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gronod
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26
lib/codecs/CMakeLists.txt Normal file
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idf_component_register(
INCLUDE_DIRS . ./inc inc/alac inc/helix-aac inc/mad inc/resample16 inc/soxr inc/vorbis inc/opus
)
if (DEFINED AAC_DISABLE_SBR)
add_prebuilt_library(libhelix-aac lib/libhelix-aac.a )
else ()
add_prebuilt_library(libhelix-aac lib/libhelix-aac-sbr.a )
endif()
add_prebuilt_library(libmad lib/libmad.a)
add_prebuilt_library(libFLAC lib/libFLAC.a )
add_prebuilt_library(libvorbisidec lib/libvorbisidec.a )
add_prebuilt_library(libogg lib/libogg.a )
add_prebuilt_library(libalac lib/libalac.a )
add_prebuilt_library(libresample16 lib/libresample16.a )
add_prebuilt_library(libopus lib/libopus.a )
target_link_libraries(${COMPONENT_LIB} INTERFACE libmad)
target_link_libraries(${COMPONENT_LIB} INTERFACE libFLAC)
target_link_libraries(${COMPONENT_LIB} INTERFACE libhelix-aac)
target_link_libraries(${COMPONENT_LIB} INTERFACE libvorbisidec)
target_link_libraries(${COMPONENT_LIB} INTERFACE libogg)
target_link_libraries(${COMPONENT_LIB} INTERFACE libalac)
target_link_libraries(${COMPONENT_LIB} INTERFACE libresample16)
target_link_libraries(${COMPONENT_LIB} INTERFACE libopus)

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lib/codecs/component.mk Normal file
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#
# "main" pseudo-component makefile.
#
# (Uses default behaviour of compiling all source files in directory, adding 'include' to include path.)
COMPONENT_ADD_LDFLAGS=-l$(COMPONENT_NAME) \
$(COMPONENT_PATH)/lib/libmad.a \
$(COMPONENT_PATH)/lib/libFLAC.a \
$(COMPONENT_PATH)/lib/libhelix-aac.a \
$(COMPONENT_PATH)/lib/libvorbisidec.a \
$(COMPONENT_PATH)/lib/libogg.a \
$(COMPONENT_PATH)/lib/libalac.a \
$(COMPONENT_PATH)/lib/libresample16.a \
$(COMPONENT_PATH)/lib/libopusfile.a \
$(COMPONENT_PATH)/lib/libopus.a
#$(COMPONENT_PATH)/lib/libFLAC.a
#$(COMPONENT_PATH)/lib/libesp-flac.a
#$(COMPONENT_PATH)/lib/libsoxr.a
#$(COMPONENT_PATH)/lib/libfaad.a
#$(COMPONENT_PATH)/lib/libvorbisidec.a
#$(COMPONENT_PATH)/lib/libesp-opus.a
#$(COMPONENT_PATH)/lib/libogg.a
#$(COMPONENT_PATH)/lib/libesp-tremor.a
#$(COMPONENT_PATH)/lib/libesp-ogg-container.a

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lib/codecs/inc/FLAC/all.h Normal file
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/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2022 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - 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.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS 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 FOUNDATION 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.
*/
#ifndef FLAC__ALL_H
#define FLAC__ALL_H
#include "export.h"
#include "assert.h"
#include "callback.h"
#include "format.h"
#include "metadata.h"
#include "ordinals.h"
#include "stream_decoder.h"
#include "stream_encoder.h"
/** \mainpage
*
* \section intro Introduction
*
* This is the documentation for the FLAC C and C++ APIs. It is
* highly interconnected; this introduction should give you a top
* level idea of the structure and how to find the information you
* need. As a prerequisite you should have at least a basic
* knowledge of the FLAC format, documented
* <A HREF="https://xiph.org/flac/format.html">here</A>.
*
* \section c_api FLAC C API
*
* The FLAC C API is the interface to libFLAC, a set of structures
* describing the components of FLAC streams, and functions for
* encoding and decoding streams, as well as manipulating FLAC
* metadata in files. The public include files will be installed
* in your include area (for example /usr/include/FLAC/...).
*
* By writing a little code and linking against libFLAC, it is
* relatively easy to add FLAC support to another program. The
* library is licensed under <A HREF="https://xiph.org/flac/license.html">Xiph's BSD license</A>.
* Complete source code of libFLAC as well as the command-line
* encoder and plugins is available and is a useful source of
* examples.
*
* Aside from encoders and decoders, libFLAC provides a powerful
* metadata interface for manipulating metadata in FLAC files. It
* allows the user to add, delete, and modify FLAC metadata blocks
* and it can automatically take advantage of PADDING blocks to avoid
* rewriting the entire FLAC file when changing the size of the
* metadata.
*
* libFLAC usually only requires the standard C library and C math
* library. In particular, threading is not used so there is no
* dependency on a thread library. However, libFLAC does not use
* global variables and should be thread-safe.
*
* libFLAC also supports encoding to and decoding from Ogg FLAC.
* However the metadata editing interfaces currently have limited
* read-only support for Ogg FLAC files.
*
* \section cpp_api FLAC C++ API
*
* The FLAC C++ API is a set of classes that encapsulate the
* structures and functions in libFLAC. They provide slightly more
* functionality with respect to metadata but are otherwise
* equivalent. For the most part, they share the same usage as
* their counterparts in libFLAC, and the FLAC C API documentation
* can be used as a supplement. The public include files
* for the C++ API will be installed in your include area (for
* example /usr/include/FLAC++/...).
*
* libFLAC++ is also licensed under
* <A HREF="https://xiph.org/flac/license.html">Xiph's BSD license</A>.
*
* \section getting_started Getting Started
*
* A good starting point for learning the API is to browse through
* the <A HREF="modules.html">modules</A>. Modules are logical
* groupings of related functions or classes, which correspond roughly
* to header files or sections of header files. Each module includes a
* detailed description of the general usage of its functions or
* classes.
*
* From there you can go on to look at the documentation of
* individual functions. You can see different views of the individual
* functions through the links in top bar across this page.
*
* If you prefer a more hands-on approach, you can jump right to some
* <A HREF="https://xiph.org/flac/documentation_example_code.html">example code</A>.
*
* \section porting_guide Porting Guide
*
* Starting with FLAC 1.1.3 a \link porting Porting Guide \endlink
* has been introduced which gives detailed instructions on how to
* port your code to newer versions of FLAC.
*
* \section embedded_developers Embedded Developers
*
* libFLAC has grown larger over time as more functionality has been
* included, but much of it may be unnecessary for a particular embedded
* implementation. Unused parts may be pruned by some simple editing of
* src/libFLAC/Makefile.am. In general, the decoders, encoders, and
* metadata interface are all independent from each other.
*
* It is easiest to just describe the dependencies:
*
* - All modules depend on the \link flac_format Format \endlink module.
* - The decoders and encoders depend on the bitbuffer.
* - The decoder is independent of the encoder. The encoder uses the
* decoder because of the verify feature, but this can be removed if
* not needed.
* - Parts of the metadata interface require the stream decoder (but not
* the encoder).
* - Ogg support is selectable through the compile time macro
* \c FLAC__HAS_OGG.
*
* For example, if your application only requires the stream decoder, no
* encoder, and no metadata interface, you can remove the stream encoder
* and the metadata interface, which will greatly reduce the size of the
* library.
*
* Also, there are several places in the libFLAC code with comments marked
* with "OPT:" where a \#define can be changed to enable code that might be
* faster on a specific platform. Experimenting with these can yield faster
* binaries.
*/
/** \defgroup porting Porting Guide for New Versions
*
* This module describes differences in the library interfaces from
* version to version. It assists in the porting of code that uses
* the libraries to newer versions of FLAC.
*
* One simple facility for making porting easier that has been added
* in FLAC 1.1.3 is a set of \#defines in \c export.h of each
* library's includes (e.g. \c include/FLAC/export.h). The
* \#defines mirror the libraries'
* <A HREF="http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning">libtool version numbers</A>,
* e.g. in libFLAC there are \c FLAC_API_VERSION_CURRENT,
* \c FLAC_API_VERSION_REVISION, and \c FLAC_API_VERSION_AGE.
* These can be used to support multiple versions of an API during the
* transition phase, e.g.
*
* \code
* #if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
* legacy code
* #else
* new code
* #endif
* \endcode
*
* The source will work for multiple versions and the legacy code can
* easily be removed when the transition is complete.
*
* Another available symbol is FLAC_API_SUPPORTS_OGG_FLAC (defined in
* include/FLAC/export.h), which can be used to determine whether or not
* the library has been compiled with support for Ogg FLAC. This is
* simpler than trying to call an Ogg init function and catching the
* error.
*/
/** \defgroup porting_1_1_2_to_1_1_3 Porting from FLAC 1.1.2 to 1.1.3
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.1.2 to FLAC 1.1.3.
*
* The main change between the APIs in 1.1.2 and 1.1.3 is that they have
* been simplified. First, libOggFLAC has been merged into libFLAC and
* libOggFLAC++ has been merged into libFLAC++. Second, both the three
* decoding layers and three encoding layers have been merged into a
* single stream decoder and stream encoder. That is, the functionality
* of FLAC__SeekableStreamDecoder and FLAC__FileDecoder has been merged
* into FLAC__StreamDecoder, and FLAC__SeekableStreamEncoder and
* FLAC__FileEncoder into FLAC__StreamEncoder. Only the
* FLAC__StreamDecoder and FLAC__StreamEncoder remain. What this means
* is there is now a single API that can be used to encode or decode
* streams to/from native FLAC or Ogg FLAC and the single API can work
* on both seekable and non-seekable streams.
*
* Instead of creating an encoder or decoder of a certain layer, now the
* client will always create a FLAC__StreamEncoder or
* FLAC__StreamDecoder. The old layers are now differentiated by the
* initialization function. For example, for the decoder,
* FLAC__stream_decoder_init() has been replaced by
* FLAC__stream_decoder_init_stream(). This init function takes
* callbacks for the I/O, and the seeking callbacks are optional. This
* allows the client to use the same object for seekable and
* non-seekable streams. For decoding a FLAC file directly, the client
* can use FLAC__stream_decoder_init_file() and pass just a filename
* and fewer callbacks; most of the other callbacks are supplied
* internally. For situations where fopen()ing by filename is not
* possible (e.g. Unicode filenames on Windows) the client can instead
* open the file itself and supply the FILE* to
* FLAC__stream_decoder_init_FILE(). The init functions now returns a
* FLAC__StreamDecoderInitStatus instead of FLAC__StreamDecoderState.
* Since the callbacks and client data are now passed to the init
* function, the FLAC__stream_decoder_set_*_callback() functions and
* FLAC__stream_decoder_set_client_data() are no longer needed. The
* rest of the calls to the decoder are the same as before.
*
* There are counterpart init functions for Ogg FLAC, e.g.
* FLAC__stream_decoder_init_ogg_stream(). All the rest of the calls
* and callbacks are the same as for native FLAC.
*
* As an example, in FLAC 1.1.2 a seekable stream decoder would have
* been set up like so:
*
* \code
* FLAC__SeekableStreamDecoder *decoder = FLAC__seekable_stream_decoder_new();
* if(decoder == NULL) do_something;
* FLAC__seekable_stream_decoder_set_md5_checking(decoder, true);
* [... other settings ...]
* FLAC__seekable_stream_decoder_set_read_callback(decoder, my_read_callback);
* FLAC__seekable_stream_decoder_set_seek_callback(decoder, my_seek_callback);
* FLAC__seekable_stream_decoder_set_tell_callback(decoder, my_tell_callback);
* FLAC__seekable_stream_decoder_set_length_callback(decoder, my_length_callback);
* FLAC__seekable_stream_decoder_set_eof_callback(decoder, my_eof_callback);
* FLAC__seekable_stream_decoder_set_write_callback(decoder, my_write_callback);
* FLAC__seekable_stream_decoder_set_metadata_callback(decoder, my_metadata_callback);
* FLAC__seekable_stream_decoder_set_error_callback(decoder, my_error_callback);
* FLAC__seekable_stream_decoder_set_client_data(decoder, my_client_data);
* if(FLAC__seekable_stream_decoder_init(decoder) != FLAC__SEEKABLE_STREAM_DECODER_OK) do_something;
* \endcode
*
* In FLAC 1.1.3 it is like this:
*
* \code
* FLAC__StreamDecoder *decoder = FLAC__stream_decoder_new();
* if(decoder == NULL) do_something;
* FLAC__stream_decoder_set_md5_checking(decoder, true);
* [... other settings ...]
* if(FLAC__stream_decoder_init_stream(
* decoder,
* my_read_callback,
* my_seek_callback, // or NULL
* my_tell_callback, // or NULL
* my_length_callback, // or NULL
* my_eof_callback, // or NULL
* my_write_callback,
* my_metadata_callback, // or NULL
* my_error_callback,
* my_client_data
* ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
* \endcode
*
* or you could do;
*
* \code
* [...]
* FILE *file = fopen("somefile.flac","rb");
* if(file == NULL) do_somthing;
* if(FLAC__stream_decoder_init_FILE(
* decoder,
* file,
* my_write_callback,
* my_metadata_callback, // or NULL
* my_error_callback,
* my_client_data
* ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
* \endcode
*
* or just:
*
* \code
* [...]
* if(FLAC__stream_decoder_init_file(
* decoder,
* "somefile.flac",
* my_write_callback,
* my_metadata_callback, // or NULL
* my_error_callback,
* my_client_data
* ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
* \endcode
*
* Another small change to the decoder is in how it handles unparseable
* streams. Before, when the decoder found an unparseable stream
* (reserved for when the decoder encounters a stream from a future
* encoder that it can't parse), it changed the state to
* \c FLAC__STREAM_DECODER_UNPARSEABLE_STREAM. Now the decoder instead
* drops sync and calls the error callback with a new error code
* \c FLAC__STREAM_DECODER_ERROR_STATUS_UNPARSEABLE_STREAM. This is
* more robust. If your error callback does not discriminate on the the
* error state, your code does not need to be changed.
*
* The encoder now has a new setting:
* FLAC__stream_encoder_set_apodization(). This is for setting the
* method used to window the data before LPC analysis. You only need to
* add a call to this function if the default is not suitable. There
* are also two new convenience functions that may be useful:
* FLAC__metadata_object_cuesheet_calculate_cddb_id() and
* FLAC__metadata_get_cuesheet().
*
* The \a bytes parameter to FLAC__StreamDecoderReadCallback,
* FLAC__StreamEncoderReadCallback, and FLAC__StreamEncoderWriteCallback
* is now \c size_t instead of \c uint32_t.
*/
/** \defgroup porting_1_1_3_to_1_1_4 Porting from FLAC 1.1.3 to 1.1.4
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.1.3 to FLAC 1.1.4.
*
* There were no changes to any of the interfaces from 1.1.3 to 1.1.4.
* There was a slight change in the implementation of
* FLAC__stream_encoder_set_metadata(); the function now makes a copy
* of the \a metadata array of pointers so the client no longer needs
* to maintain it after the call. The objects themselves that are
* pointed to by the array are still not copied though and must be
* maintained until the call to FLAC__stream_encoder_finish().
*/
/** \defgroup porting_1_1_4_to_1_2_0 Porting from FLAC 1.1.4 to 1.2.0
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.1.4 to FLAC 1.2.0.
*
* There were only very minor changes to the interfaces from 1.1.4 to 1.2.0.
* In libFLAC, \c FLAC__format_sample_rate_is_subset() was added.
* In libFLAC++, \c FLAC::Decoder::Stream::get_decode_position() was added.
*
* Finally, value of the constant \c FLAC__FRAME_HEADER_RESERVED_LEN
* has changed to reflect the conversion of one of the reserved bits
* into active use. It used to be \c 2 and now is \c 1. However the
* FLAC frame header length has not changed, so to skip the proper
* number of bits, use \c FLAC__FRAME_HEADER_RESERVED_LEN +
* \c FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN
*/
/** \defgroup porting_1_3_4_to_1_4_0 Porting from FLAC 1.3.4 to 1.4.0
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.3.4 to FLAC 1.4.0.
*
* \section porting_1_3_4_to_1_4_0_summary Summary
*
* Between FLAC 1.3.4 and FLAC 1.4.0, there have four breaking changes
* - the function get_client_data_from_decoder has been renamed to
* FLAC__get_decoder_client_data
* - some data types in the FLAC__Frame struct have changed
* - all functions resizing metadata blocks now return the object
* untouched if memory allocation fails, whereas previously the
* handling varied and was more or less undefined
* - all functions accepting a filename now take UTF-8 encoded filenames
* on Windows instead of filenames in the current codepage
*
* Furthermore, there have been the following additions
* - the functions FLAC__stream_encoder_set_limit_min_bitrate,
* FLAC__stream_encoder_get_limit_min_bitrate,
* FLAC::encoder::file::set_limit_min_bitrate() and
* FLAC::encoder::file::get_limit_min_bitrate() have been added
* - Added FLAC__STREAM_DECODER_ERROR_STATUS_BAD_METADATA to the
* FLAC__StreamDecoderErrorStatus enum
*
* \section porting_1_3_4_to_1_4_0_breaking Breaking changes
*
* The function \b get_client_data_from_decoder was added in FLAC 1.3.3
* but did not follow the API naming convention and was not properly
* exported. The function is now renamed and properly integrated as
* FLAC__stream_decoder_get_client_data
*
* To accomodate encoding and decoding 32-bit int PCM, some data types
* in the \b FLAC__frame struct were changed. Specifically, warmup
* in both the FLAC__Subframe_Fixed struc and the FLAC__Subframe_LPC
* struct is changed from FLAC__int32 to FLAC__int64. Also, value
* in the FLAC__Subframe_Constant is changed from FLAC__int32 to
* FLAC__int64. Finally, in FLAC__Subframe_Verbatim struct data is
* changes from a FLAC__int32 array to a union containing a FLAC__int32
* array and a FLAC__int64 array. Also, a new member is added,
* data_type, which clarifies whether the FLAC__int32 or FLAC__int64
* array is in use.
*
* Furthermore, the following functions now return the object untouched
* if memory allocation fails, whereas previously the handling varied
* and was more or less undefined
*
* - FLAC__metadata_object_seektable_resize_points
* - FLAC__metadata_object_vorbiscomment_resize_comments
* - FLAC__metadata_object_cuesheet_track_resize_indices
* - FLAC__metadata_object_cuesheet_resize_tracks
*
* The last breaking change is that all API functions taking a filename
* as an argument now, on Windows, must be supplied with that filename
* in the UTF-8 character encoding instead of using the current code
* page. libFLAC internally translates these UTF-8 encoded filenames to
* an appropriate representation to use with _wfopen. On all other
* systems, filename is passed to fopen without any translation, as it
* in libFLAC 1.3.4 and earlier.
*
* \section porting_1_3_4_to_1_4_0_additions Additions
*
* To aid in creating properly streamable FLAC files, a set of functions
* was added to make it possible to enfore a minimum bitrate to files
* created through libFLAC's stream_encoder.h interface. With this
* function enabled the resulting FLAC files have a minimum bitrate of
* 1bit/sample independent of the number of channels, i.e. 48kbit/s for
* 48kHz. This can be beneficial for streaming, as very low bitrates for
* silent sections compressed with 'constant' subframes can result in a
* bitrate of 1kbit/s, creating problems with clients that aren't aware
* of this possibility and buffer too much data.
*
* Finally, FLAC__STREAM_DECODER_ERROR_STATUS_BAD_METADATA was added to
* the FLAC__StreamDecoderErrorStatus enum to signal that the decoder
* encountered unreadable metadata.
*
*/
/** \defgroup flac FLAC C API
*
* The FLAC C API is the interface to libFLAC, a set of structures
* describing the components of FLAC streams, and functions for
* encoding and decoding streams, as well as manipulating FLAC
* metadata in files.
*
* You should start with the format components as all other modules
* are dependent on it.
*/
#endif

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/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2022 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - 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.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS 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 FOUNDATION 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.
*/
#ifndef FLAC__ASSERT_H
#define FLAC__ASSERT_H
/* we need this since some compilers (like MSVC) leave assert()s on release code (and we don't want to use their ASSERT) */
#ifdef FUZZING_BUILD_MODE_UNSAFE_FOR_PRODUCTION
#define FLAC__ASSERT(x) if(!(x)) __builtin_abort();
#define FLAC__ASSERT_DECLARATION(x) x
#else
#ifndef NDEBUG
#include <assert.h>
#define FLAC__ASSERT(x) assert(x)
#define FLAC__ASSERT_DECLARATION(x) x
#else
#define FLAC__ASSERT(x)
#define FLAC__ASSERT_DECLARATION(x)
#endif
#endif
#endif

