - e2[-1]; + e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0]; + e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1]; + e2[-0] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-1] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-2] - e2[-2]; + k01_21 = e0[-3] - e2[-3]; + e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2]; + e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3]; + e2[-2] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-3] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-4] - e2[-4]; + k01_21 = e0[-5] - e2[-5]; + e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4]; + e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5]; + e2[-4] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-5] = (k01_21)*A[0] + (k00_20) * A[1]; + + A += k1; + + k00_20 = e0[-6] - e2[-6]; + k01_21 = e0[-7] - e2[-7]; + e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6]; + e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7]; + e2[-6] = (k00_20)*A[0] - (k01_21) * A[1]; + e2[-7] = (k01_21)*A[0] + (k00_20) * A[1]; + + e0 -= 8; + e2 -= 8; + + A += k1; + } +} + +static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0) +{ + int i; + float A0 = A[0]; + float A1 = A[0+1]; + float A2 = A[0+a_off]; + float A3 = A[0+a_off+1]; + float A4 = A[0+a_off*2+0]; + float A5 = A[0+a_off*2+1]; + float A6 = A[0+a_off*3+0]; + float A7 = A[0+a_off*3+1]; + + float k00,k11; + + float *ee0 = e +i_off; + float *ee2 = ee0+k_off; + + for (i=n; i > 0; --i) { + k00 = ee0[ 0] - ee2[ 0]; + k11 = ee0[-1] - ee2[-1]; + ee0[ 0] = ee0[ 0] + ee2[ 0]; + ee0[-1] = ee0[-1] + ee2[-1]; + ee2[ 0] = (k00) * A0 - (k11) * A1; + ee2[-1] = (k11) * A0 + (k00) * A1; + + k00 = ee0[-2] - ee2[-2]; + k11 = ee0[-3] - ee2[-3]; + ee0[-2] = ee0[-2] + ee2[-2]; + ee0[-3] = ee0[-3] + ee2[-3]; + ee2[-2] = (k00) * A2 - (k11) * A3; + ee2[-3] = (k11) * A2 + (k00) * A3; + + k00 = ee0[-4] - ee2[-4]; + k11 = ee0[-5] - ee2[-5]; + ee0[-4] = ee0[-4] + ee2[-4]; + ee0[-5] = ee0[-5] + ee2[-5]; + ee2[-4] = (k00) * A4 - (k11) * A5; + ee2[-5] = (k11) * A4 + (k00) * A5; + + k00 = ee0[-6] - ee2[-6]; + k11 = ee0[-7] - ee2[-7]; + ee0[-6] = ee0[-6] + ee2[-6]; + ee0[-7] = ee0[-7] + ee2[-7]; + ee2[-6] = (k00) * A6 - (k11) * A7; + ee2[-7] = (k11) * A6 + (k00) * A7; + + ee0 -= k0; + ee2 -= k0; + } +} + +static __forceinline void iter_54(float *z) +{ + float k00,k11,k22,k33; + float y0,y1,y2,y3; + + k00 = z[ 0] - z[-4]; + y0 = z[ 0] + z[-4]; + y2 = z[-2] + z[-6]; + k22 = z[-2] - z[-6]; + + z[-0] = y0 + y2; // z0 + z4 + z2 + z6 + z[-2] = y0 - y2; // z0 + z4 - z2 - z6 + + // done with y0,y2 + + k33 = z[-3] - z[-7]; + + z[-4] = k00 + k33; // z0 - z4 + z3 - z7 + z[-6] = k00 - k33; // z0 - z4 - z3 + z7 + + // done with k33 + + k11 = z[-1] - z[-5]; + y1 = z[-1] + z[-5]; + y3 = z[-3] + z[-7]; + + z[-1] = y1 + y3; // z1 + z5 + z3 + z7 + z[-3] = y1 - y3; // z1 + z5 - z3 - z7 + z[-5] = k11 - k22; // z1 - z5 + z2 - z6 + z[-7] = k11 + k22; // z1 - z5 - z2 + z6 +} + +static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n) { + //int k_off = -8; + int a_off = base_n >> 3; + float A2 = A[0+a_off]; + float *z = e + i_off; + float *base = z - 16 * n; + + while (z > base) { + float k00,k11; + + k00 = z[-0] - z[-8]; + k11 = z[-1] - z[-9]; + z[-0] = z[-0] + z[-8]; + z[-1] = z[-1] + z[-9]; + z[-8] = k00; + z[-9] = k11 ; + + k00 = z[ -2] - z[-10]; + k11 = z[ -3] - z[-11]; + z[ -2] = z[ -2] + z[-10]; + z[ -3] = z[ -3] + z[-11]; + z[-10] = (k00+k11) * A2; + z[-11] = (k11-k00) * A2; + + k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation + k11 = z[ -5] - z[-13]; + z[ -4] = z[ -4] + z[-12]; + z[ -5] = z[ -5] + z[-13]; + z[-12] = k11; + z[-13] = k00; + + k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation + k11 = z[ -7] - z[-15]; + z[ -6] = z[ -6] + z[-14]; + z[ -7] = z[ -7] + z[-15]; + z[-14] = (k00+k11) * A2; + z[-15] = (k00-k11) * A2; + + iter_54(z); + iter_54(z-8); + z -= 16; + } +} + +static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) { + int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; + //int n3_4 = n - n4; + int ld; + // @OPTIMIZE: reduce register pressure by using fewer variables? + int save_point = temp_alloc_save(f); + float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2)); + float *u=NULL,*v=NULL; + // twiddle factors + float *A = f->A[blocktype]; + + // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" + // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function. + + // kernel from paper + + + // merged: + // copy and reflect spectral data + // step 0 + + // note that it turns out that the items added together during + // this step are, in fact, being added to themselves (as reflected + // by step 0). inexplicable inefficiency! this became obvious + // once I combined the passes. + + // so there's a missing 'times 2' here (for adding X to itself). + // this propogates through linearly to the end, where the numbers + // are 1/2 too small, and need to be compensated for. + + { + float *d,*e, *AA, *e_stop; + d = &buf2[n2-2]; + AA = A; + e = &buffer[0]; + e_stop = &buffer[n2]; + while (e != e_stop) { + d[1] = (e[0] * AA[0] - e[2]*AA[1]); + d[0] = (e[0] * AA[1] + e[2]*AA[0]); + d -= 2; + AA += 2; + e += 4; + } + + e = &buffer[n2-3]; + while (d >= buf2) { + d[1] = (-e[2] * AA[0] - -e[0]*AA[1]); + d[0] = (-e[2] * AA[1] + -e[0]*AA[0]); + d -= 2; + AA += 2; + e -= 4; + } + } + + // now we use symbolic names for these, so that we can + // possibly swap their meaning as we change which operations + // are in place + + u = buffer; + v = buf2; + + // step 2 (paper output is w, now u) + // this could be in place, but the data ends up in the wrong + // place... _somebody_'s got to swap it, so this is nominated + { + float *AA = &A[n2-8]; + float *d0,*d1, *e0, *e1; + + e0 = &v[n4]; + e1 = &v[0]; + + d0 = &u[n4]; + d1 = &u[0]; + + while (AA >= A) { + float v40_20, v41_21; + + v41_21 = e0[1] - e1[1]; + v40_20 = e0[0] - e1[0]; + d0[1] = e0[1] + e1[1]; + d0[0] = e0[0] + e1[0]; + d1[1] = v41_21*AA[4] - v40_20*AA[5]; + d1[0] = v40_20*AA[4] + v41_21*AA[5]; + + v41_21 = e0[3] - e1[3]; + v40_20 = e0[2] - e1[2]; + d0[3] = e0[3] + e1[3]; + d0[2] = e0[2] + e1[2]; + d1[3] = v41_21*AA[0] - v40_20*AA[1]; + d1[2] = v40_20*AA[0] + v41_21*AA[1]; + + AA -= 8; + + d0 += 4; + d1 += 4; + e0 += 4; + e1 += 4; + } + } + + // step 3 + ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + + // optimized step 3: + + // the original step3 loop can be nested r inside s or s inside r; + // it's written originally as s inside r, but this is dumb when r + // iterates many times, and s few. So I have two copies of it and + // switch between them halfway. + + // this is iteration 0 of step 3 + imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A); + imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A); + + // this is iteration 1 of step 3 + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16); + imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16); + + l=2; + for (; l < (ld-3)>>1; ++l) { + int k0 = n >> (l+2), k0_2 = k0>>1; + int lim = 1 << (l+1); + int i; + for (i=0; i < lim; ++i) + imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3)); + } + + for (; l < ld-6; ++l) { + int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1; + int rlim = n >> (l+6), r; + int lim = 1 << (l+1); + int i_off; + float *A0 = A; + i_off = n2-1; + for (r=rlim; r > 0; --r) { + imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0); + A0 += k1*4; + i_off -= 8; + } + } + + // iterations with count: + // ld-6,-5,-4 all interleaved together + // the big win comes from getting rid of needless flops + // due to the constants on pass 5 & 4 being all 1 and 0; + // combining them to be simultaneous to improve cache made little difference + imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n); + + // output is u + + // step 4, 5, and 6 + // cannot be in-place because of step 5 + { + uint16 *bitrev = f->bit_reverse[blocktype]; + // weirdly, I'd have thought reading sequentially and writing + // erratically would have been better than vice-versa, but in + // fact that's not what my testing showed. (That is, with + // j = bitreverse(i), do you read i and write j, or read j and write i.) + + float *d0 = &v[n4-4]; + float *d1 = &v[n2-4]; + while (d0 >= v) { + int k4; + + k4 = bitrev[0]; + d1[3] = u[k4+0]; + d1[2] = u[k4+1]; + d0[3] = u[k4+2]; + d0[2] = u[k4+3]; + + k4 = bitrev[1]; + d1[1] = u[k4+0]; + d1[0] = u[k4+1]; + d0[1] = u[k4+2]; + d0[0] = u[k4+3]; + + d0 -= 4; + d1 -= 4; + bitrev += 2; + } + } + // (paper output is u, now v) + + + // data must be in buf2 + assert(v == buf2); + + // step 7 (paper output is v, now v) + // this is now in place + { + float *C = f->C[blocktype]; + float *d, *e; + + d = v; + e = v + n2 - 4; + + while (d < e) { + float a02,a11,b0,b1,b2,b3; + + a02 = d[0] - e[2]; + a11 = d[1] + e[3]; + + b0 = C[1]*a02 + C[0]*a11; + b1 = C[1]*a11 - C[0]*a02; + + b2 = d[0] + e[ 2]; + b3 = d[1] - e[ 3]; + + d[0] = b2 + b0; + d[1] = b3 + b1; + e[2] = b2 - b0; + e[3] = b1 - b3; + + a02 = d[2] - e[0]; + a11 = d[3] + e[1]; + + b0 = C[3]*a02 + C[2]*a11; + b1 = C[3]*a11 - C[2]*a02; + + b2 = d[2] + e[ 0]; + b3 = d[3] - e[ 1]; + + d[2] = b2 + b0; + d[3] = b3 + b1; + e[0] = b2 - b0; + e[1] = b1 - b3; + + C += 4; + d += 4; + e -= 4; + } + } + + // data must be in buf2 + + + // step 8+decode (paper output is X, now buffer) + // this generates pairs of data a la 8 and pushes them directly through + // the decode kernel (pushing rather than pulling) to avoid having + // to make another pass later + + // this cannot POSSIBLY be in place, so we refer to the buffers directly + + { + float *d0,*d1,*d2,*d3; + + float *B = f->B[blocktype] + n2 - 8; + float *e = buf2 + n2 - 8; + d0 = &buffer[0]; + d1 = &buffer[n2-4]; + d2 = &buffer[n2]; + d3 = &buffer[n-4]; + while (e >= v) { + float p0,p1,p2,p3; + + p3 = e[6]*B[7] - e[7]*B[6]; + p2 = -e[6]*B[6] - e[7]*B[7]; + + d0[0] = p3; + d1[3] = - p3; + d2[0] = p2; + d3[3] = p2; + + p1 = e[4]*B[5] - e[5]*B[4]; + p0 = -e[4]*B[4] - e[5]*B[5]; + + d0[1] = p1; + d1[2] = - p1; + d2[1] = p0; + d3[2] = p0; + + p3 = e[2]*B[3] - e[3]*B[2]; + p2 = -e[2]*B[2] - e[3]*B[3]; + + d0[2] = p3; + d1[1] = - p3; + d2[2] = p2; + d3[1] = p2; + + p1 = e[0]*B[1] - e[1]*B[0]; + p0 = -e[0]*B[0] - e[1]*B[1]; + + d0[3] = p1; + d1[0] = - p1; + d2[3] = p0; + d3[0] = p0; + + B -= 8; + e -= 8; + d0 += 4; + d2 += 4; + d1 -= 4; + d3 -= 4; + } + } + + temp_alloc_restore(f,save_point); +} + +#if 0 +// this is the original version of the above code, if you want to optimize it from scratch +void inverse_mdct_naive(float *buffer, int n) +{ + float s; + float A[1 << 12], B[1 << 12], C[1 << 11]; + int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; + int n3_4 = n - n4, ld; + // how can they claim this only uses N words?! + // oh, because they're only used sparsely, whoops + float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13]; + // set up twiddle factors + + for (k=k2=0; k < n4; ++k,k2+=2) { + A[k2 ] = (float) cos(4*k*M_PI/n); + A[k2+1] = (float) -sin(4*k*M_PI/n); + B[k2 ] = (float) cos((k2+1)*M_PI/n/2); + B[k2+1] = (float) sin((k2+1)*M_PI/n/2); + } + for (k=k2=0; k < n8; ++k,k2+=2) { + C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); + C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); + } + + // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" + // Note there are bugs in that pseudocode, presumably due to them attempting + // to rename the arrays nicely rather than representing the way their actual + // implementation bounces buffers back and forth. As a result, even in the + // "some formulars corrected" version, a direct implementation fails. These + // are noted below as "paper bug". + + // copy and reflect spectral data + for (k=0; k < n2; ++k) u[k] = buffer[k]; + for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1]; + // kernel from paper + // step 1 + for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) { + v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1]; + v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2]; + } + // step 2 + for (k=k4=0; k < n8; k+=1, k4+=4) { + w[n2+3+k4] = v[n2+3+k4] + v[k4+3]; + w[n2+1+k4] = v[n2+1+k4] + v[k4+1]; + w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4]; + w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4]; + } + // step 3 + ld = ilog(n) - 1; // ilog is off-by-one from normal definitions + for (l=0; l < ld-3; ++l) { + int k0 = n >> (l+2), k1 = 1 << (l+3); + int rlim = n >> (l+4), r4, r; + int s2lim = 1 << (l+2), s2; + for (r=r4=0; r < rlim; r4+=4,++r) { + for (s2=0; s2 < s2lim; s2+=2) { + u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4]; + u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4]; + u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1] + - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1]; + u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1] + + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1]; + } + } + if (l+1 < ld-3) { + // paper bug: ping-ponging of u&w here is omitted + memcpy(w, u, sizeof(u)); + } + } + + // step 4 + for (i=0; i < n8; ++i) { + int j = bit_reverse(i) >> (32-ld+3); + assert(j < n8); + if (i == j) { + // paper bug: original code probably swapped in place; if copying, + // need to directly copy in this case + int i8 = i << 3; + v[i8+1] = u[i8+1]; + v[i8+3] = u[i8+3]; + v[i8+5] = u[i8+5]; + v[i8+7] = u[i8+7]; + } else if (i < j) { + int i8 = i << 3, j8 = j << 3; + v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1]; + v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3]; + v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5]; + v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7]; + } + } + // step 5 + for (k=0; k < n2; ++k) { + w[k] = v[k*2+1]; + } + // step 6 + for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) { + u[n-1-k2] = w[k4]; + u[n-2-k2] = w[k4+1]; + u[n3_4 - 1 - k2] = w[k4+2]; + u[n3_4 - 2 - k2] = w[k4+3]; + } + // step 7 + for (k=k2=0; k < n8; ++k, k2 += 2) { + v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; + v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; + v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; + v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; + } + // step 8 + for (k=k2=0; k < n4; ++k,k2 += 2) { + X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1]; + X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ]; + } + + // decode kernel to output + // determined the following value experimentally + // (by first figuring out what made inverse_mdct_slow work); then matching that here + // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?) + s = 0.5; // theoretically would be n4 + + // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code, + // so it needs to use the "old" B values to behave correctly, or else + // set s to 1.0 ]]] + for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4]; + for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1]; + for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4]; +} +#endif + +static float *get_window(vorb *f, int len) +{ + len <<= 1; + if (len == f->blocksize_0) return f->window[0]; + if (len == f->blocksize_1) return f->window[1]; + assert(0); + return NULL; +} + +#ifndef STB_VORBIS_NO_DEFER_FLOOR +typedef int16 YTYPE; +#else +typedef int YTYPE; +#endif +static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag) +{ + int n2 = n >> 1; + int s = map->chan[i].mux, floor; + floor = map->submap_floor[s]; + if (f->floor_types[floor] == 0) { + return error(f, VORBIS_invalid_stream); + } else { + Floor1 *g = &f->floor_config[floor].floor1; + int j,q; + int lx = 0, ly = finalY[0] * g->floor1_multiplier; + for (q=1; q < g->values; ++q) { + j = g->sorted_order[q]; + #ifndef STB_VORBIS_NO_DEFER_FLOOR + if (finalY[j] >= 0) + #else + if (step2_flag[j]) + #endif + { + int hy = finalY[j] * g->floor1_multiplier; + int hx = g->Xlist[j]; + draw_line(target, lx,ly, hx,hy, n2); + lx = hx, ly = hy; + } + } + if (lx < n2) + // optimization of: draw_line(target, lx,ly, n,ly, n2); + for (j=lx; j < n2; ++j) + LINE_OP(target[j], inverse_db_table[ly]); + } + return TRUE; +} + +static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) +{ + Mode *m; + int i, n, prev, next, window_center; + f->channel_buffer_start = f->channel_buffer_end = 0; + + retry: + if (f->eof) return FALSE; + if (!maybe_start_packet(f)) + return FALSE; + // check packet type + if (get_bits(f,1) != 0) { + if (IS_PUSH_MODE(f)) + return error(f,VORBIS_bad_packet_type); + while (EOP != get8_packet(f)); + goto retry; + } + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + + i = get_bits(f, ilog(f->mode_count-1)); + if (i == EOP) return FALSE; + if (i >= f->mode_count) return FALSE; + *mode = i; + m = f->mode_config + i; + if (m->blockflag) { + n = f->blocksize_1; + prev = get_bits(f,1); + next = get_bits(f,1); + } else { + prev = next = 0; + n = f->blocksize_0; + } + +// WINDOWING + + window_center = n >> 1; + if (m->blockflag && !prev) { + *p_left_start = (n - f->blocksize_0) >> 2; + *p_left_end = (n + f->blocksize_0) >> 2; + } else { + *p_left_start = 0; + *p_left_end = window_center; + } + if (m->blockflag && !next) { + *p_right_start = (n*3 - f->blocksize_0) >> 2; + *p_right_end = (n*3 + f->blocksize_0) >> 2; + } else { + *p_right_start = window_center; + *p_right_end = n; + } + return TRUE; +} + +static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left) +{ + Mapping *map; + int i,j,k,n,n2; + int zero_channel[256]; + int really_zero_channel[256]; + int window_center; + +// WINDOWING + + n = f->blocksize[m->blockflag]; + window_center = n >> 1; + + map = &f->mapping[m->mapping]; + +// FLOORS + n2 = n >> 1; + + for (i=0; i < f->channels; ++i) { + int s = map->chan[i].mux, floor; + zero_channel[i] = FALSE; + floor = map->submap_floor[s]; + if (f->floor_types[floor] == 0) { + return error(f, VORBIS_invalid_stream); + } else { + Floor1 *g = &f->floor_config[floor].floor1; + if (get_bits(f, 1)) { + short *finalY; + uint8 step2_flag[256]; + static int range_list[4] = { 256, 128, 86, 64 }; + int range = range_list[g->floor1_multiplier-1]; + int offset = 2; + finalY = f->finalY[i]; + finalY[0] = get_bits(f, ilog(range)-1); + finalY[1] = get_bits(f, ilog(range)-1); + for (j=0; j < g->partitions; ++j) { + int pclass = g->partition_class_list[j]; + int cdim = g->class_dimensions[pclass]; + int cbits = g->class_subclasses[pclass]; + int csub = (1 << cbits)-1; + int cval = 0; + if (cbits) { + Codebook *c = f->codebooks + g->class_masterbooks[pclass]; + DECODE(cval,f,c); + } + for (k=0; k < cdim; ++k) { + int book = g->subclass_books[pclass][cval & csub]; + cval = cval >> cbits; + if (book >= 0) { + int temp; + Codebook *c = f->codebooks + book; + DECODE(temp,f,c); + finalY[offset++] = temp; + } else + finalY[offset++] = 0; + } + } + if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec + step2_flag[0] = step2_flag[1] = 1; + for (j=2; j < g->values; ++j) { + int low, high, pred, highroom, lowroom, room, val; + low = g->neighbors[j][0]; + high = g->neighbors[j][1]; + //neighbors(g->Xlist, j, &low, &high); + pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]); + val = finalY[j]; + highroom = range - pred; + lowroom = pred; + if (highroom < lowroom) + room = highroom * 2; + else + room = lowroom * 2; + if (val) { + step2_flag[low] = step2_flag[high] = 1; + step2_flag[j] = 1; + if (val >= room) + if (highroom > lowroom) + finalY[j] = val - lowroom + pred; + else + finalY[j] = pred - val + highroom - 1; + else + if (val & 1) + finalY[j] = pred - ((val+1)>>1); + else + finalY[j] = pred + (val>>1); + } else { + step2_flag[j] = 0; + finalY[j] = pred; + } + } + +#ifdef STB_VORBIS_NO_DEFER_FLOOR + do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag); +#else + // defer final floor computation until _after_ residue + for (j=0; j < g->values; ++j) { + if (!step2_flag[j]) + finalY[j] = -1; + } +#endif + } else { + error: + zero_channel[i] = TRUE; + } + // So we just defer everything else to later + + // at this point we've decoded the floor into buffer + } + } + // at this point we've decoded all floors + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + + // re-enable coupled channels if necessary + memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels); + for (i=0; i < map->coupling_steps; ++i) + if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) { + zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE; + } + +// RESIDUE DECODE + for (i=0; i < map->submaps; ++i) { + float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; + int r,t; + uint8 do_not_decode[256]; + int ch = 0; + for (j=0; j < f->channels; ++j) { + if (map->chan[j].mux == i) { + if (zero_channel[j]) { + do_not_decode[ch] = TRUE; + residue_buffers[ch] = NULL; + } else { + do_not_decode[ch] = FALSE; + residue_buffers[ch] = f->channel_buffers[j]; + } + ++ch; + } + } + r = map->submap_residue[i]; + t = f->residue_types[r]; + decode_residue(f, residue_buffers, ch, n2, r, do_not_decode); + } + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + +// INVERSE COUPLING + for (i = map->coupling_steps-1; i >= 0; --i) { + int n2 = n >> 1; + float *m = f->channel_buffers[map->chan[i].magnitude]; + float *a = f->channel_buffers[map->chan[i].angle ]; + for (j=0; j < n2; ++j) { + float a2,m2; + if (m[j] > 0) + if (a[j] > 0) + m2 = m[j], a2 = m[j] - a[j]; + else + a2 = m[j], m2 = m[j] + a[j]; + else + if (a[j] > 0) + m2 = m[j], a2 = m[j] + a[j]; + else + a2 = m[j], m2 = m[j] - a[j]; + m[j] = m2; + a[j] = a2; + } + } + + // finish decoding the floors +#ifndef STB_VORBIS_NO_DEFER_FLOOR + for (i=0; i < f->channels; ++i) { + if (really_zero_channel[i]) { + memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); + } else { + do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL); + } + } +#else + for (i=0; i < f->channels; ++i) { + if (really_zero_channel[i]) { + memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); + } else { + for (j=0; j < n2; ++j) + f->channel_buffers[i][j] *= f->floor_buffers[i][j]; + } + } +#endif + +// INVERSE MDCT + for (i=0; i < f->channels; ++i) + inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); + + // this shouldn't be necessary, unless we exited on an error + // and want to flush to get to the next packet + flush_packet(f); + + if (f->first_decode) { + // assume we start so first non-discarded sample is sample 0 + // this isn't to spec, but spec would require us to read ahead + // and decode the size of all current frames--could be done, + // but presumably it's not a commonly used feature + f->current_loc = -n2; // start of first frame is positioned for discard + // we might have to discard samples "from" the next frame too, + // if we're lapping a large block then a small at the start? + f->discard_samples_deferred = n - right_end; + f->current_loc_valid = TRUE; + f->first_decode = FALSE; + } else if (f->discard_samples_deferred) { + left_start += f->discard_samples_deferred; + *p_left = left_start; + f->discard_samples_deferred = 0; + } else if (f->previous_length == 0 && f->current_loc_valid) { + // we're recovering from a seek... that means we're going to discard + // the samples from this packet even though we know our position from + // the last page header, so we need to update the position based on + // the discarded samples here + // but wait, the code below is going to add this in itself even + // on a discard, so we don't need to do it here... + } + + // check if we have ogg information about the sample # for this packet + if (f->last_seg_which == f->end_seg_with_known_loc) { + // if we have a valid current loc, and this is final: + if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) { + uint32 current_end = f->known_loc_for_packet - (n-right_end); + // then let's infer the size of the (probably) short final frame + if (current_end < f->current_loc + right_end) { + if (current_end < f->current_loc) { + // negative truncation, that's impossible! + *len = 0; + } else { + *len = current_end - f->current_loc; + } + *len += left_start; + f->current_loc += *len; + return TRUE; + } + } + // otherwise, just set our sample loc + // guess that the ogg granule pos refers to the _middle_ of the + // last frame? + // set f->current_loc to the position of left_start + f->current_loc = f->known_loc_for_packet - (n2-left_start); + f->current_loc_valid = TRUE; + } + if (f->current_loc_valid) + f->current_loc += (right_start - left_start); + + if (f->alloc.alloc_buffer) + assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); + *len = right_end; // ignore samples after the window goes to 0 + return TRUE; +} + +static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right) +{ + int mode, left_end, right_end; + if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0; + return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left); +} + +static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right) +{ + int prev,i,j; + // we use right&left (the start of the right- and left-window sin()-regions) + // to determine how much to return, rather than inferring from the rules + // (same result, clearer code); 'left' indicates where our sin() window + // starts, therefore where the previous window's right edge starts, and + // therefore where to start mixing from the previous buffer. 'right' + // indicates where our sin() ending-window starts, therefore that's where + // we start saving, and where our returned-data ends. + + // mixin from previous window + if (f->previous_length) { + int i,j, n = f->previous_length; + float *w = get_window(f, n); + for (i=0; i < f->channels; ++i) { + for (j=0; j < n; ++j) + f->channel_buffers[i][left+j] = + f->channel_buffers[i][left+j]*w[ j] + + f->previous_window[i][ j]*w[n-1-j]; + } + } + + prev = f->previous_length; + + // last half of this data becomes previous window + f->previous_length = len - right; + + // @OPTIMIZE: could avoid this copy by double-buffering the + // output (flipping previous_window with channel_buffers), but + // then previous_window would have to be 2x as large, and + // channel_buffers couldn't be temp mem (although they're NOT + // currently temp mem, they could be (unless we want to level + // performance by spreading out the computation)) + for (i=0; i < f->channels; ++i) + for (j=0; right+j < len; ++j) + f->previous_window[i][j] = f->channel_buffers[i][right+j]; + + if (!prev) + // there was no previous packet, so this data isn't valid... + // this isn't entirely true, only the would-have-overlapped data + // isn't valid, but this seems to be what the spec requires + return 0; + + // truncate a short frame + if (len < right) right = len; + + f->samples_output += right-left; + + return right - left; +} + +static void vorbis_pump_first_frame(stb_vorbis *f) +{ + int len, right, left; + if (vorbis_decode_packet(f, &len, &left, &right)) + vorbis_finish_frame(f, len, left, right); +} + +#ifndef STB_VORBIS_NO_PUSHDATA_API +static int is_whole_packet_present(stb_vorbis *f, int end_page) +{ + // make sure that we have the packet available before continuing... + // this requires a full ogg parse, but we know we can fetch from f->stream + + // instead of coding this out explicitly, we could save the current read state, + // read the next packet with get8() until end-of-packet, check f->eof, then + // reset the state? but that would be slower, esp. since we'd have over 256 bytes + // of state to restore (primarily the page segment table) + + int s = f->next_seg, first = TRUE; + uint8 *p = f->stream; + + if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag + for (; s < f->segment_count; ++s) { + p += f->segments[s]; + if (f->segments[s] < 255) // stop at first short segment + break; + } + // either this continues, or it ends it... + if (end_page) + if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream); + if (s == f->segment_count) + s = -1; // set 'crosses page' flag + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + first = FALSE; + } + for (; s == -1;) { + uint8 *q; + int n; + + // check that we have the page header ready + if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data); + // validate the page + if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream); + if (p[4] != 0) return error(f, VORBIS_invalid_stream); + if (first) { // the first segment must NOT have 'continued_packet', later ones MUST + if (f->previous_length) + if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); + // if no previous length, we're resynching, so we can come in on a continued-packet, + // which we'll just drop + } else { + if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); + } + n = p[26]; // segment counts + q = p+27; // q points to segment table + p = q + n; // advance past header + // make sure we've read the segment table + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + for (s=0; s < n; ++s) { + p += q[s]; + if (q[s] < 255) + break; + } + if (end_page) + if (s < n-1) return error(f, VORBIS_invalid_stream); + if (s == f->segment_count) + s = -1; // set 'crosses page' flag + if (p > f->stream_end) return error(f, VORBIS_need_more_data); + first = FALSE; + } + return TRUE; +} +#endif // !STB_VORBIS_NO_PUSHDATA_API + +static int start_decoder(vorb *f) +{ + uint8 header[6], x,y; + int len,i,j,k, max_submaps = 0; + int longest_floorlist=0; + + // first page, first packet + + if (!start_page(f)) return FALSE; + // validate page flag + if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page); + if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page); + if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page); + // check for expected packet length + if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page); + if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page); + // read packet + // check packet header + if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page); + if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page); + // vorbis_version + if (get32(f) != 0) return error(f, VORBIS_invalid_first_page); + f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page); + if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels); + f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page); + get32(f); // bitrate_maximum + get32(f); // bitrate_nominal + get32(f); // bitrate_minimum + x = get8(f); + { int log0,log1; + log0 = x & 15; + log1 = x >> 4; + f->blocksize_0 = 1 << log0; + f->blocksize_1 = 1 << log1; + if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup); + if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup); + if (log0 > log1) return error(f, VORBIS_invalid_setup); + } + + // framing_flag + x = get8(f); + if (!