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/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2004-2009 Josh Coalson
* Copyright (C) 2011-2022 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - 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.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS 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 FOUNDATION 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.
*/
#ifndef FLAC__CALLBACK_H
#define FLAC__CALLBACK_H
#include "ordinals.h"
#include <stdlib.h> /* for size_t */
/** \file include/FLAC/callback.h
*
* \brief
* This module defines the structures for describing I/O callbacks
* to the other FLAC interfaces.
*
* See the detailed documentation for callbacks in the
* \link flac_callbacks callbacks \endlink module.
*/
/** \defgroup flac_callbacks FLAC/callback.h: I/O callback structures
* \ingroup flac
*
* \brief
* This module defines the structures for describing I/O callbacks
* to the other FLAC interfaces.
*
* The purpose of the I/O callback functions is to create a common way
* for the metadata interfaces to handle I/O.
*
* Originally the metadata interfaces required filenames as the way of
* specifying FLAC files to operate on. This is problematic in some
* environments so there is an additional option to specify a set of
* callbacks for doing I/O on the FLAC file, instead of the filename.
*
* In addition to the callbacks, a FLAC__IOHandle type is defined as an
* opaque structure for a data source.
*
* The callback function prototypes are similar (but not identical) to the
* stdio functions fread, fwrite, fseek, ftell, feof, and fclose. If you use
* stdio streams to implement the callbacks, you can pass fread, fwrite, and
* fclose anywhere a FLAC__IOCallback_Read, FLAC__IOCallback_Write, or
* FLAC__IOCallback_Close is required, and a FILE* anywhere a FLAC__IOHandle
* is required. \warning You generally CANNOT directly use fseek or ftell
* for FLAC__IOCallback_Seek or FLAC__IOCallback_Tell since on most systems
* these use 32-bit offsets and FLAC requires 64-bit offsets to deal with
* large files. You will have to find an equivalent function (e.g. ftello),
* or write a wrapper. The same is true for feof() since this is usually
* implemented as a macro, not as a function whose address can be taken.
*
* \{
*/
#ifdef __cplusplus
extern "C" {
#endif
/** This is the opaque handle type used by the callbacks. Typically
* this is a \c FILE* or address of a file descriptor.
*/
typedef void* FLAC__IOHandle;
/** Signature for the read callback.
* The signature and semantics match POSIX fread() implementations
* and can generally be used interchangeably.
*
* \param ptr The address of the read buffer.
* \param size The size of the records to be read.
* \param nmemb The number of records to be read.
* \param handle The handle to the data source.
* \retval size_t
* The number of records read.
*/
typedef size_t (*FLAC__IOCallback_Read) (void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
/** Signature for the write callback.
* The signature and semantics match POSIX fwrite() implementations
* and can generally be used interchangeably.
*
* \param ptr The address of the write buffer.
* \param size The size of the records to be written.
* \param nmemb The number of records to be written.
* \param handle The handle to the data source.
* \retval size_t
* The number of records written.
*/
typedef size_t (*FLAC__IOCallback_Write) (const void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
/** Signature for the seek callback.
* The signature and semantics mostly match POSIX fseek() WITH ONE IMPORTANT
* EXCEPTION: the offset is a 64-bit type whereas fseek() is generally 'long'
* and 32-bits wide.
*
* \param handle The handle to the data source.
* \param offset The new position, relative to \a whence
* \param whence \c SEEK_SET, \c SEEK_CUR, or \c SEEK_END
* \retval int
* \c 0 on success, \c -1 on error.
*/
typedef int (*FLAC__IOCallback_Seek) (FLAC__IOHandle handle, FLAC__int64 offset, int whence);
/** Signature for the tell callback.
* The signature and semantics mostly match POSIX ftell() WITH ONE IMPORTANT
* EXCEPTION: the offset is a 64-bit type whereas ftell() is generally 'long'
* and 32-bits wide.
*
* \param handle The handle to the data source.
* \retval FLAC__int64
* The current position on success, \c -1 on error.
*/
typedef FLAC__int64 (*FLAC__IOCallback_Tell) (FLAC__IOHandle handle);
/** Signature for the EOF callback.
* The signature and semantics mostly match POSIX feof() but WATCHOUT:
* on many systems, feof() is a macro, so in this case a wrapper function
* must be provided instead.
*
* \param handle The handle to the data source.
* \retval int
* \c 0 if not at end of file, nonzero if at end of file.
*/
typedef int (*FLAC__IOCallback_Eof) (FLAC__IOHandle handle);
/** Signature for the close callback.
* The signature and semantics match POSIX fclose() implementations
* and can generally be used interchangeably.
*
* \param handle The handle to the data source.
* \retval int
* \c 0 on success, \c EOF on error.
*/
typedef int (*FLAC__IOCallback_Close) (FLAC__IOHandle handle);
/** A structure for holding a set of callbacks.
* Each FLAC interface that requires a FLAC__IOCallbacks structure will
* describe which of the callbacks are required. The ones that are not
* required may be set to NULL.
*
* If the seek requirement for an interface is optional, you can signify that
* a data source is not seekable by setting the \a seek field to \c NULL.
*/
typedef struct {
FLAC__IOCallback_Read read; /**< See FLAC__IOCallbacks */
FLAC__IOCallback_Write write; /**< See FLAC__IOCallbacks */
FLAC__IOCallback_Seek seek; /**< See FLAC__IOCallbacks */
FLAC__IOCallback_Tell tell; /**< See FLAC__IOCallbacks */
FLAC__IOCallback_Eof eof; /**< See FLAC__IOCallbacks */
FLAC__IOCallback_Close close; /**< See FLAC__IOCallbacks */
} FLAC__IOCallbacks;
/* \} */
#ifdef __cplusplus
}
#endif
#endif

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/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2022 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - 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.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS 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 FOUNDATION 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.
*/
#ifndef FLAC__EXPORT_H
#define FLAC__EXPORT_H
/** \file include/FLAC/export.h
*
* \brief
* This module contains \#defines and symbols for exporting function
* calls, and providing version information and compiled-in features.
*
* See the \link flac_export export \endlink module.
*/
/** \defgroup flac_export FLAC/export.h: export symbols
* \ingroup flac
*
* \brief
* This module contains \#defines and symbols for exporting function
* calls, and providing version information and compiled-in features.
*
* If you are compiling for Windows (with Visual Studio or MinGW for
* example) and will link to the static library (libFLAC++.lib) you
* should define FLAC__NO_DLL in your project to make sure the symbols
* are exported properly.
*
* \{
*/
/** This \#define is used internally in libFLAC and its headers to make
* sure the correct symbols are exported when working with shared
* libraries. On Windows, this \#define is set to __declspec(dllexport)
* when compiling libFLAC into a library and to __declspec(dllimport)
* when the headers are used to link to that DLL. On non-Windows systems
* it is used to set symbol visibility.
*
* Because of this, the define FLAC__NO_DLL must be defined when linking
* to libFLAC statically or linking will fail.
*/
/* This has grown quite complicated. FLAC__NO_DLL is used by MSVC sln
* files and CMake, which build either static or shared. autotools can
* build static, shared or **both**. Therefore, DLL_EXPORT, which is set
* by libtool, must override FLAC__NO_DLL on building shared components
*/
#if defined(_WIN32)
#if defined(FLAC__NO_DLL) && !(defined(DLL_EXPORT))
#define FLAC_API
#else
#ifdef FLAC_API_EXPORTS
#define FLAC_API __declspec(dllexport)
#else
#define FLAC_API __declspec(dllimport)
#endif
#endif
#elif defined(FLAC__USE_VISIBILITY_ATTR)
#define FLAC_API __attribute__ ((visibility ("default")))
#else
#define FLAC_API
#endif
/** These \#defines will mirror the libtool-based library version number, see
* http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning
*/
#define FLAC_API_VERSION_CURRENT 12
#define FLAC_API_VERSION_REVISION 0 /**< see above */
#define FLAC_API_VERSION_AGE 0 /**< see above */
#ifdef __cplusplus
extern "C" {
#endif
/** \c 1 if the library has been compiled with support for Ogg FLAC, else \c 0. */
extern FLAC_API int FLAC_API_SUPPORTS_OGG_FLAC;
#ifdef __cplusplus
}
#endif
/* \} */
#endif

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/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2022 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - 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.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS 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 FOUNDATION 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.
*/
#ifndef FLAC__ORDINALS_H
#define FLAC__ORDINALS_H
/* This of course assumes C99 headers */
#include <stdint.h>
#include <stdbool.h>
typedef int8_t FLAC__int8;
typedef uint8_t FLAC__uint8;
typedef int16_t FLAC__int16;
typedef int32_t FLAC__int32;
typedef int64_t FLAC__int64;
typedef uint16_t FLAC__uint16;
typedef uint32_t FLAC__uint32;
typedef uint64_t FLAC__uint64;
typedef int FLAC__bool;
typedef FLAC__uint8 FLAC__byte;
#endif

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/*****************************************************************************
* alac_wrapper.h: ALAC coder wrapper
*
* (c) Philippe G. 2019, philippe_44@outlook.com
*
* This software is released under the MIT License.
* https://opensource.org/licenses/MIT
*
*/
#ifndef __ALAC_WRAPPER_H_
#define __ALAC_WRAPPER_H_
struct alac_codec_s;
#ifdef __cplusplus
extern "C" {
#endif
struct alac_codec_s *alac_create_decoder(int magic_cookie_size, unsigned char *magic_cookie,
unsigned char *sample_size, unsigned *sample_rate,
unsigned char *channels, unsigned int *block_size);
void alac_delete_decoder(struct alac_codec_s *codec);
bool alac_to_pcm(struct alac_codec_s *codec, unsigned char* input,
unsigned char *output, char channels, unsigned *out_frames);
#ifdef __cplusplus
}
#endif
#endif

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/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: faad.h,v 1.51 2007/11/01 12:33:29 menno Exp $
**/
/* warn people for update */
#pragma message("please update faad2 include filename and function names!")
/* Backwards compatible link */
#include "neaacdec.h"

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/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: neaacdec.h,v 1.13 2009/01/26 23:51:15 menno Exp $
**/
#ifndef __NEAACDEC_H__
#define __NEAACDEC_H__
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#if 1
/* MACROS FOR BACKWARDS COMPATIBILITY */
/* structs */
#define faacDecHandle NeAACDecHandle
#define faacDecConfiguration NeAACDecConfiguration
#define faacDecConfigurationPtr NeAACDecConfigurationPtr
#define faacDecFrameInfo NeAACDecFrameInfo
/* functions */
#define faacDecGetErrorMessage NeAACDecGetErrorMessage
#define faacDecSetConfiguration NeAACDecSetConfiguration
#define faacDecGetCurrentConfiguration NeAACDecGetCurrentConfiguration
#define faacDecInit NeAACDecInit
#define faacDecInit2 NeAACDecInit2
#define faacDecInitDRM NeAACDecInitDRM
#define faacDecPostSeekReset NeAACDecPostSeekReset
#define faacDecOpen NeAACDecOpen
#define faacDecClose NeAACDecClose
#define faacDecDecode NeAACDecDecode
#define AudioSpecificConfig NeAACDecAudioSpecificConfig
#endif
#ifdef _WIN32
#pragma pack(push, 8)
#ifndef NEAACDECAPI
#define NEAACDECAPI __cdecl
#endif
#else
#ifndef NEAACDECAPI
#define NEAACDECAPI
#endif
#endif
#define FAAD2_VERSION "2.7"
/* object types for AAC */
#define MAIN 1
#define LC 2
#define SSR 3
#define LTP 4
#define HE_AAC 5
#define ER_LC 17
#define ER_LTP 19
#define LD 23
#define DRM_ER_LC 27 /* special object type for DRM */
/* header types */
#define RAW 0
#define ADIF 1
#define ADTS 2
#define LATM 3
/* SBR signalling */
#define NO_SBR 0
#define SBR_UPSAMPLED 1
#define SBR_DOWNSAMPLED 2
#define NO_SBR_UPSAMPLED 3
/* library output formats */
#define FAAD_FMT_16BIT 1
#define FAAD_FMT_24BIT 2
#define FAAD_FMT_32BIT 3
#define FAAD_FMT_FLOAT 4
#define FAAD_FMT_FIXED FAAD_FMT_FLOAT
#define FAAD_FMT_DOUBLE 5
/* Capabilities */
#define LC_DEC_CAP (1<<0) /* Can decode LC */
#define MAIN_DEC_CAP (1<<1) /* Can decode MAIN */
#define LTP_DEC_CAP (1<<2) /* Can decode LTP */
#define LD_DEC_CAP (1<<3) /* Can decode LD */
#define ERROR_RESILIENCE_CAP (1<<4) /* Can decode ER */
#define FIXED_POINT_CAP (1<<5) /* Fixed point */
/* Channel definitions */
#define FRONT_CHANNEL_CENTER (1)
#define FRONT_CHANNEL_LEFT (2)
#define FRONT_CHANNEL_RIGHT (3)
#define SIDE_CHANNEL_LEFT (4)
#define SIDE_CHANNEL_RIGHT (5)
#define BACK_CHANNEL_LEFT (6)
#define BACK_CHANNEL_RIGHT (7)
#define BACK_CHANNEL_CENTER (8)
#define LFE_CHANNEL (9)
#define UNKNOWN_CHANNEL (0)
/* DRM channel definitions */
#define DRMCH_MONO 1
#define DRMCH_STEREO 2
#define DRMCH_SBR_MONO 3
#define DRMCH_SBR_STEREO 4
#define DRMCH_SBR_PS_STEREO 5
/* A decode call can eat up to FAAD_MIN_STREAMSIZE bytes per decoded channel,
so at least so much bytes per channel should be available in this stream */
#define FAAD_MIN_STREAMSIZE 768 /* 6144 bits/channel */
typedef void *NeAACDecHandle;
typedef struct mp4AudioSpecificConfig
{
/* Audio Specific Info */
unsigned char objectTypeIndex;
unsigned char samplingFrequencyIndex;
unsigned long samplingFrequency;
unsigned char channelsConfiguration;
/* GA Specific Info */
unsigned char frameLengthFlag;
unsigned char dependsOnCoreCoder;
unsigned short coreCoderDelay;
unsigned char extensionFlag;
unsigned char aacSectionDataResilienceFlag;
unsigned char aacScalefactorDataResilienceFlag;
unsigned char aacSpectralDataResilienceFlag;
unsigned char epConfig;
char sbr_present_flag;
char forceUpSampling;
char downSampledSBR;
} mp4AudioSpecificConfig;
typedef struct NeAACDecConfiguration
{
unsigned char defObjectType;
unsigned long defSampleRate;
unsigned char outputFormat;
unsigned char downMatrix;
unsigned char useOldADTSFormat;
unsigned char dontUpSampleImplicitSBR;
} NeAACDecConfiguration, *NeAACDecConfigurationPtr;
typedef struct NeAACDecFrameInfo
{
unsigned long bytesconsumed;
unsigned long samples;
unsigned char channels;
unsigned char error;
unsigned long samplerate;
/* SBR: 0: off, 1: on; upsample, 2: on; downsampled, 3: off; upsampled */
unsigned char sbr;
/* MPEG-4 ObjectType */
unsigned char object_type;
/* AAC header type; MP4 will be signalled as RAW also */
unsigned char header_type;
/* multichannel configuration */
unsigned char num_front_channels;
unsigned char num_side_channels;
unsigned char num_back_channels;
unsigned char num_lfe_channels;
unsigned char channel_position[64];
/* PS: 0: off, 1: on */
unsigned char ps;
} NeAACDecFrameInfo;
char* NEAACDECAPI NeAACDecGetErrorMessage(unsigned char errcode);
unsigned long NEAACDECAPI NeAACDecGetCapabilities(void);
NeAACDecHandle NEAACDECAPI NeAACDecOpen(void);
NeAACDecConfigurationPtr NEAACDECAPI NeAACDecGetCurrentConfiguration(NeAACDecHandle hDecoder);
unsigned char NEAACDECAPI NeAACDecSetConfiguration(NeAACDecHandle hDecoder,
NeAACDecConfigurationPtr config);
/* Init the library based on info from the AAC file (ADTS/ADIF) */
long NEAACDECAPI NeAACDecInit(NeAACDecHandle hDecoder,
unsigned char *buffer,
unsigned long buffer_size,
unsigned long *samplerate,
unsigned char *channels);
/* Init the library using a DecoderSpecificInfo */
char NEAACDECAPI NeAACDecInit2(NeAACDecHandle hDecoder,
unsigned char *pBuffer,
unsigned long SizeOfDecoderSpecificInfo,
unsigned long *samplerate,
unsigned char *channels);
/* Init the library for DRM */
char NEAACDECAPI NeAACDecInitDRM(NeAACDecHandle *hDecoder, unsigned long samplerate,
unsigned char channels);
void NEAACDECAPI NeAACDecPostSeekReset(NeAACDecHandle hDecoder, long frame);
void NEAACDECAPI NeAACDecClose(NeAACDecHandle hDecoder);
void* NEAACDECAPI NeAACDecDecode(NeAACDecHandle hDecoder,
NeAACDecFrameInfo *hInfo,
unsigned char *buffer,
unsigned long buffer_size);
void* NEAACDECAPI NeAACDecDecode2(NeAACDecHandle hDecoder,
NeAACDecFrameInfo *hInfo,
unsigned char *buffer,
unsigned long buffer_size,
void **sample_buffer,
unsigned long sample_buffer_size);
char NEAACDECAPI NeAACDecAudioSpecificConfig(unsigned char *pBuffer,
unsigned long buffer_size,
mp4AudioSpecificConfig *mp4ASC);
#ifdef _WIN32
#pragma pack(pop)
#endif
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif

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/* ***** BEGIN LICENSE BLOCK *****
* Source last modified: $Id: aacdec.h,v 1.8 2005/11/10 00:15:08 margotm Exp $
*
* Portions Copyright (c) 1995-2005 RealNetworks, Inc. All Rights Reserved.
*
* The contents of this file, and the files included with this file,
* are subject to the current version of the RealNetworks Public
* Source License (the "RPSL") available at
* http://www.helixcommunity.org/content/rpsl unless you have licensed
* the file under the current version of the RealNetworks Community
* Source License (the "RCSL") available at
* http://www.helixcommunity.org/content/rcsl, in which case the RCSL
* will apply. You may also obtain the license terms directly from
* RealNetworks. You may not use this file except in compliance with
* the RPSL or, if you have a valid RCSL with RealNetworks applicable
* to this file, the RCSL. Please see the applicable RPSL or RCSL for
* the rights, obligations and limitations governing use of the
* contents of the file.
*
* This file is part of the Helix DNA Technology. RealNetworks is the
* developer of the Original Code and owns the copyrights in the
* portions it created.
*
* This file, and the files included with this file, is distributed
* and made available on an 'AS IS' basis, WITHOUT WARRANTY OF ANY
* KIND, EITHER EXPRESS OR IMPLIED, AND REALNETWORKS HEREBY DISCLAIMS
* ALL SUCH WARRANTIES, INCLUDING WITHOUT LIMITATION, ANY WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE, QUIET
* ENJOYMENT OR NON-INFRINGEMENT.
*
* Technology Compatibility Kit Test Suite(s) Location:
* http://www.helixcommunity.org/content/tck
*
* Contributor(s):
*
* ***** END LICENSE BLOCK ***** */
/**************************************************************************************
* Fixed-point HE-AAC decoder
* Jon Recker (jrecker@real.com)
* February 2005
*
* aacdec.h - public C API for AAC decoder
**************************************************************************************/
#ifndef _AACDEC_H
#define _AACDEC_H
#if defined(_WIN32) && !defined(_WIN32_WCE)
#
#elif defined(_WIN32) && defined(_WIN32_WCE) && defined(ARM)
#
#elif defined(_WIN32) && defined(WINCE_EMULATOR)
#
#elif defined (__arm) && defined (__ARMCC_VERSION)
#
#elif defined(_SYMBIAN) && defined(__WINS__)
#
#elif defined(__GNUC__) && defined(__arm__)
#
#elif defined(__GNUC__) && defined(__i386__)
#
#elif defined(__GNUC__) && defined(__amd64__)
#
#elif defined(__GNUC__) && (defined(__powerpc__) || defined(__POWERPC__))
#
#elif defined(_OPENWAVE_SIMULATOR) || defined(_OPENWAVE_ARMULATOR)
#
#elif defined(_SOLARIS) && !defined(__GNUC__)
#
#elif defined(__XTENSA__)
#
#else
#error No platform defined. See valid options in aacdec.h
#endif
#ifndef USE_DEFAULT_STDLIB
#define USE_DEFAULT_STDLIB
#endif
#ifdef __cplusplus
extern "C" {
#endif
/* according to spec (13818-7 section 8.2.2, 14496-3 section 4.5.3)
* max size of input buffer =
* 6144 bits = 768 bytes per SCE or CCE-I
* 12288 bits = 1536 bytes per CPE
* 0 bits = 0 bytes per CCE-D (uses bits from the SCE/CPE/CCE-I it is coupled to)
*/
#ifndef AAC_MAX_NCHANS /* if max channels isn't set in makefile, */
#define AAC_MAX_NCHANS 2 /* set to default max number of channels */
#endif
#define AAC_MAX_NSAMPS 1024
#define AAC_MAINBUF_SIZE (768 * AAC_MAX_NCHANS)
#define AAC_NUM_PROFILES 3
#define AAC_PROFILE_MP 0
#define AAC_PROFILE_LC 1
#define AAC_PROFILE_SSR 2
/* define these to enable decoder features */
#if defined(HELIX_FEATURE_AUDIO_CODEC_AAC_SBR)
#define AAC_ENABLE_SBR
#endif // HELIX_FEATURE_AUDIO_CODEC_AAC_SBR.
#define AAC_ENABLE_MPEG4
enum {
ERR_AAC_NONE = 0,
ERR_AAC_INDATA_UNDERFLOW = -1,
ERR_AAC_NULL_POINTER = -2,
ERR_AAC_INVALID_ADTS_HEADER = -3,
ERR_AAC_INVALID_ADIF_HEADER = -4,
ERR_AAC_INVALID_FRAME = -5,
ERR_AAC_MPEG4_UNSUPPORTED = -6,
ERR_AAC_CHANNEL_MAP = -7,
ERR_AAC_SYNTAX_ELEMENT = -8,
ERR_AAC_DEQUANT = -9,
ERR_AAC_STEREO_PROCESS = -10,
ERR_AAC_PNS = -11,
ERR_AAC_SHORT_BLOCK_DEINT = -12,
ERR_AAC_TNS = -13,
ERR_AAC_IMDCT = -14,
ERR_AAC_NCHANS_TOO_HIGH = -15,
ERR_AAC_SBR_INIT = -16,
ERR_AAC_SBR_BITSTREAM = -17,
ERR_AAC_SBR_DATA = -18,
ERR_AAC_SBR_PCM_FORMAT = -19,
ERR_AAC_SBR_NCHANS_TOO_HIGH = -20,
ERR_AAC_SBR_SINGLERATE_UNSUPPORTED = -21,
ERR_AAC_RAWBLOCK_PARAMS = -22,
ERR_AAC_UNKNOWN = -9999
};
typedef struct _AACFrameInfo {
int bitRate;
int nChans;
int sampRateCore;
int sampRateOut;
int bitsPerSample;
int outputSamps;
int profile;
int tnsUsed;
int pnsUsed;
} AACFrameInfo;
typedef void *HAACDecoder;
/* public C API */
HAACDecoder AACInitDecoder(void);
HAACDecoder AACInitDecoderPre(void *ptr, int sz);
void AACFreeDecoder(HAACDecoder hAACDecoder);
int AACDecode(HAACDecoder hAACDecoder, unsigned char **inbuf, int *bytesLeft, short *outbuf);
int AACFindSyncWord(unsigned char *buf, int nBytes);
void AACGetLastFrameInfo(HAACDecoder hAACDecoder, AACFrameInfo *aacFrameInfo);
int AACSetRawBlockParams(HAACDecoder hAACDecoder, int copyLast, AACFrameInfo *aacFrameInfo);
int AACFlushCodec(HAACDecoder hAACDecoder);
#ifdef HELIX_CONFIG_AAC_GENERATE_TRIGTABS_FLOAT
int AACInitTrigtabsFloat(void);
void AACFreeTrigtabsFloat(void);
#endif
#ifdef __cplusplus
}
#endif
#endif /* _AACDEC_H */