(x & 1)) return error(f, VORBIS_invalid_first_page); + + // second packet! + if (!start_page(f)) return FALSE; + + if (!start_packet(f)) return FALSE; + do { + len = next_segment(f); + skip(f, len); + f->bytes_in_seg = 0; + } while (len); + + // third packet! + if (!start_packet(f)) return FALSE; + + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (IS_PUSH_MODE(f)) { + if (!is_whole_packet_present(f, TRUE)) { + // convert error in ogg header to write type + if (f->error == VORBIS_invalid_stream) + f->error = VORBIS_invalid_setup; + return FALSE; + } + } + #endif + + crc32_init(); // always init it, to avoid multithread race conditions + + if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup); + for (i=0; i < 6; ++i) header[i] = get8_packet(f); + if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup); + + // codebooks + + f->codebook_count = get_bits(f,8) + 1; + f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count); + if (f->codebooks == NULL) return error(f, VORBIS_outofmem); + memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count); + for (i=0; i < f->codebook_count; ++i) { + uint32 *values; + int ordered, sorted_count; + int total=0; + uint8 *lengths; + Codebook *c = f->codebooks+i; + x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup); + x = get_bits(f, 8); + c->dimensions = (get_bits(f, 8)<<8) + x; + x = get_bits(f, 8); + y = get_bits(f, 8); + c->entries = (get_bits(f, 8)<<16) + (y<<8) + x; + ordered = get_bits(f,1); + c->sparse = ordered ? 0 : get_bits(f,1); + + if (c->sparse) + lengths = (uint8 *) setup_temp_malloc(f, c->entries); + else + lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); + + if (!lengths) return error(f, VORBIS_outofmem); + + if (ordered) { + int current_entry = 0; + int current_length = get_bits(f,5) + 1; + while (current_entry < c->entries) { + int limit = c->entries - current_entry; + int n = get_bits(f, ilog(limit)); + if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); } + memset(lengths + current_entry, current_length, n); + current_entry += n; + ++current_length; + } + } else { + for (j=0; j < c->entries; ++j) { + int present = c->sparse ? get_bits(f,1) : 1; + if (present) { + lengths[j] = get_bits(f, 5) + 1; + ++total; + } else { + lengths[j] = NO_CODE; + } + } + } + + if (c->sparse && total >= c->entries >> 2) { + // convert sparse items to non-sparse! + if (c->entries > (int) f->setup_temp_memory_required) + f->setup_temp_memory_required = c->entries; + + c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); + memcpy(c->codeword_lengths, lengths, c->entries); + setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs! + lengths = c->codeword_lengths; + c->sparse = 0; + } + + // compute the size of the sorted tables + if (c->sparse) { + sorted_count = total; + //assert(total != 0); + } else { + sorted_count = 0; + #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH + for (j=0; j < c->entries; ++j) + if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE) + ++sorted_count; + #endif + } + + c->sorted_entries = sorted_count; + values = NULL; + + if (!c->sparse) { + c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries); + if (!c->codewords) return error(f, VORBIS_outofmem); + } else { + unsigned int size; + if (c->sorted_entries) { + c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries); + if (!c->codeword_lengths) return error(f, VORBIS_outofmem); + c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries); + if (!c->codewords) return error(f, VORBIS_outofmem); + values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries); + if (!values) return error(f, VORBIS_outofmem); + } + size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries; + if (size > f->setup_temp_memory_required) + f->setup_temp_memory_required = size; + } + + if (!compute_codewords(c, lengths, c->entries, values)) { + if (c->sparse) setup_temp_free(f, values, 0); + return error(f, VORBIS_invalid_setup); + } + + if (c->sorted_entries) { + // allocate an extra slot for sentinels + c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1)); + // allocate an extra slot at the front so that c->sorted_values[-1] is defined + // so that we can catch that case without an extra if + c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1)); + if (c->sorted_values) { ++c->sorted_values; c->sorted_values[-1] = -1; } + compute_sorted_huffman(c, lengths, values); + } + + if (c->sparse) { + setup_temp_free(f, values, sizeof(*values)*c->sorted_entries); + setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries); + setup_temp_free(f, lengths, c->entries); + c->codewords = NULL; + } + + compute_accelerated_huffman(c); + + c->lookup_type = get_bits(f, 4); + if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup); + if (c->lookup_type > 0) { + uint16 *mults; + c->minimum_value = float32_unpack(get_bits(f, 32)); + c->delta_value = float32_unpack(get_bits(f, 32)); + c->value_bits = get_bits(f, 4)+1; + c->sequence_p = get_bits(f,1); + if (c->lookup_type == 1) { + c->lookup_values = lookup1_values(c->entries, c->dimensions); + } else { + c->lookup_values = c->entries * c->dimensions; + } + mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values); + if (mults == NULL) return error(f, VORBIS_outofmem); + for (j=0; j < (int) c->lookup_values; ++j) { + int q = get_bits(f, c->value_bits); + if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); } + mults[j] = q; + } + +#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK + if (c->lookup_type == 1) { + int len, sparse = c->sparse; + // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop + if (sparse) { + if (c->sorted_entries == 0) goto skip; + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions); + } else + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions); + if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); } + len = sparse ? c->sorted_entries : c->entries; + for (j=0; j < len; ++j) { + int z = sparse ? c->sorted_values[j] : j, div=1; + for (k=0; k < c->dimensions; ++k) { + int off = (z / div) % c->lookup_values; + c->multiplicands[j*c->dimensions + k] = + #ifndef STB_VORBIS_CODEBOOK_FLOATS + mults[off]; + #else + mults[off]*c->delta_value + c->minimum_value; + // in this case (and this case only) we could pre-expand c->sequence_p, + // and throw away the decode logic for it; have to ALSO do + // it in the case below, but it can only be done if + // STB_VORBIS_CODEBOOK_FLOATS + // !STB_VORBIS_DIVIDES_IN_CODEBOOK + #endif + div *= c->lookup_values; + } + } + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + c->lookup_type = 2; + } + else +#endif + { + c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); + #ifndef STB_VORBIS_CODEBOOK_FLOATS + memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); + #else + for (j=0; j < (int) c->lookup_values; ++j) + c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; + setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); + #endif + } + skip:; + + #ifdef STB_VORBIS_CODEBOOK_FLOATS + if (c->lookup_type == 2 && c->sequence_p) { + for (j=1; j < (int) c->lookup_values; ++j) + c->multiplicands[j] = c->multiplicands[j-1]; + c->sequence_p = 0; + } + #endif + } + } + + // time domain transfers (notused) + + x = get_bits(f, 6) + 1; + for (i=0; i < x; ++i) { + uint32 z = get_bits(f, 16); + if (z != 0) return error(f, VORBIS_invalid_setup); + } + + // Floors + f->floor_count = get_bits(f, 6)+1; + f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config)); + for (i=0; i < f->floor_count; ++i) { + f->floor_types[i] = get_bits(f, 16); + if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup); + if (f->floor_types[i] == 0) { + Floor0 *g = &f->floor_config[i].floor0; + g->order = get_bits(f,8); + g->rate = get_bits(f,16); + g->bark_map_size = get_bits(f,16); + g->amplitude_bits = get_bits(f,6); + g->amplitude_offset = get_bits(f,8); + g->number_of_books = get_bits(f,4) + 1; + for (j=0; j < g->number_of_books; ++j) + g->book_list[j] = get_bits(f,8); + return error(f, VORBIS_feature_not_supported); + } else { + Point p[31*8+2]; + Floor1 *g = &f->floor_config[i].floor1; + int max_class = -1; + g->partitions = get_bits(f, 5); + for (j=0; j < g->partitions; ++j) { + g->partition_class_list[j] = get_bits(f, 4); + if (g->partition_class_list[j] > max_class) + max_class = g->partition_class_list[j]; + } + for (j=0; j <= max_class; ++j) { + g->class_dimensions[j] = get_bits(f, 3)+1; + g->class_subclasses[j] = get_bits(f, 2); + if (g->class_subclasses[j]) { + g->class_masterbooks[j] = get_bits(f, 8); + if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } + for (k=0; k < 1 << g->class_subclasses[j]; ++k) { + g->subclass_books[j][k] = get_bits(f,8)-1; + if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } + } + g->floor1_multiplier = get_bits(f,2)+1; + g->rangebits = get_bits(f,4); + g->Xlist[0] = 0; + g->Xlist[1] = 1 << g->rangebits; + g->values = 2; + for (j=0; j < g->partitions; ++j) { + int c = g->partition_class_list[j]; + for (k=0; k < g->class_dimensions[c]; ++k) { + g->Xlist[g->values] = get_bits(f, g->rangebits); + ++g->values; + } + } + // precompute the sorting + for (j=0; j < g->values; ++j) { + p[j].x = g->Xlist[j]; + p[j].y = j; + } + qsort(p, g->values, sizeof(p[0]), point_compare); + for (j=0; j < g->values; ++j) + g->sorted_order[j] = (uint8) p[j].y; + // precompute the neighbors + for (j=2; j < g->values; ++j) { + int low,hi; + neighbors(g->Xlist, j, &low,&hi); + g->neighbors[j][0] = low; + g->neighbors[j][1] = hi; + } + + if (g->values > longest_floorlist) + longest_floorlist = g->values; + } + } + + // Residue + f->residue_count = get_bits(f, 6)+1; + f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(*f->residue_config)); + for (i=0; i < f->residue_count; ++i) { + uint8 residue_cascade[64]; + Residue *r = f->residue_config+i; + f->residue_types[i] = get_bits(f, 16); + if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup); + r->begin = get_bits(f, 24); + r->end = get_bits(f, 24); + r->part_size = get_bits(f,24)+1; + r->classifications = get_bits(f,6)+1; + r->classbook = get_bits(f,8); + for (j=0; j < r->classifications; ++j) { + uint8 high_bits=0; + uint8 low_bits=get_bits(f,3); + if (get_bits(f,1)) + high_bits = get_bits(f,5); + residue_cascade[j] = high_bits*8 + low_bits; + } + r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications); + for (j=0; j < r->classifications; ++j) { + for (k=0; k < 8; ++k) { + if (residue_cascade[j] & (1 << k)) { + r->residue_books[j][k] = get_bits(f, 8); + if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); + } else { + r->residue_books[j][k] = -1; + } + } + } + // precompute the classifications[] array to avoid inner-loop mod/divide + // call it 'classdata' since we already have r->classifications + r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); + if (!r->classdata) return error(f, VORBIS_outofmem); + memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); + for (j=0; j < f->codebooks[r->classbook].entries; ++j) { + int classwords = f->codebooks[r->classbook].dimensions; + int temp = j; + r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords); + for (k=classwords-1; k >= 0; --k) { + r->classdata[j][k] = temp % r->classifications; + temp /= r->classifications; + } + } + } + + f->mapping_count = get_bits(f,6)+1; + f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping)); + for (i=0; i < f->mapping_count; ++i) { + Mapping *m = f->mapping + i; + int mapping_type = get_bits(f,16); + if (mapping_type != 0) return error(f, VORBIS_invalid_setup); + m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan)); + if (get_bits(f,1)) + m->submaps = get_bits(f,4); + else + m->submaps = 1; + if (m->submaps > max_submaps) + max_submaps = m->submaps; + if (get_bits(f,1)) { + m->coupling_steps = get_bits(f,8)+1; + for (k=0; k < m->coupling_steps; ++k) { + m->chan[k].magnitude = get_bits(f, ilog(f->channels)-1); + m->chan[k].angle = get_bits(f, ilog(f->channels)-1); + if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup); + if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup); + if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup); + } + } else + m->coupling_steps = 0; + + // reserved field + if (get_bits(f,2)) return error(f, VORBIS_invalid_setup); + if (m->submaps > 1) { + for (j=0; j < f->channels; ++j) { + m->chan[j].mux = get_bits(f, 4); + if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup); + } + } else + // @SPECIFICATION: this case is missing from the spec + for (j=0; j < f->channels; ++j) + m->chan[j].mux = 0; + + for (j=0; j < m->submaps; ++j) { + get_bits(f,8); // discard + m->submap_floor[j] = get_bits(f,8); + m->submap_residue[j] = get_bits(f,8); + if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup); + if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup); + } + } + + // Modes + f->mode_count = get_bits(f, 6)+1; + for (i=0; i < f->mode_count; ++i) { + Mode *m = f->mode_config+i; + m->blockflag = get_bits(f,1); + m->windowtype = get_bits(f,16); + m->transformtype = get_bits(f,16); + m->mapping = get_bits(f,8); + if (m->windowtype != 0) return error(f, VORBIS_invalid_setup); + if (m->transformtype != 0) return error(f, VORBIS_invalid_setup); + if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup); + } + + flush_packet(f); + + f->previous_length = 0; + + for (i=0; i < f->channels; ++i) { + f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1); + f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); + #endif + } + + if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE; + if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE; + f->blocksize[0] = f->blocksize_0; + f->blocksize[1] = f->blocksize_1; + +#ifdef STB_VORBIS_DIVIDE_TABLE + if (integer_divide_table[1][1]==0) + for (i=0; i < DIVTAB_NUMER; ++i) + for (j=1; j < DIVTAB_DENOM; ++j) + integer_divide_table[i][j] = i / j; +#endif + + // compute how much temporary memory is needed + + // 1. + { + uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1); + uint32 classify_mem; + int i,max_part_read=0; + for (i=0; i < f->residue_count; ++i) { + Residue *r = f->residue_config + i; + int n_read = r->end - r->begin; + int part_read = n_read / r->part_size; + if (part_read > max_part_read) + max_part_read = part_read; + } + #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE + classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *)); + #else + classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *)); + #endif + + f->temp_memory_required = classify_mem; + if (imdct_mem > f->temp_memory_required) + f->temp_memory_required = imdct_mem; + } + + f->first_decode = TRUE; + + if (f->alloc.alloc_buffer) { + assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes); + // check if there's enough temp memory so we don't error later + if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset) + return error(f, VORBIS_outofmem); + } + + f->first_audio_page_offset = stb_vorbis_get_file_offset(f); + + return TRUE; +} + +static void vorbis_deinit(stb_vorbis *p) +{ + int i,j; + for (i=0; i < p->residue_count; ++i) { + Residue *r = p->residue_config+i; + if (r->classdata) { + for (j=0; j < p->codebooks[r->classbook].entries; ++j) + setup_free(p, r->classdata[j]); + setup_free(p, r->classdata); + } + setup_free(p, r->residue_books); + } + + if (p->codebooks) { + for (i=0; i < p->codebook_count; ++i) { + Codebook *c = p->codebooks + i; + setup_free(p, c->codeword_lengths); + setup_free(p, c->multiplicands); + setup_free(p, c->codewords); + setup_free(p, c->sorted_codewords); + // c->sorted_values[-1] is the first entry in the array + setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL); + } + setup_free(p, p->codebooks); + } + setup_free(p, p->floor_config); + setup_free(p, p->residue_config); + for (i=0; i < p->mapping_count; ++i) + setup_free(p, p->mapping[i].chan); + setup_free(p, p->mapping); + for (i=0; i < p->channels; ++i) { + setup_free(p, p->channel_buffers[i]); + setup_free(p, p->previous_window[i]); + #ifdef STB_VORBIS_NO_DEFER_FLOOR + setup_free(p, p->floor_buffers[i]); + #endif + setup_free(p, p->finalY[i]); + } + for (i=0; i < 2; ++i) { + setup_free(p, p->A[i]); + setup_free(p, p->B[i]); + setup_free(p, p->C[i]); + setup_free(p, p->window[i]); + } + #ifndef STB_VORBIS_NO_STDIO + if (p->close_on_free) fclose(p->f); + #endif +} + +void stb_vorbis_close(stb_vorbis *p) +{ + if (p == NULL) return; + vorbis_deinit(p); + setup_free(p,p); +} + +static void vorbis_init(stb_vorbis *p, stb_vorbis_alloc *z) +{ + memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start + if (z) { + p->alloc = *z; + p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3; + p->temp_offset = p->alloc.alloc_buffer_length_in_bytes; + } + p->eof = 0; + p->error = VORBIS__no_error; + p->stream = NULL; + p->codebooks = NULL; + p->page_crc_tests = -1; + #ifndef STB_VORBIS_NO_STDIO + p->close_on_free = FALSE; + p->f = NULL; + #endif +} + +int stb_vorbis_get_sample_offset(stb_vorbis *f) +{ + if (f->current_loc_valid) + return f->current_loc; + else + return -1; +} + +stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f) +{ + stb_vorbis_info d; + d.channels = f->channels; + d.sample_rate = f->sample_rate; + d.setup_memory_required = f->setup_memory_required; + d.setup_temp_memory_required = f->setup_temp_memory_required; + d.temp_memory_required = f->temp_memory_required; + d.max_frame_size = f->blocksize_1 >> 1; + return d; +} + +int stb_vorbis_get_error(stb_vorbis *f) +{ + int e = f->error; + f->error = VORBIS__no_error; + return e; +} + +static stb_vorbis * vorbis_alloc(stb_vorbis *f) +{ + stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p)); + return p; +} + +#ifndef STB_VORBIS_NO_PUSHDATA_API + +void stb_vorbis_flush_pushdata(stb_vorbis *f) +{ + f->previous_length = 0; + f->page_crc_tests = 0; + f->discard_samples_deferred = 0; + f->current_loc_valid = FALSE; + f->first_decode = FALSE; + f->samples_output = 0; + f->channel_buffer_start = 0; + f->channel_buffer_end = 0; +} + +static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len) +{ + int i,n; + for (i=0; i < f->page_crc_tests; ++i) + f->scan[i].bytes_done = 0; + + // if we have room for more scans, search for them first, because + // they may cause us to stop early if their header is incomplete + if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) { + if (data_len < 4) return 0; + data_len -= 3; // need to look for 4-byte sequence, so don't miss + // one that straddles a boundary + for (i=0; i < data_len; ++i) { + if (data[i] == 0x4f) { + if (0==memcmp(data+i, ogg_page_header, 4)) { + int j,len; + uint32 crc; + // make sure we have the whole page header + if (i+26 >= data_len || i+27+data[i+26] >= data_len) { + // only read up to this page start, so hopefully we'll + // have the whole page header start next time + data_len = i; + break; + } + // ok, we have it all; compute the length of the page + len = 27 + data[i+26]; + for (j=0; j < data[i+26]; ++j) + len += data[i+27+j]; + // scan everything up to the embedded crc (which we must 0) + crc = 0; + for (j=0; j < 22; ++j) + crc = crc32_update(crc, data[i+j]); + // now process 4 0-bytes + for ( ; j < 26; ++j) + crc = crc32_update(crc, 0); + // len is the total number of bytes we need to scan + n = f->page_crc_tests++; + f->scan[n].bytes_left = len-j; + f->scan[n].crc_so_far = crc; + f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24); + // if the last frame on a page is continued to the next, then + // we can't recover the sample_loc immediately + if (data[i+27+data[i+26]-1] == 255) + f->scan[n].sample_loc = ~0; + else + f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24); + f->scan[n].bytes_done = i+j; + if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT) + break; + // keep going if we still have room for more + } + } + } + } + + for (i=0; i < f->page_crc_tests;) { + uint32 crc; + int j; + int n = f->scan[i].bytes_done; + int m = f->scan[i].bytes_left; + if (m > data_len - n) m = data_len - n; + // m is the bytes to scan in the current chunk + crc = f->scan[i].crc_so_far; + for (j=0; j < m; ++j) + crc = crc32_update(crc, data[n+j]); + f->scan[i].bytes_left -= m; + f->scan[i].crc_so_far = crc; + if (f->scan[i].bytes_left == 0) { + // does it match? + if (f->scan[i].crc_so_far == f->scan[i].goal_crc) { + // Houston, we have page + data_len = n+m; // consumption amount is wherever that scan ended + f->page_crc_tests = -1; // drop out of page scan mode + f->previous_length = 0; // decode-but-don't-output one frame + f->next_seg = -1; // start a new page + f->current_loc = f->scan[i].sample_loc; // set the current sample location + // to the amount we'd have decoded had we decoded this page + f->current_loc_valid = f->current_loc != ~0; + return data_len; + } + // delete entry + f->scan[i] = f->scan[--f->page_crc_tests]; + } else { + ++i; + } + } + + return data_len; +} + +// return value: number of bytes we used +int stb_vorbis_decode_frame_pushdata( + stb_vorbis *f, // the file we're decoding + uint8 *data, int data_len, // the memory available for decoding + int *channels, // place to write number of float * buffers + float ***output, // place to write float ** array of float * buffers + int *samples // place to write number of output samples + ) +{ + int i; + int len,right,left; + + if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + if (f->page_crc_tests >= 0) { + *samples = 0; + return vorbis_search_for_page_pushdata(f, data, data_len); + } + + f->stream = data; + f->stream_end = data + data_len; + f->error = VORBIS__no_error; + + // check that we have the entire packet in memory + if (!is_whole_packet_present(f, FALSE)) { + *samples = 0; + return 0; + } + + if (!vorbis_decode_packet(f, &len, &left, &right)) { + // save the actual error we encountered + enum STBVorbisError error = f->error; + if (error == VORBIS_bad_packet_type) { + // flush and resynch + f->error = VORBIS__no_error; + while (get8_packet(f) != EOP) + if (f->eof) break; + *samples = 0; + return f->stream - data; + } + if (error == VORBIS_continued_packet_flag_invalid) { + if (f->previous_length == 0) { + // we may be resynching, in which case it's ok to hit one + // of these; just discard the packet + f->error = VORBIS__no_error; + while (get8_packet(f) != EOP) + if (f->eof) break; + *samples = 0; + return f->stream - data; + } + } + // if we get an error while parsing, what to do? + // well, it DEFINITELY won't work to continue from where we are! + stb_vorbis_flush_pushdata(f); + // restore the error that actually made us bail + f->error = error; + *samples = 0; + return 1; + } + + // success! + len = vorbis_finish_frame(f, len, left, right); + for (i=0; i < f->channels; ++i) + f->outputs[i] = f->channel_buffers[i] + left; + + if (channels) *channels = f->channels; + *samples = len; + *output = f->outputs; + return f->stream - data; +} + +stb_vorbis *stb_vorbis_open_pushdata( + unsigned char *data, int data_len, // the memory available for decoding + int *data_used, // only defined if result is not NULL + int *error, stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.stream = data; + p.stream_end = data + data_len; + p.push_mode = TRUE; + if (!start_decoder(&p)) { + if (p.eof) + *error = VORBIS_need_more_data; + else + *error = p.error; + return NULL; + } + f = vorbis_alloc(&p); + if (f) { + *f = p; + *data_used = f->stream - data; + *error = 0; + return f; + } else { + vorbis_deinit(&p); + return NULL; + } +} +#endif // STB_VORBIS_NO_PUSHDATA_API + +unsigned int stb_vorbis_get_file_offset(stb_vorbis *f) +{ + #ifndef STB_VORBIS_NO_PUSHDATA_API + if (f->push_mode) return 0; + #endif + if (USE_MEMORY(f)) return f->stream - f->stream_start; + #ifndef STB_VORBIS_NO_STDIO + return ftell(f->f) - f->f_start; + #endif +} + +#ifndef STB_VORBIS_NO_PULLDATA_API +// +// DATA-PULLING API +// + +static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last) +{ + for(;;) { + int n; + if (f->eof) return 0; + n = get8(f); + if (n == 0x4f) { // page header + unsigned int retry_loc = stb_vorbis_get_file_offset(f); + int i; + // check if we're off the end of a file_section stream + if (retry_loc - 25 > f->stream_len) + return 0; + // check the rest of the header + for (i=1; i < 4; ++i) + if (get8(f) != ogg_page_header[i]) + break; + if (f->eof) return 0; + if (i == 4) { + uint8 header[27]; + uint32 i, crc, goal, len; + for (i=0; i < 4; ++i) + header[i] = ogg_page_header[i]; + for (; i < 27; ++i) + header[i] = get8(f); + if (f->eof) return 0; + if (header[4] != 0) goto invalid; + goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24); + for (i=22; i < 26; ++i) + header[i] = 0; + crc = 0; + for (i=0; i < 27; ++i) + crc = crc32_update(crc, header[i]); + len = 0; + for (i=0; i < header[26]; ++i) { + int s = get8(f); + crc = crc32_update(crc, s); + len += s; + } + if (len && f->eof) return 0; + for (i=0; i < len; ++i) + crc = crc32_update(crc, get8(f)); + // finished parsing probable page + if (crc == goal) { + // we could now check that it's either got the last + // page flag set, OR it's followed by the capture + // pattern, but I guess TECHNICALLY you could have + // a file with garbage between each ogg page and recover + // from it automatically? So even though that paranoia + // might decrease the chance of an invalid decode by + // another 2^32, not worth it since it would hose those + // invalid-but-useful files? + if (end) + *end = stb_vorbis_get_file_offset(f); + if (last) { + if (header[5] & 0x04) + *last = 1; + else + *last = 0; + } + set_file_offset(f, retry_loc-1); + return 1; + } + } + invalid: + // not a valid page, so rewind and look for next one + set_file_offset(f, retry_loc); + } + } +} + +// seek is implemented with 'interpolation search'--this is like +// binary search, but we use the data values to estimate the likely +// location of the data item (plus a bit of a bias so when the +// estimation is wrong we don't waste overly much time) + +#define SAMPLE_unknown 0xffffffff + + +// ogg vorbis, in its insane infinite wisdom, only provides +// information about the sample at the END of the page. +// therefore we COULD have the data we need in the current +// page, and not know it. we could just use the end location +// as our only knowledge for bounds, seek back, and eventually +// the binary search finds it. or we can try to be smart and +// not waste time trying to locate more pages. we try to be +// smart, since this data is already in memory anyway, so +// doing needless I/O would be crazy! +static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) +{ + uint8 header[27], lacing[255]; + uint8 packet_type[255]; + int num_packet, packet_start, previous =0; + int i,len; + uint32 samples; + + // record where the page starts + z->page_start = stb_vorbis_get_file_offset(f); + + // parse the header + getn(f, header, 27); + assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); + getn(f, lacing, header[26]); + + // determine the length of the payload + len = 0; + for (i=0; i < header[26]; ++i) + len += lacing[i]; + + // this implies where the page ends + z->page_end = z->page_start + 27 + header[26] + len; + + // read the last-decoded sample out of the data + z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 16); + + if (header[5] & 4) { + // if this is the last page, it's not possible to work + // backwards to figure out the first sample! whoops! fuck. + z->first_decoded_sample = SAMPLE_unknown; + set_file_offset(f, z->page_start); + return 1; + } + + // scan through the frames to determine the sample-count of each one... + // our goal is the sample # of the first fully-decoded sample on the + // page, which is the first decoded sample of the 2nd page + + num_packet=0; + + packet_start = ((header[5] & 1) == 0); + + for (i=0; i < header[26]; ++i) { + if (packet_start) { + uint8 n,b,m; + if (lacing[i] == 0) goto bail; // trying to read from zero-length packet + n = get8(f); + // if bottom bit is non-zero, we've got corruption + if (n & 1) goto bail; + n >>= 1; + b = ilog(f->mode_count-1); + m = n >> b; + n &= (1 << b)-1; + if (n >= f->mode_count) goto bail; + if (num_packet == 0 && f->mode_config[n].blockflag) + previous = (m & 1); + packet_type[num_packet++] = f->mode_config[n].blockflag; + skip(f, lacing[i]-1); + } else + skip(f, lacing[i]); + packet_start = (lacing[i] < 255); + } + + // now that we know the sizes of all the pages, we can start determining + // how much sample data there is. + + samples = 0; + + // for the last packet, we step by its whole length, because the definition + // is that we encoded the end sample loc of the 'last packet completed', + // where 'completed' refers to packets being split, and we are left to guess + // what 'end sample loc' means. we assume it means ignoring the fact that + // the last half of the data is useless without windowing against the next + // packet... (so it's not REALLY complete in that sense) + if (num_packet > 1) + samples += f->blocksize[packet_type[num_packet-1]]; + + for (i=num_packet-2; i >= 1; --i) { + // now, for this packet, how many samples do we have that + // do not overlap the following packet? + if (packet_type[i] == 1) + if (packet_type[i+1] == 1) + samples += f->blocksize_1 >> 1; + else + samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); + else + samples += f->blocksize_0 >> 1; + } + // now, at this point, we've rewound to the very beginning of the + // _second_ packet. if we entirely discard the first packet after + // a seek, this will be exactly the right sample number. HOWEVER! + // we can't as easily compute this number for the LAST page. The + // only way to get the sample offset of the LAST page is to use + // the end loc from the previous page. But what that returns us + // is _exactly_ the place where we get our first non-overlapped + // sample. (I think. Stupid spec for being ambiguous.) So for + // consistency it's better to do that here, too. However, that + // will then require us to NOT discard all of the first frame we + // decode, in some cases, which means an even weirder