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/*
* libmad - MPEG audio decoder library
* Copyright (C) 2000-2004 Underbit Technologies, Inc.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* If you would like to negotiate alternate licensing terms, you may do
* so by contacting: Underbit Technologies, Inc. <info@underbit.com>
*/
# ifdef __cplusplus
extern "C" {
# endif
# define FPM_DEFAULT
# define SIZEOF_INT 4
# define SIZEOF_LONG 4
# define SIZEOF_LONG_LONG 8
/* Id: version.h,v 1.26 2004/01/23 09:41:33 rob Exp */
# ifndef LIBMAD_VERSION_H
# define LIBMAD_VERSION_H
# define MAD_VERSION_MAJOR 0
# define MAD_VERSION_MINOR 15
# define MAD_VERSION_PATCH 1
# define MAD_VERSION_EXTRA " (beta)"
# define MAD_VERSION_STRINGIZE(str) #str
# define MAD_VERSION_STRING(num) MAD_VERSION_STRINGIZE(num)
# define MAD_VERSION MAD_VERSION_STRING(MAD_VERSION_MAJOR) "." \
MAD_VERSION_STRING(MAD_VERSION_MINOR) "." \
MAD_VERSION_STRING(MAD_VERSION_PATCH) \
MAD_VERSION_EXTRA
# define MAD_PUBLISHYEAR "2000-2004"
# define MAD_AUTHOR "Underbit Technologies, Inc."
# define MAD_EMAIL "info@underbit.com"
extern char const mad_version[];
extern char const mad_copyright[];
extern char const mad_author[];
extern char const mad_build[];
# endif
/* Id: fixed.h,v 1.38 2004/02/17 02:02:03 rob Exp */
# ifndef LIBMAD_FIXED_H
# define LIBMAD_FIXED_H
# if SIZEOF_INT >= 4
typedef signed int mad_fixed_t;
typedef signed int mad_fixed64hi_t;
typedef unsigned int mad_fixed64lo_t;
# else
typedef signed long mad_fixed_t;
typedef signed long mad_fixed64hi_t;
typedef unsigned long mad_fixed64lo_t;
# endif
# if defined(_MSC_VER)
# define mad_fixed64_t signed __int64
# elif 1 || defined(__GNUC__)
# define mad_fixed64_t signed long long
# endif
# if defined(FPM_FLOAT)
typedef double mad_sample_t;
# else
typedef mad_fixed_t mad_sample_t;
# endif
/*
* Fixed-point format: 0xABBBBBBB
* A == whole part (sign + 3 bits)
* B == fractional part (28 bits)
*
* Values are signed two's complement, so the effective range is:
* 0x80000000 to 0x7fffffff
* -8.0 to +7.9999999962747097015380859375
*
* The smallest representable value is:
* 0x00000001 == 0.0000000037252902984619140625 (i.e. about 3.725e-9)
*
* 28 bits of fractional accuracy represent about
* 8.6 digits of decimal accuracy.
*
* Fixed-point numbers can be added or subtracted as normal
* integers, but multiplication requires shifting the 64-bit result
* from 56 fractional bits back to 28 (and rounding.)
*
* Changing the definition of MAD_F_FRACBITS is only partially
* supported, and must be done with care.
*/
# define MAD_F_FRACBITS 28
# if MAD_F_FRACBITS == 28
# define MAD_F(x) ((mad_fixed_t) (x##L))
# else
# if MAD_F_FRACBITS < 28
# warning "MAD_F_FRACBITS < 28"
# define MAD_F(x) ((mad_fixed_t) \
(((x##L) + \
(1L << (28 - MAD_F_FRACBITS - 1))) >> \
(28 - MAD_F_FRACBITS)))
# elif MAD_F_FRACBITS > 28
# error "MAD_F_FRACBITS > 28 not currently supported"
# define MAD_F(x) ((mad_fixed_t) \
((x##L) << (MAD_F_FRACBITS - 28)))
# endif
# endif
# define MAD_F_MIN ((mad_fixed_t) -0x80000000L)
# define MAD_F_MAX ((mad_fixed_t) +0x7fffffffL)
# define MAD_F_ONE MAD_F(0x10000000)
# define mad_f_tofixed(x) ((mad_fixed_t) \
((x) * (double) (1L << MAD_F_FRACBITS) + 0.5))
# define mad_f_todouble(x) ((double) \
((x) / (double) (1L << MAD_F_FRACBITS)))
# define mad_f_intpart(x) ((x) >> MAD_F_FRACBITS)
# define mad_f_fracpart(x) ((x) & ((1L << MAD_F_FRACBITS) - 1))
/* (x should be positive) */
# define mad_f_fromint(x) ((x) << MAD_F_FRACBITS)
# define mad_f_add(x, y) ((x) + (y))
# define mad_f_sub(x, y) ((x) - (y))
# if defined(FPM_FLOAT)
# error "FPM_FLOAT not yet supported"
# undef MAD_F
# define MAD_F(x) mad_f_todouble(x)
# define mad_f_mul(x, y) ((x) * (y))
# define mad_f_scale64
# undef ASO_ZEROCHECK
# elif defined(FPM_64BIT)
/*
* This version should be the most accurate if 64-bit types are supported by
* the compiler, although it may not be the most efficient.
*/
# if defined(OPT_ACCURACY)
# define mad_f_mul(x, y) \
((mad_fixed_t) \
((((mad_fixed64_t) (x) * (y)) + \
(1L << (MAD_F_SCALEBITS - 1))) >> MAD_F_SCALEBITS))
# else
# define mad_f_mul(x, y) \
((mad_fixed_t) (((mad_fixed64_t) (x) * (y)) >> MAD_F_SCALEBITS))
# endif
# define MAD_F_SCALEBITS MAD_F_FRACBITS
/* --- Intel --------------------------------------------------------------- */
# elif defined(FPM_INTEL)
# if defined(_MSC_VER)
# pragma warning(push)
# pragma warning(disable: 4035) /* no return value */
static __forceinline
mad_fixed_t mad_f_mul_inline(mad_fixed_t x, mad_fixed_t y)
{
enum {
fracbits = MAD_F_FRACBITS
};
__asm {
mov eax, x
imul y
shrd eax, edx, fracbits
}
/* implicit return of eax */
}
# pragma warning(pop)
# define mad_f_mul mad_f_mul_inline
# define mad_f_scale64
# else
/*
* This Intel version is fast and accurate; the disposition of the least
* significant bit depends on OPT_ACCURACY via mad_f_scale64().
*/
# define MAD_F_MLX(hi, lo, x, y) \
asm ("imull %3" \
: "=a" (lo), "=d" (hi) \
: "%a" (x), "rm" (y) \
: "cc")
# if defined(OPT_ACCURACY)
/*
* This gives best accuracy but is not very fast.
*/
# define MAD_F_MLA(hi, lo, x, y) \
({ mad_fixed64hi_t __hi; \
mad_fixed64lo_t __lo; \
MAD_F_MLX(__hi, __lo, (x), (y)); \
asm ("addl %2,%0\n\t" \
"adcl %3,%1" \
: "=rm" (lo), "=rm" (hi) \
: "r" (__lo), "r" (__hi), "0" (lo), "1" (hi) \
: "cc"); \
})
# endif /* OPT_ACCURACY */
# if defined(OPT_ACCURACY)
/*
* Surprisingly, this is faster than SHRD followed by ADC.
*/
# define mad_f_scale64(hi, lo) \
({ mad_fixed64hi_t __hi_; \
mad_fixed64lo_t __lo_; \
mad_fixed_t __result; \
asm ("addl %4,%2\n\t" \
"adcl %5,%3" \
: "=rm" (__lo_), "=rm" (__hi_) \
: "0" (lo), "1" (hi), \
"ir" (1L << (MAD_F_SCALEBITS - 1)), "ir" (0) \
: "cc"); \
asm ("shrdl %3,%2,%1" \
: "=rm" (__result) \
: "0" (__lo_), "r" (__hi_), "I" (MAD_F_SCALEBITS) \
: "cc"); \
__result; \
})
# elif defined(OPT_INTEL)
/*
* Alternate Intel scaling that may or may not perform better.
*/
# define mad_f_scale64(hi, lo) \
({ mad_fixed_t __result; \
asm ("shrl %3,%1\n\t" \
"shll %4,%2\n\t" \
"orl %2,%1" \
: "=rm" (__result) \
: "0" (lo), "r" (hi), \
"I" (MAD_F_SCALEBITS), "I" (32 - MAD_F_SCALEBITS) \
: "cc"); \
__result; \
})
# else
# define mad_f_scale64(hi, lo) \
({ mad_fixed_t __result; \
asm ("shrdl %3,%2,%1" \
: "=rm" (__result) \
: "0" (lo), "r" (hi), "I" (MAD_F_SCALEBITS) \
: "cc"); \
__result; \
})
# endif /* OPT_ACCURACY */
# define MAD_F_SCALEBITS MAD_F_FRACBITS
# endif
/* --- ARM ----------------------------------------------------------------- */
# elif defined(FPM_ARM)
/*
* This ARM V4 version is as accurate as FPM_64BIT but much faster. The
* least significant bit is properly rounded at no CPU cycle cost!
*/
# if 1
/*
* This is faster than the default implementation via MAD_F_MLX() and
* mad_f_scale64().
*/
# define mad_f_mul(x, y) \
({ mad_fixed64hi_t __hi; \
mad_fixed64lo_t __lo; \
mad_fixed_t __result; \
asm ("smull %0, %1, %3, %4\n\t" \
"movs %0, %0, lsr %5\n\t" \
"adc %2, %0, %1, lsl %6" \
: "=&r" (__lo), "=&r" (__hi), "=r" (__result) \
: "%r" (x), "r" (y), \
"M" (MAD_F_SCALEBITS), "M" (32 - MAD_F_SCALEBITS) \
: "cc"); \
__result; \
})
# endif
# define MAD_F_MLX(hi, lo, x, y) \
asm ("smull %0, %1, %2, %3" \
: "=&r" (lo), "=&r" (hi) \
: "%r" (x), "r" (y))
# define MAD_F_MLA(hi, lo, x, y) \
asm ("smlal %0, %1, %2, %3" \
: "+r" (lo), "+r" (hi) \
: "%r" (x), "r" (y))
# define MAD_F_MLN(hi, lo) \
asm ("rsbs %0, %2, #0\n\t" \
"rsc %1, %3, #0" \
: "=r" (lo), "=r" (hi) \
: "0" (lo), "1" (hi) \
: "cc")
# define mad_f_scale64(hi, lo) \
({ mad_fixed_t __result; \
asm ("movs %0, %1, lsr %3\n\t" \
"adc %0, %0, %2, lsl %4" \
: "=&r" (__result) \
: "r" (lo), "r" (hi), \
"M" (MAD_F_SCALEBITS), "M" (32 - MAD_F_SCALEBITS) \
: "cc"); \
__result; \
})
# define MAD_F_SCALEBITS MAD_F_FRACBITS
/* --- MIPS ---------------------------------------------------------------- */
# elif defined(FPM_MIPS)
/*
* This MIPS version is fast and accurate; the disposition of the least
* significant bit depends on OPT_ACCURACY via mad_f_scale64().
*/
# define MAD_F_MLX(hi, lo, x, y) \
asm ("mult %2,%3" \
: "=l" (lo), "=h" (hi) \
: "%r" (x), "r" (y))
# if defined(HAVE_MADD_ASM)
# define MAD_F_MLA(hi, lo, x, y) \
asm ("madd %2,%3" \
: "+l" (lo), "+h" (hi) \
: "%r" (x), "r" (y))
# elif defined(HAVE_MADD16_ASM)
/*
* This loses significant accuracy due to the 16-bit integer limit in the
* multiply/accumulate instruction.
*/
# define MAD_F_ML0(hi, lo, x, y) \
asm ("mult %2,%3" \
: "=l" (lo), "=h" (hi) \
: "%r" ((x) >> 12), "r" ((y) >> 16))
# define MAD_F_MLA(hi, lo, x, y) \
asm ("madd16 %2,%3" \
: "+l" (lo), "+h" (hi) \
: "%r" ((x) >> 12), "r" ((y) >> 16))
# define MAD_F_MLZ(hi, lo) ((mad_fixed_t) (lo))
# endif
# if defined(OPT_SPEED)
# define mad_f_scale64(hi, lo) \
((mad_fixed_t) ((hi) << (32 - MAD_F_SCALEBITS)))
# define MAD_F_SCALEBITS MAD_F_FRACBITS
# endif
/* --- SPARC --------------------------------------------------------------- */
# elif defined(FPM_SPARC)
/*
* This SPARC V8 version is fast and accurate; the disposition of the least
* significant bit depends on OPT_ACCURACY via mad_f_scale64().
*/
# define MAD_F_MLX(hi, lo, x, y) \
asm ("smul %2, %3, %0\n\t" \
"rd %%y, %1" \
: "=r" (lo), "=r" (hi) \
: "%r" (x), "rI" (y))
/* --- PowerPC ------------------------------------------------------------- */
# elif defined(FPM_PPC)
/*
* This PowerPC version is fast and accurate; the disposition of the least
* significant bit depends on OPT_ACCURACY via mad_f_scale64().
*/
# define MAD_F_MLX(hi, lo, x, y) \
do { \
asm ("mullw %0,%1,%2" \
: "=r" (lo) \
: "%r" (x), "r" (y)); \
asm ("mulhw %0,%1,%2" \
: "=r" (hi) \
: "%r" (x), "r" (y)); \
} \
while (0)
# if defined(OPT_ACCURACY)
/*
* This gives best accuracy but is not very fast.
*/
# define MAD_F_MLA(hi, lo, x, y) \
({ mad_fixed64hi_t __hi; \
mad_fixed64lo_t __lo; \
MAD_F_MLX(__hi, __lo, (x), (y)); \
asm ("addc %0,%2,%3\n\t" \
"adde %1,%4,%5" \
: "=r" (lo), "=r" (hi) \
: "%r" (lo), "r" (__lo), \
"%r" (hi), "r" (__hi) \
: "xer"); \
})
# endif
# if defined(OPT_ACCURACY)
/*
* This is slower than the truncating version below it.
*/
# define mad_f_scale64(hi, lo) \
({ mad_fixed_t __result, __round; \
asm ("rotrwi %0,%1,%2" \
: "=r" (__result) \
: "r" (lo), "i" (MAD_F_SCALEBITS)); \
asm ("extrwi %0,%1,1,0" \
: "=r" (__round) \
: "r" (__result)); \
asm ("insrwi %0,%1,%2,0" \
: "+r" (__result) \
: "r" (hi), "i" (MAD_F_SCALEBITS)); \
asm ("add %0,%1,%2" \
: "=r" (__result) \
: "%r" (__result), "r" (__round)); \
__result; \
})
# else
# define mad_f_scale64(hi, lo) \
({ mad_fixed_t __result; \
asm ("rotrwi %0,%1,%2" \
: "=r" (__result) \
: "r" (lo), "i" (MAD_F_SCALEBITS)); \
asm ("insrwi %0,%1,%2,0" \
: "+r" (__result) \
: "r" (hi), "i" (MAD_F_SCALEBITS)); \
__result; \
})
# endif
# define MAD_F_SCALEBITS MAD_F_FRACBITS
/* --- Default ------------------------------------------------------------- */
# elif defined(FPM_DEFAULT)
/*
* This version is the most portable but it loses significant accuracy.
* Furthermore, accuracy is biased against the second argument, so care
* should be taken when ordering operands.
*
* The scale factors are constant as this is not used with SSO.
*
* Pre-rounding is required to stay within the limits of compliance.
*/
# if defined(OPT_SPEED)
# define mad_f_mul(x, y) (((x) >> 12) * ((y) >> 16))
# else
# define mad_f_mul(x, y) ((((x) + (1L << 11)) >> 12) * \
(((y) + (1L << 15)) >> 16))
# endif
/* ------------------------------------------------------------------------- */
# else
# error "no FPM selected"
# endif
/* default implementations */
# if !defined(mad_f_mul)
# define mad_f_mul(x, y) \
({ register mad_fixed64hi_t __hi; \
register mad_fixed64lo_t __lo; \
MAD_F_MLX(__hi, __lo, (x), (y)); \
mad_f_scale64(__hi, __lo); \
})
# endif
# if !defined(MAD_F_MLA)
# define MAD_F_ML0(hi, lo, x, y) ((lo) = mad_f_mul((x), (y)))
# define MAD_F_MLA(hi, lo, x, y) ((lo) += mad_f_mul((x), (y)))
# define MAD_F_MLN(hi, lo) ((lo) = -(lo))
# define MAD_F_MLZ(hi, lo) ((void) (hi), (mad_fixed_t) (lo))
# endif
# if !defined(MAD_F_ML0)
# define MAD_F_ML0(hi, lo, x, y) MAD_F_MLX((hi), (lo), (x), (y))
# endif
# if !defined(MAD_F_MLN)
# define MAD_F_MLN(hi, lo) ((hi) = ((lo) = -(lo)) ? ~(hi) : -(hi))
# endif
# if !defined(MAD_F_MLZ)
# define MAD_F_MLZ(hi, lo) mad_f_scale64((hi), (lo))
# endif
# if !defined(mad_f_scale64)
# if defined(OPT_ACCURACY)
# define mad_f_scale64(hi, lo) \
((((mad_fixed_t) \
(((hi) << (32 - (MAD_F_SCALEBITS - 1))) | \
((lo) >> (MAD_F_SCALEBITS - 1)))) + 1) >> 1)
# else
# define mad_f_scale64(hi, lo) \
((mad_fixed_t) \
(((hi) << (32 - MAD_F_SCALEBITS)) | \
((lo) >> MAD_F_SCALEBITS)))
# endif
# define MAD_F_SCALEBITS MAD_F_FRACBITS
# endif
/* C routines */
mad_fixed_t mad_f_abs(mad_fixed_t);
mad_fixed_t mad_f_div(mad_fixed_t, mad_fixed_t);
# endif
/* Id: bit.h,v 1.12 2004/01/23 09:41:32 rob Exp */
# ifndef LIBMAD_BIT_H
# define LIBMAD_BIT_H
struct mad_bitptr {
unsigned char const *byte;
unsigned short cache;
unsigned short left;
};
void mad_bit_init(struct mad_bitptr *, unsigned char const *);
# define mad_bit_finish(bitptr) /* nothing */
unsigned int mad_bit_length(struct mad_bitptr const *,
struct mad_bitptr const *);
# define mad_bit_bitsleft(bitptr) ((bitptr)->left)
unsigned char const *mad_bit_nextbyte(struct mad_bitptr const *);
void mad_bit_skip(struct mad_bitptr *, unsigned int);
unsigned long mad_bit_read(struct mad_bitptr *, unsigned int);
void mad_bit_write(struct mad_bitptr *, unsigned int, unsigned long);
unsigned short mad_bit_crc(struct mad_bitptr, unsigned int, unsigned short);
# endif
/* Id: timer.h,v 1.16 2004/01/23 09:41:33 rob Exp */
# ifndef LIBMAD_TIMER_H
# define LIBMAD_TIMER_H
typedef struct {
signed long seconds; /* whole seconds */
unsigned long fraction; /* 1/MAD_TIMER_RESOLUTION seconds */
} mad_timer_t;
extern mad_timer_t const mad_timer_zero;
# define MAD_TIMER_RESOLUTION 352800000UL
enum mad_units {
MAD_UNITS_HOURS = -2,
MAD_UNITS_MINUTES = -1,
MAD_UNITS_SECONDS = 0,
/* metric units */
MAD_UNITS_DECISECONDS = 10,
MAD_UNITS_CENTISECONDS = 100,
MAD_UNITS_MILLISECONDS = 1000,
/* audio sample units */
MAD_UNITS_8000_HZ = 8000,
MAD_UNITS_11025_HZ = 11025,
MAD_UNITS_12000_HZ = 12000,
MAD_UNITS_16000_HZ = 16000,
MAD_UNITS_22050_HZ = 22050,
MAD_UNITS_24000_HZ = 24000,
MAD_UNITS_32000_HZ = 32000,
MAD_UNITS_44100_HZ = 44100,
MAD_UNITS_48000_HZ = 48000,
/* video frame/field units */
MAD_UNITS_24_FPS = 24,
MAD_UNITS_25_FPS = 25,
MAD_UNITS_30_FPS = 30,
MAD_UNITS_48_FPS = 48,
MAD_UNITS_50_FPS = 50,
MAD_UNITS_60_FPS = 60,
/* CD audio frames */
MAD_UNITS_75_FPS = 75,
/* video drop-frame units */
MAD_UNITS_23_976_FPS = -24,
MAD_UNITS_24_975_FPS = -25,
MAD_UNITS_29_97_FPS = -30,
MAD_UNITS_47_952_FPS = -48,
MAD_UNITS_49_95_FPS = -50,
MAD_UNITS_59_94_FPS = -60
};
# define mad_timer_reset(timer) ((void) (*(timer) = mad_timer_zero))
int mad_timer_compare(mad_timer_t, mad_timer_t);
# define mad_timer_sign(timer) mad_timer_compare((timer), mad_timer_zero)
void mad_timer_negate(mad_timer_t *);
mad_timer_t mad_timer_abs(mad_timer_t);
void mad_timer_set(mad_timer_t *, unsigned long, unsigned long, unsigned long);
void mad_timer_add(mad_timer_t *, mad_timer_t);
void mad_timer_multiply(mad_timer_t *, signed long);
signed long mad_timer_count(mad_timer_t, enum mad_units);
unsigned long mad_timer_fraction(mad_timer_t, unsigned long);
void mad_timer_string(mad_timer_t, char *, char const *,
enum mad_units, enum mad_units, unsigned long);
# endif
/* Id: stream.h,v 1.20 2004/02/05 09:02:39 rob Exp */
# ifndef LIBMAD_STREAM_H
# define LIBMAD_STREAM_H
# define MAD_BUFFER_GUARD 8
# define MAD_BUFFER_MDLEN (511 + 2048 + MAD_BUFFER_GUARD)
enum mad_error {
MAD_ERROR_NONE = 0x0000, /* no error */
MAD_ERROR_BUFLEN = 0x0001, /* input buffer too small (or EOF) */
MAD_ERROR_BUFPTR = 0x0002, /* invalid (null) buffer pointer */
MAD_ERROR_NOMEM = 0x0031, /* not enough memory */
MAD_ERROR_LOSTSYNC = 0x0101, /* lost synchronization */
MAD_ERROR_BADLAYER = 0x0102, /* reserved header layer value */
MAD_ERROR_BADBITRATE = 0x0103, /* forbidden bitrate value */
MAD_ERROR_BADSAMPLERATE = 0x0104, /* reserved sample frequency value */
MAD_ERROR_BADEMPHASIS = 0x0105, /* reserved emphasis value */
MAD_ERROR_BADCRC = 0x0201, /* CRC check failed */
MAD_ERROR_BADBITALLOC = 0x0211, /* forbidden bit allocation value */
MAD_ERROR_BADSCALEFACTOR = 0x0221, /* bad scalefactor index */
MAD_ERROR_BADMODE = 0x0222, /* bad bitrate/mode combination */
MAD_ERROR_BADFRAMELEN = 0x0231, /* bad frame length */
MAD_ERROR_BADBIGVALUES = 0x0232, /* bad big_values count */
MAD_ERROR_BADBLOCKTYPE = 0x0233, /* reserved block_type */
MAD_ERROR_BADSCFSI = 0x0234, /* bad scalefactor selection info */
MAD_ERROR_BADDATAPTR = 0x0235, /* bad main_data_begin pointer */
MAD_ERROR_BADPART3LEN = 0x0236, /* bad audio data length */
MAD_ERROR_BADHUFFTABLE = 0x0237, /* bad Huffman table select */
MAD_ERROR_BADHUFFDATA = 0x0238, /* Huffman data overrun */
MAD_ERROR_BADSTEREO = 0x0239 /* incompatible block_type for JS */
};
# define MAD_RECOVERABLE(error) ((error) & 0xff00)
struct mad_stream {
unsigned char const *buffer; /* input bitstream buffer */
unsigned char const *bufend; /* end of buffer */
unsigned long skiplen; /* bytes to skip before next frame */
int sync; /* stream sync found */
unsigned long freerate; /* free bitrate (fixed) */
unsigned char const *this_frame; /* start of current frame */
unsigned char const *next_frame; /* start of next frame */
struct mad_bitptr ptr; /* current processing bit pointer */
struct mad_bitptr anc_ptr; /* ancillary bits pointer */
unsigned int anc_bitlen; /* number of ancillary bits */
unsigned char main_data[MAD_BUFFER_MDLEN];
/* Layer III main_data() */
unsigned int md_len; /* bytes in main_data */
int options; /* decoding options (see below) */
enum mad_error error; /* error code (see above) */
};
enum {
MAD_OPTION_IGNORECRC = 0x0001, /* ignore CRC errors */
MAD_OPTION_HALFSAMPLERATE = 0x0002 /* generate PCM at 1/2 sample rate */
# if 0 /* not yet implemented */
MAD_OPTION_LEFTCHANNEL = 0x0010, /* decode left channel only */
MAD_OPTION_RIGHTCHANNEL = 0x0020, /* decode right channel only */
MAD_OPTION_SINGLECHANNEL = 0x0030 /* combine channels */
# endif
};
void mad_stream_init(struct mad_stream *);
void mad_stream_finish(struct mad_stream *);
# define mad_stream_options(stream, opts) \
((void) ((stream)->options = (opts)))
void mad_stream_buffer(struct mad_stream *,
unsigned char const *, unsigned long);
void mad_stream_skip(struct mad_stream *, unsigned long);
int mad_stream_sync(struct mad_stream *);
char const *mad_stream_errorstr(struct mad_stream const *);
# endif
/* Id: frame.h,v 1.20 2004/01/23 09:41:32 rob Exp */
# ifndef LIBMAD_FRAME_H
# define LIBMAD_FRAME_H
enum mad_layer {
MAD_LAYER_I = 1, /* Layer I */
MAD_LAYER_II = 2, /* Layer II */
MAD_LAYER_III = 3 /* Layer III */
};
enum mad_mode {
MAD_MODE_SINGLE_CHANNEL = 0, /* single channel */
MAD_MODE_DUAL_CHANNEL = 1, /* dual channel */
MAD_MODE_JOINT_STEREO = 2, /* joint (MS/intensity) stereo */
MAD_MODE_STEREO = 3 /* normal LR stereo */
};
enum mad_emphasis {
MAD_EMPHASIS_NONE = 0, /* no emphasis */
MAD_EMPHASIS_50_15_US = 1, /* 50/15 microseconds emphasis */
MAD_EMPHASIS_CCITT_J_17 = 3, /* CCITT J.17 emphasis */
MAD_EMPHASIS_RESERVED = 2 /* unknown emphasis */
};
struct mad_header {
enum mad_layer layer; /* audio layer (1, 2, or 3) */
enum mad_mode mode; /* channel mode (see above) */
int mode_extension; /* additional mode info */
enum mad_emphasis emphasis; /* de-emphasis to use (see above) */
unsigned long bitrate; /* stream bitrate (bps) */
unsigned int samplerate; /* sampling frequency (Hz) */
unsigned short crc_check; /* frame CRC accumulator */
unsigned short crc_target; /* final target CRC checksum */
int flags; /* flags (see below) */
int private_bits; /* private bits (see below) */
mad_timer_t duration; /* audio playing time of frame */
};
struct mad_frame {
struct mad_header header; /* MPEG audio header */
int options; /* decoding options (from stream) */
mad_fixed_t sbsample[2][36][32]; /* synthesis subband filter samples */
mad_fixed_t overlap[2][32][18]; /* Layer III block overlap data */
mad_fixed_t xr[2][576];
mad_fixed_t tmp[32][3][6];
};
# define MAD_NCHANNELS(header) ((header)->mode ? 2 : 1)
# define MAD_NSBSAMPLES(header) \
((header)->layer == MAD_LAYER_I ? 12 : \
(((header)->layer == MAD_LAYER_III && \
((header)->flags & MAD_FLAG_LSF_EXT)) ? 18 : 36))
enum {
MAD_FLAG_NPRIVATE_III = 0x0007, /* number of Layer III private bits */
MAD_FLAG_INCOMPLETE = 0x0008, /* header but not data is decoded */
MAD_FLAG_PROTECTION = 0x0010, /* frame has CRC protection */
MAD_FLAG_COPYRIGHT = 0x0020, /* frame is copyright */
MAD_FLAG_ORIGINAL = 0x0040, /* frame is original (else copy) */
MAD_FLAG_PADDING = 0x0080, /* frame has additional slot */
MAD_FLAG_I_STEREO = 0x0100, /* uses intensity joint stereo */
MAD_FLAG_MS_STEREO = 0x0200, /* uses middle/side joint stereo */
MAD_FLAG_FREEFORMAT = 0x0400, /* uses free format bitrate */
MAD_FLAG_LSF_EXT = 0x1000, /* lower sampling freq. extension */
MAD_FLAG_MC_EXT = 0x2000, /* multichannel audio extension */
MAD_FLAG_MPEG_2_5_EXT = 0x4000 /* MPEG 2.5 (unofficial) extension */
};
enum {
MAD_PRIVATE_HEADER = 0x0100, /* header private bit */
MAD_PRIVATE_III = 0x001f /* Layer III private bits (up to 5) */
};
void mad_header_init(struct mad_header *);
# define mad_header_finish(header) /* nothing */
int mad_header_decode(struct mad_header *, struct mad_stream *);
void mad_frame_init(struct mad_frame *);
void mad_frame_finish(struct mad_frame *);
int mad_frame_decode(struct mad_frame *, struct mad_stream *);
void mad_frame_mute(struct mad_frame *);
# endif
/* Id: synth.h,v 1.15 2004/01/23 09:41:33 rob Exp */
# ifndef LIBMAD_SYNTH_H
# define LIBMAD_SYNTH_H
struct mad_pcm {
unsigned int samplerate; /* sampling frequency (Hz) */
unsigned short channels; /* number of channels */
unsigned short length; /* number of samples per channel */
mad_fixed_t samples[2][1152]; /* PCM output samples [ch][sample] */
};
struct mad_synth {
mad_fixed_t filter[2][2][2][16][8]; /* polyphase filterbank outputs */
/* [ch][eo][peo][s][v] */
unsigned int phase; /* current processing phase */
struct mad_pcm pcm; /* PCM output */
};
/* single channel PCM selector */
enum {
MAD_PCM_CHANNEL_SINGLE = 0
};
/* dual channel PCM selector */
enum {
MAD_PCM_CHANNEL_DUAL_1 = 0,
MAD_PCM_CHANNEL_DUAL_2 = 1
};
/* stereo PCM selector */
enum {
MAD_PCM_CHANNEL_STEREO_LEFT = 0,
MAD_PCM_CHANNEL_STEREO_RIGHT = 1
};
void mad_synth_init(struct mad_synth *);
# define mad_synth_finish(synth) /* nothing */
void mad_synth_mute(struct mad_synth *);
void mad_synth_frame(struct mad_synth *, struct mad_frame const *);
# endif
/* Id: decoder.h,v 1.17 2004/01/23 09:41:32 rob Exp */
# ifndef LIBMAD_DECODER_H
# define LIBMAD_DECODER_H
enum mad_decoder_mode {
MAD_DECODER_MODE_SYNC = 0,
MAD_DECODER_MODE_ASYNC
};
enum mad_flow {
MAD_FLOW_CONTINUE = 0x0000, /* continue normally */
MAD_FLOW_STOP = 0x0010, /* stop decoding normally */
MAD_FLOW_BREAK = 0x0011, /* stop decoding and signal an error */
MAD_FLOW_IGNORE = 0x0020 /* ignore the current frame */
};
struct mad_decoder {
enum mad_decoder_mode mode;
int options;
struct {
long pid;
int in;
int out;
} async;
struct {
struct mad_stream stream;
struct mad_frame frame;
struct mad_synth synth;
} *sync;
void *cb_data;
enum mad_flow (*input_func)(void *, struct mad_stream *);
enum mad_flow (*header_func)(void *, struct mad_header const *);
enum mad_flow (*filter_func)(void *,
struct mad_stream const *, struct mad_frame *);
enum mad_flow (*output_func)(void *,
struct mad_header const *, struct mad_pcm *);
enum mad_flow (*error_func)(void *, struct mad_stream *, struct mad_frame *);
enum mad_flow (*message_func)(void *, void *, unsigned int *);
};
void mad_decoder_init(struct mad_decoder *, void *,
enum mad_flow (*)(void *, struct mad_stream *),
enum mad_flow (*)(void *, struct mad_header const *),
enum mad_flow (*)(void *,
struct mad_stream const *,
struct mad_frame *),
enum mad_flow (*)(void *,
struct mad_header const *,
struct mad_pcm *),
enum mad_flow (*)(void *,
struct mad_stream *,
struct mad_frame *),
enum mad_flow (*)(void *, void *, unsigned int *));
int mad_decoder_finish(struct mad_decoder *);
# define mad_decoder_options(decoder, opts) \
((void) ((decoder)->options = (opts)))
int mad_decoder_run(struct mad_decoder *, enum mad_decoder_mode);
int mad_decoder_message(struct mad_decoder *, void *, unsigned int *);
# endif
# ifdef __cplusplus
}
# endif