frame size + // and extra code. what a fucking pain. + + // we're going to discard the first packet if we + // start the seek here, so we don't care about it. (we could actually + // do better; if the first packet is long, and the previous packet + // is short, there's actually data in the first half of the first + // packet that doesn't need discarding... but not worth paying the + // effort of tracking that of that here and in the seeking logic) + // except crap, if we infer it from the _previous_ packet's end + // location, we DO need to use that definition... and we HAVE to + // infer the start loc of the LAST packet from the previous packet's + // end location. fuck you, ogg vorbis. + + z->first_decoded_sample = z->last_decoded_sample - samples; + + // restore file state to where we were + set_file_offset(f, z->page_start); + return 1; + + // restore file state to where we were + bail: + set_file_offset(f, z->page_start); + return 0; +} + +static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) +{ + int left_start, left_end, right_start, right_end, mode,i; + int frame=0; + uint32 frame_start; + int frames_to_skip, data_to_skip; + + // first_sample is the sample # of the first sample that doesn't + // overlap the previous page... note that this requires us to + // _partially_ discard the first packet! bleh. + set_file_offset(f, page_start); + + f->next_seg = -1; // force page resync + + frame_start = first_sample; + // frame start is where the previous packet's last decoded sample + // was, which corresponds to left_end... EXCEPT if the previous + // packet was long and this packet is short? Probably a bug here. + + + // now, we can start decoding frames... we'll only FAKE decode them, + // until we find the frame that contains our sample; then we'll rewind, + // and try again + for (;;) { + int start; + + if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) + return error(f, VORBIS_seek_failed); + + if (frame == 0) + start = left_end; + else + start = left_start; + + // the window starts at left_start; the last valid sample we generate + // before the next frame's window start is right_start-1 + if (target_sample < frame_start + right_start-start) + break; + + flush_packet(f); + if (f->eof) + return error(f, VORBIS_seek_failed); + + frame_start += right_start - start; + + ++frame; + } + + // ok, at this point, the sample we want is contained in frame #'frame' + + // to decode frame #'frame' normally, we have to decode the + // previous frame first... but if it's the FIRST frame of the page + // we can't. if it's the first frame, it means it falls in the part + // of the first frame that doesn't overlap either of the other frames. + // so, if we have to handle that case for the first frame, we might + // as well handle it for all of them, so: + if (target_sample > frame_start + (left_end - left_start)) { + // so what we want to do is go ahead and just immediately decode + // this frame, but then make it so the next get_frame_float() uses + // this already-decoded data? or do we want to go ahead and rewind, + // and leave a flag saying to skip the first N data? let's do that + frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) + data_to_skip = left_end - left_start; + } else { + // otherwise, we want to skip frames 0, 1, 2, ... frame-2 + // (which means frame-2+1 total frames) then decode frame-1, + // then leave frame pending + frames_to_skip = frame - 1; + assert(frames_to_skip >= 0); + data_to_skip = -1; + } + + set_file_offset(f, page_start); + f->next_seg = - 1; // force page resync + + for (i=0; i < frames_to_skip; ++i) { + maybe_start_packet(f); + flush_packet(f); + } + + if (data_to_skip >= 0) { + int i,j,n = f->blocksize_0 >> 1; + f->discard_samples_deferred = data_to_skip; + for (i=0; i < f->channels; ++i) + for (j=0; j < n; ++j) + f->previous_window[i][j] = 0; + f->previous_length = n; + frame_start += data_to_skip; + } else { + f->previous_length = 0; + vorbis_pump_first_frame(f); + } + + // at this point, the NEXT decoded frame will generate the desired sample + if (fine) { + // so if we're doing sample accurate streaming, we want to go ahead and decode it! + if (target_sample != frame_start) { + int n; + stb_vorbis_get_frame_float(f, &n, NULL); + assert(target_sample > frame_start); + assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); + f->channel_buffer_start += (target_sample - frame_start); + } + } + + return 0; +} + +static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) +{ + ProbedPage p[2],q; + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + // do we know the location of the last page? + if (f->p_last.page_start == 0) { + uint32 z = stb_vorbis_stream_length_in_samples(f); + if (z == 0) return error(f, VORBIS_cant_find_last_page); + } + + p[0] = f->p_first; + p[1] = f->p_last; + + if (sample_number >= f->p_last.last_decoded_sample) + sample_number = f->p_last.last_decoded_sample-1; + + if (sample_number < f->p_first.last_decoded_sample) { + vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); + return 0; + } else { + int attempts=0; + while (p[0].page_end < p[1].page_start) { + uint32 probe; + uint32 start_offset, end_offset; + uint32 start_sample, end_sample; + + // copy these into local variables so we can tweak them + // if any are unknown + start_offset = p[0].page_end; + end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] + start_sample = p[0].last_decoded_sample; + end_sample = p[1].last_decoded_sample; + + // currently there is no such tweaking logic needed/possible? + if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) + return error(f, VORBIS_seek_failed); + + // now we want to lerp between these for the target samples... + + // step 1: we need to bias towards the page start... + if (start_offset + 4000 < end_offset) + end_offset -= 4000; + + // now compute an interpolated search loc + probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); + + // next we need to bias towards binary search... + // code is a little wonky to allow for full 32-bit unsigned values + if (attempts >= 4) { + uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); + if (attempts >= 8) + probe = probe2; + else if (probe < probe2) + probe = probe + ((probe2 - probe) >> 1); + else + probe = probe2 + ((probe - probe2) >> 1); + } + ++attempts; + + set_file_offset(f, probe); + if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); + if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); + q.after_previous_page_start = probe; + + // it's possible we've just found the last page again + if (q.page_start == p[1].page_start) { + p[1] = q; + continue; + } + + if (sample_number < q.last_decoded_sample) + p[1] = q; + else + p[0] = q; + } + + if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { + vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); + return 0; + } + return error(f, VORBIS_seek_failed); + } +} + +int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) +{ + return vorbis_seek_base(f, sample_number, FALSE); +} + +int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) +{ + return vorbis_seek_base(f, sample_number, TRUE); +} + +void stb_vorbis_seek_start(stb_vorbis *f) +{ + if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; } + set_file_offset(f, f->first_audio_page_offset); + f->previous_length = 0; + f->first_decode = TRUE; + f->next_seg = -1; + vorbis_pump_first_frame(f); +} + +unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f) +{ + unsigned int restore_offset, previous_safe; + unsigned int end, last_page_loc; + + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + if (!f->total_samples) { + int last; + uint32 lo,hi; + char header[6]; + + // first, store the current decode position so we can restore it + restore_offset = stb_vorbis_get_file_offset(f); + + // now we want to seek back 64K from the end (the last page must + // be at most a little less than 64K, but let's allow a little slop) + if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset) + previous_safe = f->stream_len - 65536; + else + previous_safe = f->first_audio_page_offset; + + set_file_offset(f, previous_safe); + // previous_safe is now our candidate 'earliest known place that seeking + // to will lead to the final page' + + if (!vorbis_find_page(f, &end, (int unsigned *)&last)) { + // if we can't find a page, we're hosed! + f->error = VORBIS_cant_find_last_page; + f->total_samples = 0xffffffff; + goto done; + } + + // check if there are more pages + last_page_loc = stb_vorbis_get_file_offset(f); + + // stop when the last_page flag is set, not when we reach eof; + // this allows us to stop short of a 'file_section' end without + // explicitly checking the length of the section + while (!last) { + set_file_offset(f, end); + if (!vorbis_find_page(f, &end, (int unsigned *)&last)) { + // the last page we found didn't have the 'last page' flag + // set. whoops! + break; + } + previous_safe = last_page_loc+1; + last_page_loc = stb_vorbis_get_file_offset(f); + } + + set_file_offset(f, last_page_loc); + + // parse the header + getn(f, (unsigned char *)header, 6); + // extract the absolute granule position + lo = get32(f); + hi = get32(f); + if (lo == 0xffffffff && hi == 0xffffffff) { + f->error = VORBIS_cant_find_last_page; + f->total_samples = SAMPLE_unknown; + goto done; + } + if (hi) + lo = 0xfffffffe; // saturate + f->total_samples = lo; + + f->p_last.page_start = last_page_loc; + f->p_last.page_end = end; + f->p_last.last_decoded_sample = lo; + f->p_last.first_decoded_sample = SAMPLE_unknown; + f->p_last.after_previous_page_start = previous_safe; + + done: + set_file_offset(f, restore_offset); + } + return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples; +} + +float stb_vorbis_stream_length_in_seconds(stb_vorbis *f) +{ + return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate; +} + + + +int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output) +{ + int len, right,left,i; + if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); + + if (!vorbis_decode_packet(f, &len, &left, &right)) { + f->channel_buffer_start = f->channel_buffer_end = 