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#ifndef __CONFIG_TYPES_H__
#define __CONFIG_TYPES_H__
/* these are filled in by configure or cmake*/
#define INCLUDE_INTTYPES_H 1
#define INCLUDE_STDINT_H 1
#define INCLUDE_SYS_TYPES_H 1
#if INCLUDE_INTTYPES_H
# include <inttypes.h>
#endif
#if INCLUDE_STDINT_H
# include <stdint.h>
#endif
#if INCLUDE_SYS_TYPES_H
# include <sys/types.h>
#endif
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
#endif

209
lib/codecs/inc/ogg/ogg.h Normal file
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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: toplevel libogg include
********************************************************************/
#ifndef _OGG_H
#define _OGG_H
#ifdef __cplusplus
extern "C" {
#endif
#include <stddef.h>
#include <ogg/os_types.h>
typedef struct {
void *iov_base;
size_t iov_len;
} ogg_iovec_t;
typedef struct {
long endbyte;
int endbit;
unsigned char *buffer;
unsigned char *ptr;
long storage;
} oggpack_buffer;
/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/
typedef struct {
unsigned char *header;
long header_len;
unsigned char *body;
long body_len;
} ogg_page;
/* ogg_stream_state contains the current encode/decode state of a logical
Ogg bitstream **********************************************************/
typedef struct {
unsigned char *body_data; /* bytes from packet bodies */
long body_storage; /* storage elements allocated */
long body_fill; /* elements stored; fill mark */
long body_returned; /* elements of fill returned */
int *lacing_vals; /* The values that will go to the segment table */
ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact
this way, but it is simple coupled to the
lacing fifo */
long lacing_storage;
long lacing_fill;
long lacing_packet;
long lacing_returned;
unsigned char header[282]; /* working space for header encode */
int header_fill;
int e_o_s; /* set when we have buffered the last packet in the
logical bitstream */
int b_o_s; /* set after we've written the initial page
of a logical bitstream */
long serialno;
long pageno;
ogg_int64_t packetno; /* sequence number for decode; the framing
knows where there's a hole in the data,
but we need coupling so that the codec
(which is in a separate abstraction
layer) also knows about the gap */
ogg_int64_t granulepos;
} ogg_stream_state;
/* ogg_packet is used to encapsulate the data and metadata belonging
to a single raw Ogg/Vorbis packet *************************************/
typedef struct {
unsigned char *packet;
long bytes;
long b_o_s;
long e_o_s;
ogg_int64_t granulepos;
ogg_int64_t packetno; /* sequence number for decode; the framing
knows where there's a hole in the data,
but we need coupling so that the codec
(which is in a separate abstraction
layer) also knows about the gap */
} ogg_packet;
typedef struct {
unsigned char *data;
int storage;
int fill;
int returned;
int unsynced;
int headerbytes;
int bodybytes;
} ogg_sync_state;
/* Ogg BITSTREAM PRIMITIVES: bitstream ************************/
extern void oggpack_writeinit(oggpack_buffer *b);
extern int oggpack_writecheck(oggpack_buffer *b);
extern void oggpack_writetrunc(oggpack_buffer *b,long bits);
extern void oggpack_writealign(oggpack_buffer *b);
extern void oggpack_writecopy(oggpack_buffer *b,void *source,long bits);
extern void oggpack_reset(oggpack_buffer *b);
extern void oggpack_writeclear(oggpack_buffer *b);
extern void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
extern void oggpack_write(oggpack_buffer *b,unsigned long value,int bits);
extern long oggpack_look(oggpack_buffer *b,int bits);
extern long oggpack_look1(oggpack_buffer *b);
extern void oggpack_adv(oggpack_buffer *b,int bits);
extern void oggpack_adv1(oggpack_buffer *b);
extern long oggpack_read(oggpack_buffer *b,int bits);
extern long oggpack_read1(oggpack_buffer *b);
extern long oggpack_bytes(oggpack_buffer *b);
extern long oggpack_bits(oggpack_buffer *b);
extern unsigned char *oggpack_get_buffer(oggpack_buffer *b);
extern void oggpackB_writeinit(oggpack_buffer *b);
extern int oggpackB_writecheck(oggpack_buffer *b);
extern void oggpackB_writetrunc(oggpack_buffer *b,long bits);
extern void oggpackB_writealign(oggpack_buffer *b);
extern void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits);
extern void oggpackB_reset(oggpack_buffer *b);
extern void oggpackB_writeclear(oggpack_buffer *b);
extern void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
extern void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits);
extern long oggpackB_look(oggpack_buffer *b,int bits);
extern long oggpackB_look1(oggpack_buffer *b);
extern void oggpackB_adv(oggpack_buffer *b,int bits);
extern void oggpackB_adv1(oggpack_buffer *b);
extern long oggpackB_read(oggpack_buffer *b,int bits);
extern long oggpackB_read1(oggpack_buffer *b);
extern long oggpackB_bytes(oggpack_buffer *b);
extern long oggpackB_bits(oggpack_buffer *b);
extern unsigned char *oggpackB_get_buffer(oggpack_buffer *b);
/* Ogg BITSTREAM PRIMITIVES: encoding **************************/
extern int ogg_stream_packetin(ogg_stream_state *os, ogg_packet *op);
extern int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov,
int count, long e_o_s, ogg_int64_t granulepos);
extern int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill);
extern int ogg_stream_flush(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_flush_fill(ogg_stream_state *os, ogg_page *og, int nfill);
/* Ogg BITSTREAM PRIMITIVES: decoding **************************/
extern int ogg_sync_init(ogg_sync_state *oy);
extern int ogg_sync_clear(ogg_sync_state *oy);
extern int ogg_sync_reset(ogg_sync_state *oy);
extern int ogg_sync_destroy(ogg_sync_state *oy);
extern int ogg_sync_check(ogg_sync_state *oy);
extern char *ogg_sync_buffer(ogg_sync_state *oy, long size);
extern int ogg_sync_wrote(ogg_sync_state *oy, long bytes);
extern long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og);
extern int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og);
extern int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op);
extern int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op);
/* Ogg BITSTREAM PRIMITIVES: general ***************************/
extern int ogg_stream_init(ogg_stream_state *os,int serialno);
extern int ogg_stream_clear(ogg_stream_state *os);
extern int ogg_stream_reset(ogg_stream_state *os);
extern int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno);
extern int ogg_stream_destroy(ogg_stream_state *os);
extern int ogg_stream_check(ogg_stream_state *os);
extern int ogg_stream_eos(ogg_stream_state *os);
extern void ogg_page_checksum_set(ogg_page *og);
extern int ogg_page_version(const ogg_page *og);
extern int ogg_page_continued(const ogg_page *og);
extern int ogg_page_bos(const ogg_page *og);
extern int ogg_page_eos(const ogg_page *og);
extern ogg_int64_t ogg_page_granulepos(const ogg_page *og);
extern int ogg_page_serialno(const ogg_page *og);
extern long ogg_page_pageno(const ogg_page *og);
extern int ogg_page_packets(const ogg_page *og);
extern void ogg_packet_clear(ogg_packet *op);
#ifdef __cplusplus
}
#endif
#endif /* _OGG_H */

158
lib/codecs/inc/ogg/os_types.h Normal file
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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: Define a consistent set of types on each platform.
********************************************************************/
#ifndef _OS_TYPES_H
#define _OS_TYPES_H
/* make it easy on the folks that want to compile the libs with a
different malloc than stdlib */
#define _ogg_malloc malloc
#define _ogg_calloc calloc
#define _ogg_realloc realloc
#define _ogg_free free
#if defined(_WIN32)
# if defined(__CYGWIN__)
# include <stdint.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
# elif defined(__MINGW32__)
# include <sys/types.h>
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
# elif defined(__MWERKS__)
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
# else
# if defined(_MSC_VER) && (_MSC_VER >= 1800) /* MSVC 2013 and newer */
# include <stdint.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
# else
/* MSVC/Borland */
typedef __int64 ogg_int64_t;
typedef __int32 ogg_int32_t;
typedef unsigned __int32 ogg_uint32_t;
typedef unsigned __int64 ogg_uint64_t;
typedef __int16 ogg_int16_t;
typedef unsigned __int16 ogg_uint16_t;
# endif
# endif
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <sys/types.h>
typedef int16_t ogg_int16_t;
typedef u_int16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef u_int32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef u_int64_t ogg_uint64_t;
#elif defined(__HAIKU__)
/* Haiku */
# include <sys/types.h>
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
#elif defined (DJGPP)
/* DJGPP */
typedef short ogg_int16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
#elif defined(R5900)
/* PS2 EE */
typedef long ogg_int64_t;
typedef unsigned long ogg_uint64_t;
typedef int ogg_int32_t;
typedef unsigned ogg_uint32_t;
typedef short ogg_int16_t;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef signed int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long int ogg_int64_t;
typedef unsigned long long int ogg_uint64_t;
#elif defined(__TMS320C6X__)
/* TI C64x compiler */
typedef signed short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef signed int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long int ogg_int64_t;
typedef unsigned long long int ogg_uint64_t;
#else
# include <ogg/config_types.h>
#endif
#endif /* _OS_TYPES_H */

981
lib/codecs/inc/opus/opus.h Normal file
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/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
Written by Jean-Marc Valin and Koen Vos */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- 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 COPYRIGHT HOLDERS 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 COPYRIGHT OWNER
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.
*/
/**
* @file opus.h
* @brief Opus reference implementation API
*/
#ifndef OPUS_H
#define OPUS_H
#include "opus_types.h"
#include "opus_defines.h"
#ifdef __cplusplus
extern "C" {
#endif
/**
* @mainpage Opus
*
* The Opus codec is designed for interactive speech and audio transmission over the Internet.
* It is designed by the IETF Codec Working Group and incorporates technology from
* Skype's SILK codec and Xiph.Org's CELT codec.
*
* The Opus codec is designed to handle a wide range of interactive audio applications,
* including Voice over IP, videoconferencing, in-game chat, and even remote live music
* performances. It can scale from low bit-rate narrowband speech to very high quality
* stereo music. Its main features are:
* @li Sampling rates from 8 to 48 kHz
* @li Bit-rates from 6 kb/s to 510 kb/s
* @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
* @li Audio bandwidth from narrowband to full-band
* @li Support for speech and music
* @li Support for mono and stereo
* @li Support for multichannel (up to 255 channels)
* @li Frame sizes from 2.5 ms to 60 ms
* @li Good loss robustness and packet loss concealment (PLC)
* @li Floating point and fixed-point implementation
*
* Documentation sections:
* @li @ref opus_encoder
* @li @ref opus_decoder
* @li @ref opus_repacketizer
* @li @ref opus_multistream
* @li @ref opus_libinfo
* @li @ref opus_custom
*/
/** @defgroup opus_encoder Opus Encoder
* @{
*
* @brief This page describes the process and functions used to encode Opus.
*
* Since Opus is a stateful codec, the encoding process starts with creating an encoder
* state. This can be done with:
*
* @code
* int error;
* OpusEncoder *enc;
* enc = opus_encoder_create(Fs, channels, application, &error);
* @endcode
*
* From this point, @c enc can be used for encoding an audio stream. An encoder state
* @b must @b not be used for more than one stream at the same time. Similarly, the encoder
* state @b must @b not be re-initialized for each frame.
*
* While opus_encoder_create() allocates memory for the state, it's also possible
* to initialize pre-allocated memory:
*
* @code
* int size;
* int error;
* OpusEncoder *enc;
* size = opus_encoder_get_size(channels);
* enc = malloc(size);
* error = opus_encoder_init(enc, Fs, channels, application);
* @endcode
*
* where opus_encoder_get_size() returns the required size for the encoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
*
* The encoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
*
* It is possible to change some of the encoder's settings using the opus_encoder_ctl()
* interface. All these settings already default to the recommended value, so they should
* only be changed when necessary. The most common settings one may want to change are:
*
* @code
* opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
* opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
* opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
* @endcode
*
* where
*
* @arg bitrate is in bits per second (b/s)
* @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
* @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
*
* See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
*
* To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
* @code
* len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
* @endcode
*
* where
* <ul>
* <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
* <li>frame_size is the duration of the frame in samples (per channel)</li>
* <li>packet is the byte array to which the compressed data is written</li>
* <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
* Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
* </ul>
*
* opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
* The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
* is 2 bytes or less, then the packet does not need to be transmitted (DTX).
*
* Once the encoder state if no longer needed, it can be destroyed with
*
* @code
* opus_encoder_destroy(enc);
* @endcode
*
* If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
* then no action is required aside from potentially freeing the memory that was manually
* allocated for it (calling free(enc) for the example above)
*
*/
/** Opus encoder state.
* This contains the complete state of an Opus encoder.
* It is position independent and can be freely copied.
* @see opus_encoder_create,opus_encoder_init
*/
typedef struct OpusEncoder OpusEncoder;
/** Gets the size of an <code>OpusEncoder</code> structure.
* @param[in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
/**
*/
/** Allocates and initializes an encoder state.
* There are three coding modes:
*
* @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
* signals. It enhances the input signal by high-pass filtering and
* emphasizing formants and harmonics. Optionally it includes in-band
* forward error correction to protect against packet loss. Use this
* mode for typical VoIP applications. Because of the enhancement,
* even at high bitrates the output may sound different from the input.
*
* @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
* non-voice signals like music. Use this mode for music and mixed
* (music/voice) content, broadcast, and applications requiring less
* than 15 ms of coding delay.
*
* @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
* disables the speech-optimized mode in exchange for slightly reduced delay.
* This mode can only be set on an newly initialized or freshly reset encoder
* because it changes the codec delay.
*
* This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @param [out] error <tt>int*</tt>: @ref opus_errorcodes
* @note Regardless of the sampling rate and number channels selected, the Opus encoder
* can switch to a lower audio bandwidth or number of channels if the bitrate
* selected is too low. This also means that it is safe to always use 48 kHz stereo input
* and let the encoder optimize the encoding.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
opus_int32 Fs,
int channels,
int application,
int *error
);
/** Initializes a previously allocated encoder state
* The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_encoder_create(),opus_encoder_get_size()
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (one of OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, or OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_encoder_init(
OpusEncoder *st,
opus_int32 Fs,
int channels,
int application
) OPUS_ARG_NONNULL(1);
/** Encodes an Opus frame.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
OpusEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes an Opus frame from floating point input.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
* Samples with a range beyond +/-1.0 are supported but will
* be clipped by decoders using the integer API and should
* only be used if it is known that the far end supports
* extended dynamic range.
* length is frame_size*channels*sizeof(float)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
OpusEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
* @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
/** Perform a CTL function on an Opus encoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param st <tt>OpusEncoder*</tt>: Encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_encoderctls.
* @see opus_genericctls
* @see opus_encoderctls
*/
OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/** @defgroup opus_decoder Opus Decoder
* @{
*
* @brief This page describes the process and functions used to decode Opus.
*
* The decoding process also starts with creating a decoder
* state. This can be done with:
* @code
* int error;
* OpusDecoder *dec;
* dec = opus_decoder_create(Fs, channels, &error);
* @endcode
* where
* @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
* @li channels is the number of channels (1 or 2)
* @li error will hold the error code in case of failure (or #OPUS_OK on success)
* @li the return value is a newly created decoder state to be used for decoding
*
* While opus_decoder_create() allocates memory for the state, it's also possible
* to initialize pre-allocated memory:
* @code
* int size;
* int error;
* OpusDecoder *dec;
* size = opus_decoder_get_size(channels);
* dec = malloc(size);
* error = opus_decoder_init(dec, Fs, channels);
* @endcode
* where opus_decoder_get_size() returns the required size for the decoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
*
* The decoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
*
* To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
* @code
* frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
* @endcode
* where
*
* @li packet is the byte array containing the compressed data
* @li len is the exact number of bytes contained in the packet
* @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
* @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
*
* opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
* If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
* buffer is too small to hold the decoded audio.
*
* Opus is a stateful codec with overlapping blocks and as a result Opus
* packets are not coded independently of each other. Packets must be
* passed into the decoder serially and in the correct order for a correct
* decode. Lost packets can be replaced with loss concealment by calling
* the decoder with a null pointer and zero length for the missing packet.
*
* A single codec state may only be accessed from a single thread at
* a time and any required locking must be performed by the caller. Separate
* streams must be decoded with separate decoder states and can be decoded
* in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
* defined.
*
*/
/** Opus decoder state.
* This contains the complete state of an Opus decoder.
* It is position independent and can be freely copied.
* @see opus_decoder_create,opus_decoder_init
*/
typedef struct OpusDecoder OpusDecoder;
/** Gets the size of an <code>OpusDecoder</code> structure.
* @param [in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
/** Allocates and initializes a decoder state.
* @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
* @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
*
* Internally Opus stores data at 48000 Hz, so that should be the default
* value for Fs. However, the decoder can efficiently decode to buffers
* at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
* data at the full sample rate, or knows the compressed data doesn't
* use the full frequency range, it can request decoding at a reduced
* rate. Likewise, the decoder is capable of filling in either mono or
* interleaved stereo pcm buffers, at the caller's request.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
opus_int32 Fs,
int channels,
int *error
);
/** Initializes a previously allocated decoder state.
* The state must be at least the size returned by opus_decoder_get_size().
* This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_decoder_init(
OpusDecoder *st,
opus_int32 Fs,
int channels
) OPUS_ARG_NONNULL(1);
/** Decode an Opus packet.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available, the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an Opus packet with floating point output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(float)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus decoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param st <tt>OpusDecoder*</tt>: Decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_decoderctls.
* @see opus_genericctls
* @see opus_decoderctls
*/
OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
* @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
/** Parse an opus packet into one or more frames.
* Opus_decode will perform this operation internally so most applications do
* not need to use this function.
* This function does not copy the frames, the returned pointers are pointers into
* the input packet.
* @param [in] data <tt>char*</tt>: Opus packet to be parsed
* @param [in] len <tt>opus_int32</tt>: size of data
* @param [out] out_toc <tt>char*</tt>: TOC pointer
* @param [out] frames <tt>char*[48]</tt> encapsulated frames
* @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
* @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
* @returns number of frames
*/
OPUS_EXPORT int opus_packet_parse(
const unsigned char *data,
opus_int32 len,
unsigned char *out_toc,
const unsigned char *frames[48],
opus_int16 size[48],
int *payload_offset
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
/** Gets the bandwidth of an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet
* @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
* @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
* @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
* @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
* @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
/** Gets the number of samples per frame from an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet.
* This must contain at least one byte of
* data.
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
* This must be a multiple of 400, or
* inaccurate results will be returned.
* @returns Number of samples per frame.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Gets the number of channels from an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet
* @returns Number of channels
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
/** Gets the number of frames in an Opus packet.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns Number of frames
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
/** Gets the number of samples of an Opus packet.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
* This must be a multiple of 400, or
* inaccurate results will be returned.
* @returns Number of samples
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Gets the number of samples of an Opus packet.
* @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns Number of samples
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
* the signal is already in that range, nothing is done. If there are values
* outside of [-1,1], then the signal is clipped as smoothly as possible to
* both fit in the range and avoid creating excessive distortion in the
* process.
* @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
* @param [in] frame_size <tt>int</tt> Number of samples per channel to process
* @param [in] channels <tt>int</tt>: Number of channels
* @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
*/
OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
/**@}*/
/** @defgroup opus_repacketizer Repacketizer
* @{
*
* The repacketizer can be used to merge multiple Opus packets into a single
* packet or alternatively to split Opus packets that have previously been
* merged. Splitting valid Opus packets is always guaranteed to succeed,
* whereas merging valid packets only succeeds if all frames have the same
* mode, bandwidth, and frame size, and when the total duration of the merged
* packet is no more than 120 ms. The 120 ms limit comes from the
* specification and limits decoder memory requirements at a point where
* framing overhead becomes negligible.
*
* The repacketizer currently only operates on elementary Opus
* streams. It will not manipualte multistream packets successfully, except in
* the degenerate case where they consist of data from a single stream.
*
* The repacketizing process starts with creating a repacketizer state, either
* by calling opus_repacketizer_create() or by allocating the memory yourself,
* e.g.,
* @code
* OpusRepacketizer *rp;
* rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
* if (rp != NULL)
* opus_repacketizer_init(rp);
* @endcode
*
* Then the application should submit packets with opus_repacketizer_cat(),
* extract new packets with opus_repacketizer_out() or
* opus_repacketizer_out_range(), and then reset the state for the next set of
* input packets via opus_repacketizer_init().
*
* For example, to split a sequence of packets into individual frames:
* @code
* unsigned char *data;
* int len;
* while (get_next_packet(&data, &len))
* {
* unsigned char out[1276];
* opus_int32 out_len;
* int nb_frames;
* int err;
* int i;
* err = opus_repacketizer_cat(rp, data, len);
* if (err != OPUS_OK)
* {
* release_packet(data);
* return err;
* }
* nb_frames = opus_repacketizer_get_nb_frames(rp);
* for (i = 0; i < nb_frames; i++)
* {
* out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
* if (out_len < 0)
* {
* release_packet(data);
* return (int)out_len;
* }
* output_next_packet(out, out_len);
* }
* opus_repacketizer_init(rp);
* release_packet(data);
* }
* @endcode
*
* Alternatively, to combine a sequence of frames into packets that each
* contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
* @code
* // The maximum number of packets with duration TARGET_DURATION_MS occurs
* // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
* // packets.
* unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
* opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
* int nb_packets;
* unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
* opus_int32 out_len;
* int prev_toc;
* nb_packets = 0;
* while (get_next_packet(data+nb_packets, len+nb_packets))
* {
* int nb_frames;
* int err;
* nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
* if (nb_frames < 1)
* {
* release_packets(data, nb_packets+1);
* return nb_frames;
* }
* nb_frames += opus_repacketizer_get_nb_frames(rp);
* // If adding the next packet would exceed our target, or it has an
* // incompatible TOC sequence, output the packets we already have before
* // submitting it.
* // N.B., The nb_packets > 0 check ensures we've submitted at least one
* // packet since the last call to opus_repacketizer_init(). Otherwise a
* // single packet longer than TARGET_DURATION_MS would cause us to try to
* // output an (invalid) empty packet. It also ensures that prev_toc has
* // been set to a valid value. Additionally, len[nb_packets] > 0 is
* // guaranteed by the call to opus_packet_get_nb_frames() above, so the
* // reference to data[nb_packets][0] should be valid.
* if (nb_packets > 0 && (
* ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
* opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
* TARGET_DURATION_MS*48))
* {
* out_len = opus_repacketizer_out(rp, out, sizeof(out));
* if (out_len < 0)
* {
* release_packets(data, nb_packets+1);
* return (int)out_len;
* }
* output_next_packet(out, out_len);
* opus_repacketizer_init(rp);
* release_packets(data, nb_packets);
* data[0] = data[nb_packets];
* len[0] = len[nb_packets];
* nb_packets = 0;
* }
* err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
* if (err != OPUS_OK)
* {
* release_packets(data, nb_packets+1);
* return err;
* }
* prev_toc = data[nb_packets][0];
* nb_packets++;
* }
* // Output the final, partial packet.
* if (nb_packets > 0)
* {
* out_len = opus_repacketizer_out(rp, out, sizeof(out));
* release_packets(data, nb_packets);
* if (out_len < 0)
* return (int)out_len;
* output_next_packet(out, out_len);
* }
* @endcode
*
* An alternate way of merging packets is to simply call opus_repacketizer_cat()
* unconditionally until it fails. At that point, the merged packet can be
* obtained with opus_repacketizer_out() and the input packet for which
* opus_repacketizer_cat() needs to be re-added to a newly reinitialized
* repacketizer state.
*/
typedef struct OpusRepacketizer OpusRepacketizer;
/** Gets the size of an <code>OpusRepacketizer</code> structure.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
/** (Re)initializes a previously allocated repacketizer state.
* The state must be at least the size returned by opus_repacketizer_get_size().
* This can be used for applications which use their own allocator instead of
* malloc().
* It must also be called to reset the queue of packets waiting to be
* repacketized, which is necessary if the maximum packet duration of 120 ms
* is reached or if you wish to submit packets with a different Opus
* configuration (coding mode, audio bandwidth, frame size, or channel count).
* Failure to do so will prevent a new packet from being added with
* opus_repacketizer_cat().
* @see opus_repacketizer_create
* @see opus_repacketizer_get_size
* @see opus_repacketizer_cat
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
* (re)initialize.
* @returns A pointer to the same repacketizer state that was passed in.
*/
OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
/** Allocates memory and initializes the new repacketizer with
* opus_repacketizer_init().
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
/** Frees an <code>OpusRepacketizer</code> allocated by
* opus_repacketizer_create().
* @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
/** Add a packet to the current repacketizer state.
* This packet must match the configuration of any packets already submitted
* for repacketization since the last call to opus_repacketizer_init().
* This means that it must have the same coding mode, audio bandwidth, frame
* size, and channel count.
* This can be checked in advance by examining the top 6 bits of the first
* byte of the packet, and ensuring they match the top 6 bits of the first
* byte of any previously submitted packet.
* The total duration of audio in the repacketizer state also must not exceed
* 120 ms, the maximum duration of a single packet, after adding this packet.
*
* The contents of the current repacketizer state can be extracted into new
* packets using opus_repacketizer_out() or opus_repacketizer_out_range().
*
* In order to add a packet with a different configuration or to add more
* audio beyond 120 ms, you must clear the repacketizer state by calling
* opus_repacketizer_init().
* If a packet is too large to add to the current repacketizer state, no part
* of it is added, even if it contains multiple frames, some of which might
* fit.
* If you wish to be able to add parts of such packets, you should first use
* another repacketizer to split the packet into pieces and add them
* individually.
* @see opus_repacketizer_out_range
* @see opus_repacketizer_out
* @see opus_repacketizer_init
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
* add the packet.
* @param[in] data <tt>const unsigned char*</tt>: The packet data.
* The application must ensure
* this pointer remains valid
* until the next call to
* opus_repacketizer_init() or
* opus_repacketizer_destroy().
* @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
* @returns An error code indicating whether or not the operation succeeded.
* @retval #OPUS_OK The packet's contents have been added to the repacketizer
* state.
* @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
* the packet's TOC sequence was not compatible
* with previously submitted packets (because
* the coding mode, audio bandwidth, frame size,
* or channel count did not match), or adding
* this packet would increase the total amount of
* audio stored in the repacketizer state to more
* than 120 ms.
*/
OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** Construct a new packet from data previously submitted to the repacketizer
* state via opus_repacketizer_cat().
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
* construct the new packet.
* @param begin <tt>int</tt>: The index of the first frame in the current
* repacketizer state to include in the output.
* @param end <tt>int</tt>: One past the index of the last frame in the
* current repacketizer state to include in the
* output.
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
* store the output packet.
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
* the output buffer. In order to guarantee
* success, this should be at least
* <code>1276</code> for a single frame,
* or for multiple frames,
* <code>1277*(end-begin)</code>.
* However, <code>1*(end-begin)</code> plus
* the size of all packet data submitted to
* the repacketizer since the last call to
* opus_repacketizer_init() or
* opus_repacketizer_create() is also
* sufficient, and possibly much smaller.
* @returns The total size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
* frames (begin < 0, begin >= end, or end >
* opus_repacketizer_get_nb_frames()).
* @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
* complete output packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Return the total number of frames contained in packet data submitted to
* the repacketizer state so far via opus_repacketizer_cat() since the last
* call to opus_repacketizer_init() or opus_repacketizer_create().
* This defines the valid range of packets that can be extracted with
* opus_repacketizer_out_range() or opus_repacketizer_out().
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
* frames.
* @returns The total number of frames contained in the packet data submitted
* to the repacketizer state.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
/** Construct a new packet from data previously submitted to the repacketizer
* state via opus_repacketizer_cat().
* This is a convenience routine that returns all the data submitted so far
* in a single packet.
* It is equivalent to calling
* @code
* opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
* data, maxlen)
* @endcode
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
* construct the new packet.
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
* store the output packet.
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
* the output buffer. In order to guarantee
* success, this should be at least
* <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
* However,
* <code>1*opus_repacketizer_get_nb_frames(rp)</code>
* plus the size of all packet data
* submitted to the repacketizer since the
* last call to opus_repacketizer_init() or
* opus_repacketizer_create() is also
* sufficient, and possibly much smaller.
* @returns The total size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
* complete output packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to pad.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
* This must be at least as large as len.
* @returns an error code
* @retval #OPUS_OK \a on success.
* @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
* minimize space usage.
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to strip.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @returns The new size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG \a len was less than 1.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to pad.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
* This must be at least 1.
* @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
* This must be at least as large as len.
* @returns an error code
* @retval #OPUS_OK \a on success.
* @retval #OPUS_BAD_ARG \a len was less than 1.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
* minimize space usage.
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to strip.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
* This must be at least 1.
* @returns The new size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_H */