0; + return 0; + } + + len = vorbis_finish_frame(f, len, left, right); + for (i=0; i < f->channels; ++i) + f->outputs[i] = f->channel_buffers[i] + left; + + f->channel_buffer_start = left; + f->channel_buffer_end = left+len; + + if (channels) *channels = f->channels; + if (output) *output = f->outputs; + return len; +} + +#ifndef STB_VORBIS_NO_STDIO + +stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc, unsigned int length) +{ + stb_vorbis *f, p; + vorbis_init(&p, alloc); + p.f = file; + p.f_start = ftell(file); + p.stream_len = length; + p.close_on_free = close_on_free; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + *f = p; + vorbis_pump_first_frame(f); + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc) +{ + unsigned int len, start; + start = ftell(file); + fseek(file, 0, SEEK_END); + len = ftell(file) - start; + fseek(file, start, SEEK_SET); + return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len); +} + +stb_vorbis * stb_vorbis_open_filename(char *filename, int *error, stb_vorbis_alloc *alloc) +{ + FILE *f = fopen(filename, "rb"); + if (f) + return stb_vorbis_open_file(f, TRUE, error, alloc); + if (error) *error = VORBIS_file_open_failure; + return NULL; +} +#endif // STB_VORBIS_NO_STDIO + +stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc) +{ + stb_vorbis *f, p; + if (data == NULL) return NULL; + vorbis_init(&p, alloc); + p.stream = data; + p.stream_end = data + len; + p.stream_start = p.stream; + p.stream_len = len; + p.push_mode = FALSE; + if (start_decoder(&p)) { + f = vorbis_alloc(&p); + if (f) { + *f = p; + vorbis_pump_first_frame(f); + return f; + } + } + if (error) *error = p.error; + vorbis_deinit(&p); + return NULL; +} + +#ifndef STB_VORBIS_NO_INTEGER_CONVERSION +#define PLAYBACK_MONO 1 +#define PLAYBACK_LEFT 2 +#define PLAYBACK_RIGHT 4 + +#define L (PLAYBACK_LEFT | PLAYBACK_MONO) +#define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO) +#define R (PLAYBACK_RIGHT | PLAYBACK_MONO) + +static int8 channel_position[7][6] = +{ + { 0 }, + { C }, + { L, R }, + { L, C, R }, + { L, R, L, R }, + { L, C, R, L, R }, + { L, C, R, L, R, C }, +}; + + +#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT + typedef union { + float f; + int i; + } float_conv; + typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4]; + #define FASTDEF(x) float_conv x + // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round + #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT)) + #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22)) + #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s)) + #define check_endianness() +#else + #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s)))) + #define check_endianness() + #define FASTDEF(x) +#endif + +static void copy_samples(short *dest, float *src, int len) +{ + int i; + check_endianness(); + for (i=0; i < len; ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + dest[i] = v; + } +} + +static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len) +{ + #define BUFFER_SIZE 32 + float buffer[BUFFER_SIZE]; + int i,j,o,n = BUFFER_SIZE; + check_endianness(); + for (o = 0; o < len; o += BUFFER_SIZE) { + memset(buffer, 0, sizeof(buffer)); + if (o + n > len) n = len - o; + for (j=0; j < num_c; ++j) { + if (channel_position[num_c][j] & mask) { + for (i=0; i < n; ++i) + buffer[i] += data[j][d_offset+o+i]; + } + } + for (i=0; i < n; ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + output[o+i] = v; + } + } +} + +//static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; +static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len) +{ + #define BUFFER_SIZE 32 + float buffer[BUFFER_SIZE]; + int i,j,o,n = BUFFER_SIZE >> 1; + // o is the offset in the source data + check_endianness(); + for (o = 0; o < len; o += BUFFER_SIZE >> 1) { + // o2 is the offset in the output data + int o2 = o << 1; + memset(buffer, 0, sizeof(buffer)); + if (o + n > len) n = len - o; + for (j=0; j < num_c; ++j) { + int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT); + if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) { + for (i=0; i < n; ++i) { + buffer[i*2+0] += data[j][d_offset+o+i]; + buffer[i*2+1] += data[j][d_offset+o+i]; + } + } else if (m == PLAYBACK_LEFT) { + for (i=0; i < n; ++i) { + buffer[i*2+0] += data[j][d_offset+o+i]; + } + } else if (m == PLAYBACK_RIGHT) { + for (i=0; i < n; ++i) { + buffer[i*2+1] += data[j][d_offset+o+i]; + } + } + } + for (i=0; i < (n<<1); ++i) { + FASTDEF(temp); + int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + output[o2+i] = v; + } + } +} + +static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples) +{ + int i; + if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { + static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; + for (i=0; i < buf_c; ++i) + compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples); + } else { + int limit = buf_c < data_c ? buf_c : data_c; + for (i=0; i < limit; ++i) + copy_samples(buffer[i]+b_offset, data[i], samples); + for ( ; i < buf_c; ++i) + memset(buffer[i]+b_offset, 0, sizeof(short) * samples); + } +} + +int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) +{ + float **output; + int len = stb_vorbis_get_frame_float(f, NULL, &output); + if (len > num_samples) len = num_samples; + if (len) + convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len); + return len; +} + +static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len) +{ + int i; + check_endianness(); + if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { + assert(buf_c == 2); + for (i=0; i < buf_c; ++i) + compute_stereo_samples(buffer, data_c, data, d_offset, len); + } else { + int limit = buf_c < data_c ? buf_c : data_c; + int j; + for (j=0; j < len; ++j) { + for (i=0; i < limit; ++i) { + FASTDEF(temp); + float f = data[i][d_offset+j]; + int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15); + if ((unsigned int) (v + 32768) > 65535) + v = v < 0 ? -32768 : 32767; + *buffer++ = v; + } + for ( ; i < buf_c; ++i) + *buffer++ = 0; + } + } +} + +int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts) +{ + float **output; + int len; + if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts); + len = stb_vorbis_get_frame_float(f, NULL, &output); + if (len) { + if (len*num_c > num_shorts) len = num_shorts / num_c; + convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len); + } + return len; +} + +int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts) +{ + float **outputs; + int len = num_shorts / channels; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + if (k) + convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k); + buffer += k*channels; + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return n; +} + +int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len) +{ + float **outputs; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + if (k) + convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k); + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return n; +} + +#ifndef STB_VORBIS_NO_STDIO +int stb_vorbis_decode_filename(char *filename, int *channels, int* sample_rate, short **output) +{ + int data_len, offset, total, limit, error; + short *data; + stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL); + if (v == NULL) return -1; + limit = v->channels * 4096; + *channels = v->channels; + *sample_rate = v->sample_rate; + offset = data_len = 0; + total = limit; + data = (short *) malloc(total * sizeof(*data)); + if (data == NULL) { + stb_vorbis_close(v); + return -2; + } + for (;;) { + int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); + if (n == 0) break; + data_len += n; + offset += n * v->channels; + if (offset + limit > total) { + short *data2; + total *= 2; + data2 = (short *) realloc(data, total * sizeof(*data)); + if (data2 == NULL) { + free(data); + stb_vorbis_close(v); + return -2; + } + data = data2; + } + } + *output = data; + return data_len; +} +#endif // NO_STDIO + +int stb_vorbis_decode_memory(uint8 *mem, int len, int *channels, int* sample_rate, short **output) +{ + int data_len, offset, total, limit, error; + short *data; + stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL); + if (v == NULL) return -1; + limit = v->channels * 4096; + *channels = v->channels; + *sample_rate = v->sample_rate; + offset = data_len = 0; + total = limit; + data = (short *) malloc(total * sizeof(*data)); + if (data == NULL) { + stb_vorbis_close(v); + return -2; + } + for (;;) { + int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); + if (n == 0) break; + data_len += n; + offset += n * v->channels; + if (offset + limit > total) { + short *data2; + total *= 2; + data2 = (short *) realloc(data, total * sizeof(*data)); + if (data2 == NULL) { + free(data); + stb_vorbis_close(v); + return -2; + } + data = data2; + } + } + *output = data; + return data_len; +} +#endif + +int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats) +{ + float **outputs; + int len = num_floats / channels; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < len) { + int i,j; + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= len) k = len - n; + for (j=0; j < k; ++j) { + for (i=0; i < z; ++i) + *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j]; + for ( ; i < channels; ++i) + *buffer++ = 0; + } + n += k; + f->channel_buffer_start += k; + if (n == len) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return n; +} + +int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples) +{ + float **outputs; + int n=0; + int z = f->channels; + if (z > channels) z = channels; + while (n < num_samples) { + int i; + int k = f->channel_buffer_end - f->channel_buffer_start; + if (n+k >= num_samples) k = num_samples - n; + if (k) { + for (i=0; i < z; ++i) + memcpy(buffer[i]+n, f->channel_buffers+f->channel_buffer_start, sizeof(float)*k); + for ( ; i < channels; ++i) + memset(buffer[i]+n, 0, sizeof(float) * k); + } + n += k; + f->channel_buffer_start += k; + if (n == num_samples) break; + if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; + } + return n; +} +#endif // STB_VORBIS_NO_PULLDATA_API + +#endif // STB_VORBIS_HEADER_ONLY