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/* Copyright (c) 2007-2008 CSIRO
Copyright (c) 2007-2009 Xiph.Org Foundation
Copyright (c) 2008-2012 Gregory Maxwell
Written by Jean-Marc Valin and Gregory Maxwell */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- 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 COPYRIGHT HOLDERS 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 COPYRIGHT OWNER
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.
*/
/**
@file opus_custom.h
@brief Opus-Custom reference implementation API
*/
#ifndef OPUS_CUSTOM_H
#define OPUS_CUSTOM_H
#include "opus_defines.h"
#ifdef __cplusplus
extern "C" {
#endif
#ifdef CUSTOM_MODES
# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
#else
# define OPUS_CUSTOM_EXPORT
# ifdef OPUS_BUILD
# define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
# else
# define OPUS_CUSTOM_EXPORT_STATIC
# endif
#endif
/** @defgroup opus_custom Opus Custom
* @{
* Opus Custom is an optional part of the Opus specification and
* reference implementation which uses a distinct API from the regular
* API and supports frame sizes that are not normally supported.\ Use
* of Opus Custom is discouraged for all but very special applications
* for which a frame size different from 2.5, 5, 10, or 20 ms is needed
* (for either complexity or latency reasons) and where interoperability
* is less important.
*
* In addition to the interoperability limitations the use of Opus custom
* disables a substantial chunk of the codec and generally lowers the
* quality available at a given bitrate. Normally when an application needs
* a different frame size from the codec it should buffer to match the
* sizes but this adds a small amount of delay which may be important
* in some very low latency applications. Some transports (especially
* constant rate RF transports) may also work best with frames of
* particular durations.
*
* Libopus only supports custom modes if they are enabled at compile time.
*
* The Opus Custom API is similar to the regular API but the
* @ref opus_encoder_create and @ref opus_decoder_create calls take
* an additional mode parameter which is a structure produced by
* a call to @ref opus_custom_mode_create. Both the encoder and decoder
* must create a mode using the same sample rate (fs) and frame size
* (frame size) so these parameters must either be signaled out of band
* or fixed in a particular implementation.
*
* Similar to regular Opus the custom modes support on the fly frame size
* switching, but the sizes available depend on the particular frame size in
* use. For some initial frame sizes on a single on the fly size is available.
*/
/** Contains the state of an encoder. One encoder state is needed
for each stream. It is initialized once at the beginning of the
stream. Do *not* re-initialize the state for every frame.
@brief Encoder state
*/
typedef struct OpusCustomEncoder OpusCustomEncoder;
/** State of the decoder. One decoder state is needed for each stream.
It is initialized once at the beginning of the stream. Do *not*
re-initialize the state for every frame.
@brief Decoder state
*/
typedef struct OpusCustomDecoder OpusCustomDecoder;
/** The mode contains all the information necessary to create an
encoder. Both the encoder and decoder need to be initialized
with exactly the same mode, otherwise the output will be
corrupted. The mode MUST NOT BE DESTROYED until the encoders and
decoders that use it are destroyed as well.
@brief Mode configuration
*/
typedef struct OpusCustomMode OpusCustomMode;
/** Creates a new mode struct. This will be passed to an encoder or
* decoder. The mode MUST NOT BE DESTROYED until the encoders and
* decoders that use it are destroyed as well.
* @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
* @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
* packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
* @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
* @return A newly created mode
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
/** Destroys a mode struct. Only call this after all encoders and
* decoders using this mode are destroyed as well.
* @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
*/
OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
/* Encoder */
/** Gets the size of an OpusCustomEncoder structure.
* @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
* @param [in] channels <tt>int</tt>: Number of channels
* @returns size
*/
OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1);
# ifdef CUSTOM_MODES
/** Initializes a previously allocated encoder state
* The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
* To reset a previously initialized state use the OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
* the stream (must be the same characteristics as used for the
* decoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @return OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
OpusCustomEncoder *st,
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
# endif
#endif
/** Creates a new encoder state. Each stream needs its own encoder
* state (can't be shared across simultaneous streams).
* @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
* the stream (must be the same characteristics as used for the
* decoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @param [out] error <tt>int*</tt>: Returns an error code
* @return Newly created encoder state.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
const OpusCustomMode *mode,
int channels,
int *error
) OPUS_ARG_NONNULL(1);
/** Destroys an encoder state.
* @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
*/
OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
/** Encodes a frame of audio.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
* Samples with a range beyond +/-1.0 are supported but will
* be clipped by decoders using the integer API and should
* only be used if it is known that the far end supports
* extended dynamic range. There must be exactly
* frame_size samples per channel.
* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
* @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
* @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
* (can change from one frame to another)
* @return Number of bytes written to "compressed".
* If negative, an error has occurred (see error codes). It is IMPORTANT that
* the length returned be somehow transmitted to the decoder. Otherwise, no
* decoding is possible.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
OpusCustomEncoder *st,
const float *pcm,
int frame_size,
unsigned char *compressed,
int maxCompressedBytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a frame of audio.
* @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
* There must be exactly frame_size samples per channel.
* @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
* @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
* @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
* (can change from one frame to another)
* @return Number of bytes written to "compressed".
* If negative, an error has occurred (see error codes). It is IMPORTANT that
* the length returned be somehow transmitted to the decoder. Otherwise, no
* decoding is possible.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
OpusCustomEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *compressed,
int maxCompressedBytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus custom encoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @see opus_encoderctls
*/
OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
/* Decoder */
/** Gets the size of an OpusCustomDecoder structure.
* @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
* @param [in] channels <tt>int</tt>: Number of channels
* @returns size
*/
OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1);
/** Initializes a previously allocated decoder state
* The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
* To reset a previously initialized state use the OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
* the stream (must be the same characteristics as used for the
* encoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @return OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
OpusCustomDecoder *st,
const OpusCustomMode *mode,
int channels
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
#endif
/** Creates a new decoder state. Each stream needs its own decoder state (can't
* be shared across simultaneous streams).
* @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
* stream (must be the same characteristics as used for the encoder)
* @param [in] channels <tt>int</tt>: Number of channels
* @param [out] error <tt>int*</tt>: Returns an error code
* @return Newly created decoder state.
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
const OpusCustomMode *mode,
int channels,
int *error
) OPUS_ARG_NONNULL(1);
/** Destroys a decoder state.
* @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
*/
OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
/** Decode an opus custom frame with floating point output
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>int</tt>: Number of bytes in payload
* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(float)
* @param [in] frame_size Number of samples per channel of available space in *pcm.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
OpusCustomDecoder *st,
const unsigned char *data,
int len,
float *pcm,
int frame_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an opus custom frame
* @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>int</tt>: Number of bytes in payload
* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel of available space in *pcm.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
OpusCustomDecoder *st,
const unsigned char *data,
int len,
opus_int16 *pcm,
int frame_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus custom decoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @see opus_genericctls
*/
OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_CUSTOM_H */

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/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
Written by Jean-Marc Valin and Koen Vos */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- 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 COPYRIGHT HOLDERS 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 COPYRIGHT OWNER
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.
*/
/**
* @file opus_defines.h
* @brief Opus reference implementation constants
*/
#ifndef OPUS_DEFINES_H
#define OPUS_DEFINES_H
#include "opus_types.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @defgroup opus_errorcodes Error codes
* @{
*/
/** No error @hideinitializer*/
#define OPUS_OK 0
/** One or more invalid/out of range arguments @hideinitializer*/
#define OPUS_BAD_ARG -1
/** Not enough bytes allocated in the buffer @hideinitializer*/
#define OPUS_BUFFER_TOO_SMALL -2
/** An internal error was detected @hideinitializer*/
#define OPUS_INTERNAL_ERROR -3
/** The compressed data passed is corrupted @hideinitializer*/
#define OPUS_INVALID_PACKET -4
/** Invalid/unsupported request number @hideinitializer*/
#define OPUS_UNIMPLEMENTED -5
/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
#define OPUS_INVALID_STATE -6
/** Memory allocation has failed @hideinitializer*/
#define OPUS_ALLOC_FAIL -7
/**@}*/
/** @cond OPUS_INTERNAL_DOC */
/**Export control for opus functions */
#ifndef OPUS_EXPORT
# if defined(_WIN32)
# if defined(OPUS_BUILD) && defined(DLL_EXPORT)
# define OPUS_EXPORT __declspec(dllexport)
# else
# define OPUS_EXPORT
# endif
# elif defined(__GNUC__) && defined(OPUS_BUILD)
# define OPUS_EXPORT __attribute__ ((visibility ("default")))
# else
# define OPUS_EXPORT
# endif
#endif
# if !defined(OPUS_GNUC_PREREQ)
# if defined(__GNUC__)&&defined(__GNUC_MINOR__)
# define OPUS_GNUC_PREREQ(_maj,_min) \
((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
# else
# define OPUS_GNUC_PREREQ(_maj,_min) 0
# endif
# endif
#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
# if OPUS_GNUC_PREREQ(3,0)
# define OPUS_RESTRICT __restrict__
# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
# define OPUS_RESTRICT __restrict
# else
# define OPUS_RESTRICT
# endif
#else
# define OPUS_RESTRICT restrict
#endif
#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
# if OPUS_GNUC_PREREQ(2,7)
# define OPUS_INLINE __inline__
# elif (defined(_MSC_VER))
# define OPUS_INLINE __inline
# else
# define OPUS_INLINE
# endif
#else
# define OPUS_INLINE inline
#endif
/**Warning attributes for opus functions
* NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
* some paranoid null checks. */
#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
#else
# define OPUS_WARN_UNUSED_RESULT
#endif
#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x)))
#else
# define OPUS_ARG_NONNULL(_x)
#endif
/** These are the actual Encoder CTL ID numbers.
* They should not be used directly by applications.
* In general, SETs should be even and GETs should be odd.*/
#define OPUS_SET_APPLICATION_REQUEST 4000
#define OPUS_GET_APPLICATION_REQUEST 4001
#define OPUS_SET_BITRATE_REQUEST 4002
#define OPUS_GET_BITRATE_REQUEST 4003
#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004
#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005
#define OPUS_SET_VBR_REQUEST 4006
#define OPUS_GET_VBR_REQUEST 4007
#define OPUS_SET_BANDWIDTH_REQUEST 4008
#define OPUS_GET_BANDWIDTH_REQUEST 4009
#define OPUS_SET_COMPLEXITY_REQUEST 4010
#define OPUS_GET_COMPLEXITY_REQUEST 4011
#define OPUS_SET_INBAND_FEC_REQUEST 4012
#define OPUS_GET_INBAND_FEC_REQUEST 4013
#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014
#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015
#define OPUS_SET_DTX_REQUEST 4016
#define OPUS_GET_DTX_REQUEST 4017
#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020
#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021
#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022
#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023
#define OPUS_SET_SIGNAL_REQUEST 4024
#define OPUS_GET_SIGNAL_REQUEST 4025
#define OPUS_GET_LOOKAHEAD_REQUEST 4027
/* #define OPUS_RESET_STATE 4028 */
#define OPUS_GET_SAMPLE_RATE_REQUEST 4029
#define OPUS_GET_FINAL_RANGE_REQUEST 4031
#define OPUS_GET_PITCH_REQUEST 4033
#define OPUS_SET_GAIN_REQUEST 4034
#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */
#define OPUS_SET_LSB_DEPTH_REQUEST 4036
#define OPUS_GET_LSB_DEPTH_REQUEST 4037
#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
#define OPUS_GET_IN_DTX_REQUEST 4049
/** Defines for the presence of extended APIs. */
#define OPUS_HAVE_OPUS_PROJECTION_H
/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
/** @endcond */
/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
* @see opus_genericctls, opus_encoderctls
* @{
*/
/* Values for the various encoder CTLs */
#define OPUS_AUTO -1000 /**<Auto/default setting @hideinitializer*/
#define OPUS_BITRATE_MAX -1 /**<Maximum bitrate @hideinitializer*/
/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
* @hideinitializer */
#define OPUS_APPLICATION_VOIP 2048
/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
* @hideinitializer */
#define OPUS_APPLICATION_AUDIO 2049
/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
* @hideinitializer */
#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
#define OPUS_SIGNAL_VOICE 3001 /**< Signal being encoded is voice */
#define OPUS_SIGNAL_MUSIC 3002 /**< Signal being encoded is music */
#define OPUS_BANDWIDTH_NARROWBAND 1101 /**< 4 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_MEDIUMBAND 1102 /**< 6 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_WIDEBAND 1103 /**< 8 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_SUPERWIDEBAND 1104 /**<12 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_FULLBAND 1105 /**<20 kHz bandpass @hideinitializer*/
#define OPUS_FRAMESIZE_ARG 5000 /**< Select frame size from the argument (default) */
#define OPUS_FRAMESIZE_2_5_MS 5001 /**< Use 2.5 ms frames */
#define OPUS_FRAMESIZE_5_MS 5002 /**< Use 5 ms frames */
#define OPUS_FRAMESIZE_10_MS 5003 /**< Use 10 ms frames */
#define OPUS_FRAMESIZE_20_MS 5004 /**< Use 20 ms frames */
#define OPUS_FRAMESIZE_40_MS 5005 /**< Use 40 ms frames */
#define OPUS_FRAMESIZE_60_MS 5006 /**< Use 60 ms frames */
#define OPUS_FRAMESIZE_80_MS 5007 /**< Use 80 ms frames */
#define OPUS_FRAMESIZE_100_MS 5008 /**< Use 100 ms frames */
#define OPUS_FRAMESIZE_120_MS 5009 /**< Use 120 ms frames */
/**@}*/
/** @defgroup opus_encoderctls Encoder related CTLs
*
* These are convenience macros for use with the \c opus_encode_ctl
* interface. They are used to generate the appropriate series of
* arguments for that call, passing the correct type, size and so
* on as expected for each particular request.
*
* Some usage examples:
*
* @code
* int ret;
* ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
* if (ret != OPUS_OK) return ret;
*
* opus_int32 rate;
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
*
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
* @endcode
*
* @see opus_genericctls, opus_encoder
* @{
*/
/** Configures the encoder's computational complexity.
* The supported range is 0-10 inclusive with 10 representing the highest complexity.
* @see OPUS_GET_COMPLEXITY
* @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
*
* @hideinitializer */
#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
/** Gets the encoder's complexity configuration.
* @see OPUS_SET_COMPLEXITY
* @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
* inclusive.
* @hideinitializer */
#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
/** Configures the bitrate in the encoder.
* Rates from 500 to 512000 bits per second are meaningful, as well as the
* special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
* The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
* rate as it can, which is useful for controlling the rate by adjusting the
* output buffer size.
* @see OPUS_GET_BITRATE
* @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
* is determined based on the number of
* channels and the input sampling rate.
* @hideinitializer */
#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
/** Gets the encoder's bitrate configuration.
* @see OPUS_SET_BITRATE
* @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
* The default is determined based on the
* number of channels and the input
* sampling rate.
* @hideinitializer */
#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
/** Enables or disables variable bitrate (VBR) in the encoder.
* The configured bitrate may not be met exactly because frames must
* be an integer number of bytes in length.
* @see OPUS_GET_VBR
* @see OPUS_SET_VBR_CONSTRAINT
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
* cause noticeable quality degradation.</dd>
* <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
* #OPUS_SET_VBR_CONSTRAINT.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
/** Determine if variable bitrate (VBR) is enabled in the encoder.
* @see OPUS_SET_VBR
* @see OPUS_GET_VBR_CONSTRAINT
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Hard CBR.</dd>
* <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
* #OPUS_GET_VBR_CONSTRAINT.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
/** Enables or disables constrained VBR in the encoder.
* This setting is ignored when the encoder is in CBR mode.
* @warning Only the MDCT mode of Opus currently heeds the constraint.
* Speech mode ignores it completely, hybrid mode may fail to obey it
* if the LPC layer uses more bitrate than the constraint would have
* permitted.
* @see OPUS_GET_VBR_CONSTRAINT
* @see OPUS_SET_VBR
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Unconstrained VBR.</dd>
* <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
* frame of buffering delay assuming a transport with a
* serialization speed of the nominal bitrate.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
/** Determine if constrained VBR is enabled in the encoder.
* @see OPUS_SET_VBR_CONSTRAINT
* @see OPUS_GET_VBR
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Unconstrained VBR.</dd>
* <dt>1</dt><dd>Constrained VBR (default).</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
/** Configures mono/stereo forcing in the encoder.
* This can force the encoder to produce packets encoded as either mono or
* stereo, regardless of the format of the input audio. This is useful when
* the caller knows that the input signal is currently a mono source embedded
* in a stereo stream.
* @see OPUS_GET_FORCE_CHANNELS
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
* <dt>1</dt> <dd>Forced mono</dd>
* <dt>2</dt> <dd>Forced stereo</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
/** Gets the encoder's forced channel configuration.
* @see OPUS_SET_FORCE_CHANNELS
* @param[out] x <tt>opus_int32 *</tt>:
* <dl>
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
* <dt>1</dt> <dd>Forced mono</dd>
* <dt>2</dt> <dd>Forced stereo</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
/** Configures the maximum bandpass that the encoder will select automatically.
* Applications should normally use this instead of #OPUS_SET_BANDWIDTH
* (leaving that set to the default, #OPUS_AUTO). This allows the
* application to set an upper bound based on the type of input it is
* providing, but still gives the encoder the freedom to reduce the bandpass
* when the bitrate becomes too low, for better overall quality.
* @see OPUS_GET_MAX_BANDWIDTH
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured maximum allowed bandpass.
* @see OPUS_SET_MAX_BANDWIDTH
* @param[out] x <tt>opus_int32 *</tt>: Allowed values:
* <dl>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
/** Sets the encoder's bandpass to a specific value.
* This prevents the encoder from automatically selecting the bandpass based
* on the available bitrate. If an application knows the bandpass of the input
* audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
* instead, which still gives the encoder the freedom to reduce the bandpass
* when the bitrate becomes too low, for better overall quality.
* @see OPUS_GET_BANDWIDTH
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
/** Configures the type of signal being encoded.
* This is a hint which helps the encoder's mode selection.
* @see OPUS_GET_SIGNAL
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured signal type.
* @see OPUS_SET_SIGNAL
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's intended application.
* The initial value is a mandatory argument to the encoder_create function.
* @see OPUS_GET_APPLICATION
* @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured application.
* @see OPUS_SET_APPLICATION
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
/** Gets the total samples of delay added by the entire codec.
* This can be queried by the encoder and then the provided number of samples can be
* skipped on from the start of the decoder's output to provide time aligned input
* and output. From the perspective of a decoding application the real data begins this many
* samples late.
*
* The decoder contribution to this delay is identical for all decoders, but the
* encoder portion of the delay may vary from implementation to implementation,
* version to version, or even depend on the encoder's initial configuration.
* Applications needing delay compensation should call this CTL rather than
* hard-coding a value.
* @param[out] x <tt>opus_int32 *</tt>: Number of lookahead samples
* @hideinitializer */
#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of inband forward error correction (FEC).
* @note This is only applicable to the LPC layer
* @see OPUS_GET_INBAND_FEC
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable inband FEC (default).</dd>
* <dt>1</dt><dd>Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.</dd>
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
/** Gets encoder's configured use of inband forward error correction.
* @see OPUS_SET_INBAND_FEC
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Inband FEC disabled (default).</dd>
* <dt>1</dt><dd>Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.</dd>
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's expected packet loss percentage.
* Higher values trigger progressively more loss resistant behavior in the encoder
* at the expense of quality at a given bitrate in the absence of packet loss, but
* greater quality under loss.
* @see OPUS_GET_PACKET_LOSS_PERC
* @param[in] x <tt>opus_int32</tt>: Loss percentage in the range 0-100, inclusive (default: 0).
* @hideinitializer */
#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured packet loss percentage.
* @see OPUS_SET_PACKET_LOSS_PERC
* @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
* in the range 0-100, inclusive (default: 0).
* @hideinitializer */
#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of discontinuous transmission (DTX).
* @note This is only applicable to the LPC layer
* @see OPUS_GET_DTX
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable DTX (default).</dd>
* <dt>1</dt><dd>Enabled DTX.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
/** Gets encoder's configured use of discontinuous transmission.
* @see OPUS_SET_DTX
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>DTX disabled (default).</dd>
* <dt>1</dt><dd>DTX enabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
/** Configures the depth of signal being encoded.
*
* This is a hint which helps the encoder identify silence and near-silence.
* It represents the number of significant bits of linear intensity below
* which the signal contains ignorable quantization or other noise.
*
* For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
* for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
* for 16-bit linear pcm input with opus_encode_float().
*
* When using opus_encode() instead of opus_encode_float(), or when libopus
* is compiled for fixed-point, the encoder uses the minimum of the value
* set here and the value 16.
*
* @see OPUS_GET_LSB_DEPTH
* @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
* (default: 24).
* @hideinitializer */
#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured signal depth.
* @see OPUS_SET_LSB_DEPTH
* @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
* 24 (default: 24).
* @hideinitializer */
#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of variable duration frames.
* When variable duration is enabled, the encoder is free to use a shorter frame
* size than the one requested in the opus_encode*() call.
* It is then the user's responsibility
* to verify how much audio was encoded by checking the ToC byte of the encoded
* packet. The part of the audio that was not encoded needs to be resent to the
* encoder for the next call. Do not use this option unless you <b>really</b>
* know what you are doing.
* @see OPUS_GET_EXPERT_FRAME_DURATION
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured use of variable duration frames.
* @see OPUS_SET_EXPERT_FRAME_DURATION
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
/** If set to 1, disables almost all use of prediction, making frames almost
* completely independent. This reduces quality.
* @see OPUS_GET_PREDICTION_DISABLED
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Enable prediction (default).</dd>
* <dt>1</dt><dd>Disable prediction.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured prediction status.
* @see OPUS_SET_PREDICTION_DISABLED
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Prediction enabled (default).</dd>
* <dt>1</dt><dd>Prediction disabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_genericctls Generic CTLs
*
* These macros are used with the \c opus_decoder_ctl and
* \c opus_encoder_ctl calls to generate a particular
* request.
*
* When called on an \c OpusDecoder they apply to that
* particular decoder instance. When called on an
* \c OpusEncoder they apply to the corresponding setting
* on that encoder instance, if present.
*
* Some usage examples:
*
* @code
* int ret;
* opus_int32 pitch;
* ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
* if (ret == OPUS_OK) return ret;
*
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
* opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
*
* opus_int32 enc_bw, dec_bw;
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
* opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
* if (enc_bw != dec_bw) {
* printf("packet bandwidth mismatch!\n");
* }
* @endcode
*
* @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
* @{
*/
/** Resets the codec state to be equivalent to a freshly initialized state.
* This should be called when switching streams in order to prevent
* the back to back decoding from giving different results from
* one at a time decoding.
* @hideinitializer */
#define OPUS_RESET_STATE 4028
/** Gets the final state of the codec's entropy coder.
* This is used for testing purposes,
* The encoder and decoder state should be identical after coding a payload
* (assuming no data corruption or software bugs)
*
* @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
*
* @hideinitializer */
#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
/** Gets the encoder's configured bandpass or the decoder's last bandpass.
* @see OPUS_SET_BANDWIDTH
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
/** Gets the sampling rate the encoder or decoder was initialized with.
* This simply returns the <code>Fs</code> value passed to opus_encoder_init()
* or opus_decoder_init().
* @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
* @hideinitializer
*/
#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
/** If set to 1, disables the use of phase inversion for intensity stereo,
* improving the quality of mono downmixes, but slightly reducing normal
* stereo quality. Disabling phase inversion in the decoder does not comply
* with RFC 6716, although it does not cause any interoperability issue and
* is expected to become part of the Opus standard once RFC 6716 is updated
* by draft-ietf-codec-opus-update.
* @see OPUS_GET_PHASE_INVERSION_DISABLED
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Enable phase inversion (default).</dd>
* <dt>1</dt><dd>Disable phase inversion.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured phase inversion status.
* @see OPUS_SET_PHASE_INVERSION_DISABLED
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
* <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/** Gets the DTX state of the encoder.
* Returns whether the last encoded frame was either a comfort noise update
* during DTX or not encoded because of DTX.
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>The encoder is not in DTX.</dd>
* <dt>1</dt><dd>The encoder is in DTX.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_decoderctls Decoder related CTLs
* @see opus_genericctls, opus_encoderctls, opus_decoder
* @{
*/
/** Configures decoder gain adjustment.
* Scales the decoded output by a factor specified in Q8 dB units.
* This has a maximum range of -32768 to 32767 inclusive, and returns
* OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
* This setting survives decoder reset.
*
* gain = pow(10, x/(20.0*256))
*
* @param[in] x <tt>opus_int32</tt>: Amount to scale PCM signal by in Q8 dB units.
* @hideinitializer */
#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
*
* @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
* @hideinitializer */
#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
* @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
* @hideinitializer */
#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
/** Gets the pitch of the last decoded frame, if available.
* This can be used for any post-processing algorithm requiring the use of pitch,
* e.g. time stretching/shortening. If the last frame was not voiced, or if the
* pitch was not coded in the frame, then zero is returned.
*
* This CTL is only implemented for decoder instances.
*
* @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
*
* @hideinitializer */
#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_libinfo Opus library information functions
* @{
*/
/** Converts an opus error code into a human readable string.
*
* @param[in] error <tt>int</tt>: Error number
* @returns Error string
*/
OPUS_EXPORT const char *opus_strerror(int error);
/** Gets the libopus version string.
*
* Applications may look for the substring "-fixed" in the version string to
* determine whether they have a fixed-point or floating-point build at
* runtime.
*
* @returns Version string
*/
OPUS_EXPORT const char *opus_get_version_string(void);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_DEFINES_H */

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/* Copyright (c) 2011 Xiph.Org Foundation
Written by Jean-Marc Valin */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- 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 COPYRIGHT HOLDERS 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 COPYRIGHT OWNER
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.
*/
/**
* @file opus_multistream.h
* @brief Opus reference implementation multistream API
*/
#ifndef OPUS_MULTISTREAM_H
#define OPUS_MULTISTREAM_H
#include "opus.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @cond OPUS_INTERNAL_DOC */
/** Macros to trigger compilation errors when the wrong types are provided to a
* CTL. */
/**@{*/
#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
/**@}*/
/** These are the actual encoder and decoder CTL ID numbers.
* They should not be used directly by applications.
* In general, SETs should be even and GETs should be odd.*/
/**@{*/
#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
/**@}*/
/** @endcond */
/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
*
* These are convenience macros that are specific to the
* opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
* interface.
* The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
* @ref opus_decoderctls may be applied to a multistream encoder or decoder as
* well.
* In addition, you may retrieve the encoder or decoder state for an specific
* stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
* #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
*/
/**@{*/
/** Gets the encoder state for an individual stream of a multistream encoder.
* @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
* wish to retrieve.
* This must be non-negative and less than
* the <code>streams</code> parameter used
* to initialize the encoder.
* @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
* encoder state.
* @retval OPUS_BAD_ARG The index of the requested stream was out of range.
* @hideinitializer
*/
#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
/** Gets the decoder state for an individual stream of a multistream decoder.
* @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
* wish to retrieve.
* This must be non-negative and less than
* the <code>streams</code> parameter used
* to initialize the decoder.
* @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
* decoder state.
* @retval OPUS_BAD_ARG The index of the requested stream was out of range.
* @hideinitializer
*/
#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
/**@}*/
/** @defgroup opus_multistream Opus Multistream API
* @{
*
* The multistream API allows individual Opus streams to be combined into a
* single packet, enabling support for up to 255 channels. Unlike an
* elementary Opus stream, the encoder and decoder must negotiate the channel
* configuration before the decoder can successfully interpret the data in the
* packets produced by the encoder. Some basic information, such as packet
* duration, can be computed without any special negotiation.
*
* The format for multistream Opus packets is defined in
* <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
* and is based on the self-delimited Opus framing described in Appendix B of
* <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
* Normal Opus packets are just a degenerate case of multistream Opus packets,
* and can be encoded or decoded with the multistream API by setting
* <code>streams</code> to <code>1</code> when initializing the encoder or
* decoder.
*
* Multistream Opus streams can contain up to 255 elementary Opus streams.
* These may be either "uncoupled" or "coupled", indicating that the decoder
* is configured to decode them to either 1 or 2 channels, respectively.
* The streams are ordered so that all coupled streams appear at the
* beginning.
*
* A <code>mapping</code> table defines which decoded channel <code>i</code>
* should be used for each input/output (I/O) channel <code>j</code>. This table is
* typically provided as an unsigned char array.
* Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
* If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
* encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
* is even, or as the right channel of stream <code>(i/2)</code> if
* <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
* mono in stream <code>(i - coupled_streams)</code>, unless it has the special
* value 255, in which case it is omitted from the encoding entirely (the
* decoder will reproduce it as silence). Each value <code>i</code> must either
* be the special value 255 or be less than <code>streams + coupled_streams</code>.
*
* The output channels specified by the encoder
* should use the
* <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
* channel ordering</a>. A decoder may wish to apply an additional permutation
* to the mapping the encoder used to achieve a different output channel
* order (e.g. for outputing in WAV order).
*
* Each multistream packet contains an Opus packet for each stream, and all of
* the Opus packets in a single multistream packet must have the same
* duration. Therefore the duration of a multistream packet can be extracted
* from the TOC sequence of the first stream, which is located at the
* beginning of the packet, just like an elementary Opus stream:
*
* @code
* int nb_samples;
* int nb_frames;
* nb_frames = opus_packet_get_nb_frames(data, len);
* if (nb_frames < 1)
* return nb_frames;
* nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
* @endcode
*
* The general encoding and decoding process proceeds exactly the same as in
* the normal @ref opus_encoder and @ref opus_decoder APIs.
* See their documentation for an overview of how to use the corresponding
* multistream functions.
*/
/** Opus multistream encoder state.
* This contains the complete state of a multistream Opus encoder.
* It is position independent and can be freely copied.
* @see opus_multistream_encoder_create
* @see opus_multistream_encoder_init
*/
typedef struct OpusMSEncoder OpusMSEncoder;
/** Opus multistream decoder state.
* This contains the complete state of a multistream Opus decoder.
* It is position independent and can be freely copied.
* @see opus_multistream_decoder_create
* @see opus_multistream_decoder_init
*/
typedef struct OpusMSDecoder OpusMSDecoder;
/**\name Multistream encoder functions */
/**@{*/
/** Gets the size of an OpusMSEncoder structure.
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
int streams,
int coupled_streams
);
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
int channels,
int mapping_family
);
/** Allocates and initializes a multistream encoder state.
* Call opus_multistream_encoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* encoded channels to input channels, as described in
* @ref opus_multistream. As an extra constraint, the
* multistream encoder does not allow encoding coupled
* streams for which one channel is unused since this
* is never a good idea.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int application,
int *error
) OPUS_ARG_NONNULL(5);
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
unsigned char *mapping,
int application,
int *error
) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
/** Initialize a previously allocated multistream encoder state.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_multistream_encoder_create
* @see opus_multistream_encoder_get_size
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* encoded channels to input channels, as described in
* @ref opus_multistream. As an extra constraint, the
* multistream encoder does not allow encoding coupled
* streams for which one channel is unused since this
* is never a good idea.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_multistream_encoder_init(
OpusMSEncoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
OPUS_EXPORT int opus_multistream_surround_encoder_init(
OpusMSEncoder *st,
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
unsigned char *mapping,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
/** Encodes a multistream Opus frame.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
* samples.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
OpusMSEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a multistream Opus frame from floating point input.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
* samples with a normal range of
* +/-1.0.
* Samples with a range beyond +/-1.0
* are supported but will be clipped by
* decoders using the integer API and
* should only be used if it is known
* that the far end supports extended
* dynamic range.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
OpusMSEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusMSEncoder</code> allocated by
* opus_multistream_encoder_create().
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
*/
OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
/** Perform a CTL function on a multistream Opus encoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_encoderctls, or @ref opus_multistream_ctls.
* @see opus_genericctls
* @see opus_encoderctls
* @see opus_multistream_ctls
*/
OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/**\name Multistream decoder functions */
/**@{*/
/** Gets the size of an <code>OpusMSDecoder</code> structure.
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
int streams,
int coupled_streams
);
/** Allocates and initializes a multistream decoder state.
* Call opus_multistream_decoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* coded channels to output channels, as described in
* @ref opus_multistream.
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_multistream_decoder_create
* @see opus_multistream_deocder_get_size
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* coded channels to output channels, as described in
* @ref opus_multistream.
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_multistream_decoder_init(
OpusMSDecoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
/** Decode a multistream Opus packet.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a multistream Opus packet with floating point output.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on a multistream Opus decoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_decoderctls, or @ref opus_multistream_ctls.
* @see opus_genericctls
* @see opus_decoderctls
* @see opus_multistream_ctls
*/
OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusMSDecoder</code> allocated by
* opus_multistream_decoder_create().
* @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
*/
OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
/**@}*/
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_MULTISTREAM_H */

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@@ -0,0 +1,568 @@
/* Copyright (c) 2017 Google Inc.
Written by Andrew Allen */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- 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 COPYRIGHT HOLDERS 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 COPYRIGHT OWNER
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.
*/
/**
* @file opus_projection.h
* @brief Opus projection reference API
*/
#ifndef OPUS_PROJECTION_H
#define OPUS_PROJECTION_H
#include "opus_multistream.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @cond OPUS_INTERNAL_DOC */
/** These are the actual encoder and decoder CTL ID numbers.
* They should not be used directly by applications.c
* In general, SETs should be even and GETs should be odd.*/
/**@{*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST 6001
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST 6003
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST 6005
/**@}*/
/** @endcond */
/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
*
* These are convenience macros that are specific to the
* opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
* interface.
* The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
* @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
* projection encoder or decoder as well.
*/
/**@{*/
/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
* @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
* of the demixing matrix.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
/** Gets the size in bytes of the demixing matrix from the encoder.
* @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
* demixing matrix.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
/** Copies the demixing matrix to the supplied pointer location.
* @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
* supplied pointer location.
* @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
* pointer location.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
/**@}*/
/** Opus projection encoder state.
* This contains the complete state of a projection Opus encoder.
* It is position independent and can be freely copied.
* @see opus_projection_ambisonics_encoder_create
*/
typedef struct OpusProjectionEncoder OpusProjectionEncoder;
/** Opus projection decoder state.
* This contains the complete state of a projection Opus decoder.
* It is position independent and can be freely copied.
* @see opus_projection_decoder_create
* @see opus_projection_decoder_init
*/
typedef struct OpusProjectionDecoder OpusProjectionDecoder;
/**\name Projection encoder functions */
/**@{*/
/** Gets the size of an OpusProjectionEncoder structure.
* @param channels <tt>int</tt>: The total number of input channels to encode.
* This must be no more than 255.
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
* the appropriate projection.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
int channels,
int mapping_family
);
/** Allocates and initializes a projection encoder state.
* Call opus_projection_encoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
* the appropriate projection.
* @param[out] streams <tt>int *</tt>: The total number of streams that will
* be encoded from the input.
* @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
* streams that will be encoded from the input.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
int application,
int *error
) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
/** Initialize a previously allocated projection encoder state.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_ambisonics_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_projection_ambisonics_encoder_create
* @see opus_projection_ambisonics_encoder_get_size
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
OpusProjectionEncoder *st,
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
/** Encodes a projection Opus frame.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
* samples.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
OpusProjectionEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a projection Opus frame from floating point input.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
* samples with a normal range of
* +/-1.0.
* Samples with a range beyond +/-1.0
* are supported but will be clipped by
* decoders using the integer API and
* should only be used if it is known
* that the far end supports extended
* dynamic range.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
OpusProjectionEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusProjectionEncoder</code> allocated by
* opus_projection_ambisonics_encoder_create().
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
*/
OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
/** Perform a CTL function on a projection Opus encoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_encoderctls, @ref opus_multistream_ctls, or
* @ref opus_projection_ctls
* @see opus_genericctls
* @see opus_encoderctls
* @see opus_multistream_ctls
* @see opus_projection_ctls
*/
OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/**\name Projection decoder functions */
/**@{*/
/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
* @param channels <tt>int</tt>: The total number of output channels.
* This must be no more than 255.
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
int channels,
int streams,
int coupled_streams
);
/** Allocates and initializes a projection decoder state.
* Call opus_projection_decoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
* that mapping from coded channels to output channels,
* as described in @ref opus_projection and
* @ref opus_projection_ctls.
* @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
* demixing matrix, as
* described in @ref
* opus_projection_ctls.
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
unsigned char *demixing_matrix,
opus_int32 demixing_matrix_size,
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated projection decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_decoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_projection_decoder_create
* @see opus_projection_deocder_get_size
* @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
* that mapping from coded channels to output channels,
* as described in @ref opus_projection and
* @ref opus_projection_ctls.
* @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
* demixing matrix, as
* described in @ref
* opus_projection_ctls.
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_projection_decoder_init(
OpusProjectionDecoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
unsigned char *demixing_matrix,
opus_int32 demixing_matrix_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
/** Decode a projection Opus packet.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a projection Opus packet with floating point output.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on a projection Opus decoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_decoderctls, @ref opus_multistream_ctls, or
* @ref opus_projection_ctls.
* @see opus_genericctls
* @see opus_decoderctls
* @see opus_multistream_ctls
* @see opus_projection_ctls
*/
OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusProjectionDecoder</code> allocated by
* opus_projection_decoder_create().
* @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
*/
OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
/**@}*/
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_PROJECTION_H */

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/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
/* Modified by Jean-Marc Valin */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- 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 COPYRIGHT HOLDERS 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 COPYRIGHT OWNER
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.
*/
/* opus_types.h based on ogg_types.h from libogg */
/**
@file opus_types.h
@brief Opus reference implementation types
*/
#ifndef OPUS_TYPES_H
#define OPUS_TYPES_H
#define opus_int int /* used for counters etc; at least 16 bits */
#define opus_int64 long long
#define opus_int8 signed char
#define opus_uint unsigned int /* used for counters etc; at least 16 bits */
#define opus_uint64 unsigned long long
#define opus_uint8 unsigned char
/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
#include <stdint.h>
# undef opus_int64
# undef opus_int8
# undef opus_uint64
# undef opus_uint8
typedef int8_t opus_int8;
typedef uint8_t opus_uint8;
typedef int16_t opus_int16;
typedef uint16_t opus_uint16;
typedef int32_t opus_int32;
typedef uint32_t opus_uint32;
typedef int64_t opus_int64;
typedef uint64_t opus_uint64;
#elif defined(_WIN32)
# if defined(__CYGWIN__)
# include <_G_config.h>
typedef _G_int32_t opus_int32;
typedef _G_uint32_t opus_uint32;
typedef _G_int16 opus_int16;
typedef _G_uint16 opus_uint16;
# elif defined(__MINGW32__)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
# elif defined(__MWERKS__)
typedef int opus_int32;
typedef unsigned int opus_uint32;
typedef short opus_int16;
typedef unsigned short opus_uint16;
# else
/* MSVC/Borland */
typedef __int32 opus_int32;
typedef unsigned __int32 opus_uint32;
typedef __int16 opus_int16;
typedef unsigned __int16 opus_uint16;
# endif
#elif defined(__MACOS__)
# include <sys/types.h>
typedef SInt16 opus_int16;
typedef UInt16 opus_uint16;
typedef SInt32 opus_int32;
typedef UInt32 opus_uint32;
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <sys/types.h>
typedef int16_t opus_int16;
typedef u_int16_t opus_uint16;
typedef int32_t opus_int32;
typedef u_int32_t opus_uint32;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16 opus_int16;
typedef u_int16 opus_uint16;
typedef int32_t opus_int32;
typedef u_int32_t opus_uint32;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#elif defined (DJGPP)
/* DJGPP */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#elif defined(R5900)
/* PS2 EE */
typedef int opus_int32;
typedef unsigned opus_uint32;
typedef short opus_int16;
typedef unsigned short opus_uint16;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short opus_int16;
typedef unsigned short opus_uint16;
typedef signed int opus_int32;
typedef unsigned int opus_uint32;
#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef long opus_int32;
typedef unsigned long opus_uint32;
#elif defined(CONFIG_TI_C6X)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#else
/* Give up, take a reasonable guess */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#endif
#endif /* OPUS_TYPES_H */

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lib/codecs/inc/resample16/resample16.h Normal file
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/*
* FILE: resample16.h
*
* The configuration constants below govern
* the number of bits in the input sample and filter coefficients, the
* number of bits to the right of the binary-point for fixed-point math, etc.
*
*/
/* Conversion constants */
#define Nhc 8
#define Na 7
#define Np (Nhc+Na)
#define Npc (1<<Nhc)
#define Amask ((1<<Na)-1)
#define Pmask ((1<<Np)-1)
#define Nh 16
#define Nb 16
#define Nhxn 14
#define Nhg (Nh-Nhxn)
#define NLpScl 13
/* Description of constants:
*
* Npc - is the number of look-up values available for the lowpass filter
* between the beginning of its impulse response and the "cutoff time"
* of the filter. The cutoff time is defined as the reciprocal of the
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Npc would be the index of the
* impulse-response lookup-table corresponding to the first zero-
* crossing of the sinc function. (The inverse first zero-crossing
* time of a sinc function equals its nominal cutoff frequency in Hz.)
* Npc must be a power of 2 due to the details of the current
* implementation. The default value of 512 is sufficiently high that
* using linear interpolation to fill in between the table entries
* gives approximately 16-bit accuracy in filter coefficients.
*
* Nhc - is log base 2 of Npc.
*
* Na - is the number of bits devoted to linear interpolation of the
* filter coefficients.
*
* Np - is Na + Nhc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. Np must be less than 16 in
* this implementation.
*
* Nh - is the number of bits in the filter coefficients. The sum of Nh and
* the number of bits in the input data (typically 16) cannot exceed 32.
* Thus Nh should be 16. The largest filter coefficient should nearly
* fill 16 bits (32767).
*
* Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
* exceed 32.
*
* Nhxn - is the number of bits to right shift after multiplying each input
* sample times a filter coefficient. It can be as great as Nh and as
* small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
* accumulation. If Nhxn=0, the accumulation will soon overflow 32 bits.
*
* Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
*
* NLpScl - is the number of bits allocated to the unity-gain normalization
* factor. The output of the lowpass filter is multiplied by LpScl and
* then right-shifted NLpScl bits. To avoid overflow, we must have
* Nb+Nhg+NLpScl < 32.
*/
typedef char BOOL;
typedef short HWORD;
typedef unsigned short UHWORD;
typedef int WORD;
typedef unsigned int UWORD;
struct resample16_s;
typedef struct {
UHWORD LpScl; /* Unity-gain scale factor */
UHWORD Nwing; /* Filter table size */
UHWORD Nmult; /* Filter length for up-conversions */
const HWORD *Imp; /* Filter coefficients */
const HWORD *ImpD; /* ImpD[n] = Imp[n+1]-Imp[n] */
} resample16_filter_t;
typedef enum { RESAMPLE16_BASIC, RESAMPLE16_LOW, RESAMPLE16_MED } resample16_filter_e;
WORD resample16(struct resample16_s *r, HWORD X[], int inCount, HWORD Y[]);
struct resample16_s* resample16_create(float factor, resample16_filter_e filter, resample16_filter_t *custom, BOOL interp);
void resample16_delete(struct resample16_s *r);
void resample16_flush(struct resample16_s *r);

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lib/codecs/inc/soxr/soxr.h Normal file
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/* SoX Resampler Library Copyright (c) 2007-13 robs@users.sourceforge.net
*
* This library is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or (at
* your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
* General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this library; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* -------------------------------- Gubbins --------------------------------- */
#if !defined soxr_included
#define soxr_included
#if defined __cplusplus
#include <cstddef>
extern "C" {
#else
#include <stddef.h>
#endif
#if defined SOXR_DLL
#if defined soxr_EXPORTS
#define SOXR __declspec(dllexport)
#else
#define SOXR __declspec(dllimport)
#endif
#elif defined SOXR_VISIBILITY && defined __GNUC__ && (__GNUC__ > 4 || __GNUC__ == 4 && __GNUC_MINOR__ >= 1)
#define SOXR __attribute__ ((visibility("default")))
#else
#define SOXR
#endif
typedef struct soxr_io_spec soxr_io_spec_t;
typedef struct soxr_quality_spec soxr_quality_spec_t;
typedef struct soxr_runtime_spec soxr_runtime_spec_t;
/* ---------------------------- API conventions --------------------------------
Buffer lengths (and occupancies) are expressed as the number of contained
samples per channel.
Parameter names for buffer lengths have the suffix `len'.
A single-character `i' or 'o' is often used in names to give context as
input or output (e.g. ilen, olen). */
/* --------------------------- Version management --------------------------- */
/* E.g. #if SOXR_THIS_VERSION >= SOXR_VERSION(0,1,1) ... */
#define SOXR_VERSION(x,y,z) (((x)<<16)|((y)<<8)|(z))
#define SOXR_THIS_VERSION SOXR_VERSION(0,1,2)
#define SOXR_THIS_VERSION_STR "0.1.2"
/* --------------------------- Type declarations ---------------------------- */
typedef struct soxr * soxr_t; /* A resampler for 1 or more channels. */
typedef char const * soxr_error_t; /* 0:no-error; non-0:error. */
typedef void * soxr_buf_t; /* 1 buffer of channel-interleaved samples. */
typedef void const * soxr_cbuf_t; /* Ditto; read-only. */
typedef soxr_buf_t const * soxr_bufs_t;/* Or, a separate buffer for each ch. */
typedef soxr_cbuf_t const * soxr_cbufs_t; /* Ditto; read-only. */
typedef void const * soxr_in_t; /* Either a soxr_cbuf_t or soxr_cbufs_t,
depending on itype in soxr_io_spec_t. */
typedef void * soxr_out_t; /* Either a soxr_buf_t or soxr_bufs_t,
depending on otype in soxr_io_spec_t. */
/* --------------------------- API main functions --------------------------- */
SOXR char const * soxr_version(void); /* Query library version: "libsoxr-x.y.z" */
#define soxr_strerror(e) /* Soxr counterpart to strerror. */ \
((e)?(e):"no error")
/* Create a stream resampler: */
SOXR soxr_t soxr_create(
double input_rate, /* Input sample-rate. */
double output_rate, /* Output sample-rate. */
unsigned num_channels, /* Number of channels to be used. */
/* All following arguments are optional (may be set to NULL). */
soxr_error_t *, /* To report any error during creation. */
soxr_io_spec_t const *, /* To specify non-default I/O formats. */
soxr_quality_spec_t const *, /* To specify non-default resampling quality.*/
soxr_runtime_spec_t const *);/* To specify non-default runtime resources.
Default io_spec is per soxr_io_spec(SOXR_FLOAT32_I, SOXR_FLOAT32_I)
Default quality_spec is per soxr_quality_spec(SOXR_HQ, 0)
Default runtime_spec is per soxr_runtime_spec(1) */
/* If not using an app-supplied input function, after creating a stream
* resampler, repeatedly call: */
SOXR soxr_error_t soxr_process(
soxr_t resampler, /* As returned by soxr_create. */
/* Input (to be resampled): */
soxr_in_t in, /* Input buffer(s); may be NULL (see below). */
size_t ilen, /* Input buf. length (samples per channel). */
size_t * idone, /* To return actual # samples used (<= ilen). */
/* Output (resampled): */
soxr_out_t out, /* Output buffer(s).*/
size_t olen, /* Output buf. length (samples per channel). */
size_t * odone); /* To return actual # samples out (<= olen).
Note that no special meaning is associated with ilen or olen equal to
zero. End-of-input (i.e. no data is available nor shall be available)
may be indicated by seting `in' to NULL. */
/* If using an app-supplied input function, it must look and behave like this:*/
typedef size_t /* data_len */
(* soxr_input_fn_t)( /* Supply data to be resampled. */
void * input_fn_state, /* As given to soxr_set_input_fn (below). */
soxr_in_t * data, /* Returned data; see below. N.B. ptr to ptr(s)*/
size_t requested_len); /* Samples per channel, >= returned data_len.
data_len *data Indicates Meaning
------- ------- ------------ -------------------------
!=0 !=0 Success *data contains data to be
input to the resampler.
0 !=0 (or End-of-input No data is available nor
not set) shall be available.
0 0 Failure An error occurred whilst trying to
source data to be input to the resampler. */
/* and be registered with a previously created stream resampler using: */
SOXR soxr_error_t soxr_set_input_fn(/* Set (or reset) an input function.*/
soxr_t resampler, /* As returned by soxr_create. */
soxr_input_fn_t, /* Function to supply data to be resampled.*/
void * input_fn_state, /* If needed by the input function. */
size_t max_ilen); /* Maximum value for input fn. requested_len.*/
/* then repeatedly call: */
SOXR size_t /*odone*/ soxr_output(/* Resample and output a block of data.*/
soxr_t resampler, /* As returned by soxr_create. */
soxr_out_t data, /* App-supplied buffer(s) for resampled data.*/
size_t olen); /* Amount of data to output; >= odone. */
/* Common stream resampler operations: */
SOXR soxr_error_t soxr_error(soxr_t); /* Query error status. */
SOXR size_t * soxr_num_clips(soxr_t); /* Query int. clip counter (for R/W). */
SOXR double soxr_delay(soxr_t); /* Query current delay in output samples.*/
SOXR char const * soxr_engine(soxr_t p); /* Query resampling engine name. */
SOXR soxr_error_t soxr_clear(soxr_t); /* Ready for fresh signal, same config. */
SOXR void soxr_delete(soxr_t); /* Free resources. */
/* `Short-cut', single call to resample a (probably short) signal held entirely
* in memory. See soxr_create and soxr_process above for parameter details.
* Note that unlike soxr_create however, the default quality spec. for
* soxr_oneshot is per soxr_quality_spec(SOXR_LQ, 0). */
SOXR soxr_error_t soxr_oneshot(
double input_rate,
double output_rate,
unsigned num_channels,
soxr_in_t in , size_t ilen, size_t * idone,
soxr_out_t out, size_t olen, size_t * odone,
soxr_io_spec_t const *,
soxr_quality_spec_t const *,
soxr_runtime_spec_t const *);
/* For variable-rate resampling. See example # 5 for how to create a
* variable-rate resampler and how to use this function. */
SOXR soxr_error_t soxr_set_io_ratio(soxr_t, double io_ratio, size_t slew_len);
/* -------------------------- API type definitions -------------------------- */
typedef enum { /* Datatypes supported for I/O to/from the resampler: */
/* Internal; do not use: */
SOXR_FLOAT32, SOXR_FLOAT64, SOXR_INT32, SOXR_INT16, SOXR_SPLIT = 4,
/* Use for interleaved channels: */
SOXR_FLOAT32_I = SOXR_FLOAT32, SOXR_FLOAT64_I, SOXR_INT32_I, SOXR_INT16_I,
/* Use for split channels: */
SOXR_FLOAT32_S = SOXR_SPLIT , SOXR_FLOAT64_S, SOXR_INT32_S, SOXR_INT16_S
} soxr_datatype_t;
#define soxr_datatype_size(x) /* Returns `sizeof' a soxr_datatype_t sample. */\
((unsigned char *)"\4\10\4\2")[(x)&3]
struct soxr_io_spec { /* Typically */
soxr_datatype_t itype; /* Input datatype. SOXR_FLOAT32_I */
soxr_datatype_t otype; /* Output datatype. SOXR_FLOAT32_I */
double scale; /* Linear gain to apply during resampling. 1 */
void * e; /* Reserved for internal use 0 */
unsigned long flags; /* Per the following #defines. 0 */
};
#define SOXR_TPDF 0 /* Applicable only if otype is INT16. */
#define SOXR_NO_DITHER 8u /* Disable the above. */
struct soxr_quality_spec { /* Typically */
double precision; /* Conversion precision (in bits). 20 */
double phase_response; /* 0=minimum, ... 50=linear, ... 100=maximum 50 */
double passband_end; /* 0dB pt. bandwidth to preserve; nyquist=1 0.913*/
double stopband_begin; /* Aliasing/imaging control; > passband_end 1 */
void * e; /* Reserved for internal use. 0 */
unsigned long flags; /* Per the following #defines. 0 */
};
#define SOXR_ROLLOFF_SMALL 0u /* <= 0.01 dB */
#define SOXR_ROLLOFF_MEDIUM 1u /* <= 0.35 dB */
#define SOXR_ROLLOFF_NONE 2u /* For Chebyshev bandwidth. */
#define SOXR_MAINTAIN_3DB_PT 4u /* Reserved for internal use. */
#define SOXR_HI_PREC_CLOCK 8u /* Increase `irrational' ratio accuracy. */
#define SOXR_DOUBLE_PRECISION 16u /* Use D.P. calcs even if precision <= 20. */
#define SOXR_VR 32u /* Variable-rate resampling. */
struct soxr_runtime_spec { /* Typically */
unsigned log2_min_dft_size;/* For DFT efficiency. [8,15] 10 */
unsigned log2_large_dft_size;/* For DFT efficiency. [16,20] 17 */
unsigned coef_size_kbytes; /* For SOXR_COEF_INTERP_AUTO (below). 400 */
unsigned num_threads; /* If built so. 0 means `automatic'. 1 */
void * e; /* Reserved for internal use. 0 */
unsigned long flags; /* Per the following #defines. 0 */
};
/* For `irrational' ratios only: */
#define SOXR_COEF_INTERP_AUTO 0u /* Auto select coef. interpolation. */
#define SOXR_COEF_INTERP_LOW 2u /* Man. select: less CPU, more memory. */
#define SOXR_COEF_INTERP_HIGH 3u /* Man. select: more CPU, less memory. */
#define SOXR_STRICT_BUFFERING 4u /* Reserved for future use. */
#define SOXR_NOSMALLINTOPT 8u /* For test purposes only. */
/* -------------------------- API type constructors ------------------------- */
/* These functions allow setting of the most commonly-used structure
* parameters, with other parameters being given default values. The default
* values may then be overridden, directly in the structure, if needed. */
SOXR soxr_quality_spec_t soxr_quality_spec(
unsigned long recipe, /* Per the #defines immediately below. */
unsigned long flags); /* As soxr_quality_spec_t.flags. */
/* The 5 standard qualities found in SoX: */
#define SOXR_QQ 0 /* 'Quick' cubic interpolation. */
#define SOXR_LQ 1 /* 'Low' 16-bit with larger rolloff. */
#define SOXR_MQ 2 /* 'Medium' 16-bit with medium rolloff. */
#define SOXR_HQ SOXR_20_BITQ /* 'High quality'. */
#define SOXR_VHQ SOXR_28_BITQ /* 'Very high quality'. */
#define SOXR_16_BITQ 3
#define SOXR_20_BITQ 4
#define SOXR_24_BITQ 5
#define SOXR_28_BITQ 6
#define SOXR_32_BITQ 7
/* Libsamplerate equivalent qualities: */
#define SOXR_LSR0Q 8 /* 'Best sinc'. */
#define SOXR_LSR1Q 9 /* 'Medium sinc'. */
#define SOXR_LSR2Q 10 /* 'Fast sinc'. */
#define SOXR_LINEAR_PHASE 0x00
#define SOXR_INTERMEDIATE_PHASE 0x10
#define SOXR_MINIMUM_PHASE 0x30
#define SOXR_STEEP_FILTER 0x40
#define SOXR_ALLOW_ALIASING 0x80 /* Reserved for future use. */
SOXR soxr_runtime_spec_t soxr_runtime_spec(
unsigned num_threads);
SOXR soxr_io_spec_t soxr_io_spec(
soxr_datatype_t itype,
soxr_datatype_t otype);
/* --------------------------- Advanced use only ---------------------------- */
/* For new designs, the following functions/usage will probably not be needed.
* They might be useful when adding soxr into an existing design where values
* for the resampling-rate and/or number-of-channels parameters to soxr_create
* are not available when that function will be called. In such cases, the
* relevant soxr_create parameter(s) can be given as 0, then one or both of the
* following (as appropriate) later invoked (but prior to calling soxr_process
* or soxr_output):
*
* soxr_set_error(soxr, soxr_set_io_ratio(soxr, io_ratio, 0));
* soxr_set_error(soxr, soxr_set_num_channels(soxr, num_channels));
*/
SOXR soxr_error_t soxr_set_error(soxr_t, soxr_error_t);
SOXR soxr_error_t soxr_set_num_channels(soxr_t, unsigned);
#undef SOXR
#if defined __cplusplus
}
#endif
#endif

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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis 'TREMOR' CODEC SOURCE CODE. *
* *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis 'TREMOR' SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
* BY THE Xiph.Org FOUNDATION http://www.xiph.org/ *
* *
********************************************************************
function: libvorbis codec headers
********************************************************************/
#ifndef _vorbis_codec_h_
#define _vorbis_codec_h_
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#include <ogg/ogg.h>
typedef struct vorbis_info{
int version;
int channels;
long rate;
/* The below bitrate declarations are *hints*.
Combinations of the three values carry the following implications:
all three set to the same value:
implies a fixed rate bitstream
only nominal set:
implies a VBR stream that averages the nominal bitrate. No hard
upper/lower limit
upper and or lower set:
implies a VBR bitstream that obeys the bitrate limits. nominal
may also be set to give a nominal rate.
none set:
the coder does not care to speculate.
*/
long bitrate_upper;
long bitrate_nominal;
long bitrate_lower;
long bitrate_window;
void *codec_setup;
} vorbis_info;
/* vorbis_dsp_state buffers the current vorbis audio
analysis/synthesis state. The DSP state belongs to a specific
logical bitstream ****************************************************/
typedef struct vorbis_dsp_state{
int analysisp;
vorbis_info *vi;
ogg_int32_t **pcm;
ogg_int32_t **pcmret;
int pcm_storage;
int pcm_current;
int pcm_returned;
int preextrapolate;
int eofflag;
long lW;
long W;
long nW;
long centerW;
ogg_int64_t granulepos;
ogg_int64_t sequence;
void *backend_state;
} vorbis_dsp_state;
typedef struct vorbis_block{
/* necessary stream state for linking to the framing abstraction */
ogg_int32_t **pcm; /* this is a pointer into local storage */
oggpack_buffer opb;
long lW;
long W;
long nW;
int pcmend;
int mode;
int eofflag;
ogg_int64_t granulepos;
ogg_int64_t sequence;
vorbis_dsp_state *vd; /* For read-only access of configuration */
/* local storage to avoid remallocing; it's up to the mapping to
structure it */
void *localstore;
long localtop;
long localalloc;
long totaluse;
struct alloc_chain *reap;
} vorbis_block;
/* vorbis_block is a single block of data to be processed as part of
the analysis/synthesis stream; it belongs to a specific logical
bitstream, but is independant from other vorbis_blocks belonging to
that logical bitstream. *************************************************/
struct alloc_chain{
void *ptr;
struct alloc_chain *next;
};
/* vorbis_info contains all the setup information specific to the
specific compression/decompression mode in progress (eg,
psychoacoustic settings, channel setup, options, codebook
etc). vorbis_info and substructures are in backends.h.
*********************************************************************/
/* the comments are not part of vorbis_info so that vorbis_info can be
static storage */
typedef struct vorbis_comment{
/* unlimited user comment fields. libvorbis writes 'libvorbis'
whatever vendor is set to in encode */
char **user_comments;
int *comment_lengths;
int comments;
char *vendor;
} vorbis_comment;
/* libvorbis encodes in two abstraction layers; first we perform DSP
and produce a packet (see docs/analysis.txt). The packet is then
coded into a framed OggSquish bitstream by the second layer (see
docs/framing.txt). Decode is the reverse process; we sync/frame
the bitstream and extract individual packets, then decode the
packet back into PCM audio.
The extra framing/packetizing is used in streaming formats, such as
files. Over the net (such as with UDP), the framing and
packetization aren't necessary as they're provided by the transport
and the streaming layer is not used */
/* Vorbis PRIMITIVES: general ***************************************/
extern void vorbis_info_init(vorbis_info *vi);
extern void vorbis_info_clear(vorbis_info *vi);
extern int vorbis_info_blocksize(vorbis_info *vi,int zo);
extern void vorbis_comment_init(vorbis_comment *vc);
extern void vorbis_comment_add(vorbis_comment *vc, char *comment);
extern void vorbis_comment_add_tag(vorbis_comment *vc,
char *tag, char *contents);
extern char *vorbis_comment_query(vorbis_comment *vc, char *tag, int count);
extern int vorbis_comment_query_count(vorbis_comment *vc, char *tag);
extern void vorbis_comment_clear(vorbis_comment *vc);
extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb);
extern int vorbis_block_clear(vorbis_block *vb);
extern void vorbis_dsp_clear(vorbis_dsp_state *v);
/* Vorbis PRIMITIVES: synthesis layer *******************************/
extern int vorbis_synthesis_idheader(ogg_packet *op);
extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,
ogg_packet *op);
extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi);
extern int vorbis_synthesis_restart(vorbis_dsp_state *v);
extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op);
extern int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op);
extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb);
extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,ogg_int32_t ***pcm);
extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples);
extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op);
/* Vorbis ERRORS and return codes ***********************************/
#define OV_FALSE -1
#define OV_EOF -2
#define OV_HOLE -3
#define OV_EREAD -128
#define OV_EFAULT -129
#define OV_EIMPL -130
#define OV_EINVAL -131
#define OV_ENOTVORBIS -132
#define OV_EBADHEADER -133
#define OV_EVERSION -134
#define OV_ENOTAUDIO -135
#define OV_EBADPACKET -136
#define OV_EBADLINK -137
#define OV_ENOSEEK -138
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif

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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis 'TREMOR' CODEC SOURCE CODE. *
* *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis 'TREMOR' SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
* BY THE Xiph.Org FOUNDATION http://www.xiph.org/ *
* *
********************************************************************
function: stdio-based convenience library for opening/seeking/decoding
********************************************************************/
#ifndef _OV_FILE_H_
#define _OV_FILE_H_
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#include <stdio.h>
#include "ivorbiscodec.h"
#define CHUNKSIZE 65535
#define READSIZE 1024
/* The function prototypes for the callbacks are basically the same as for
* the stdio functions fread, fseek, fclose, ftell.
* The one difference is that the FILE * arguments have been replaced with
* a void * - this is to be used as a pointer to whatever internal data these
* functions might need. In the stdio case, it's just a FILE * cast to a void *
*
* If you use other functions, check the docs for these functions and return
* the right values. For seek_func(), you *MUST* return -1 if the stream is
* unseekable
*/
typedef struct {
size_t (*read_func) (void *ptr, size_t size, size_t nmemb, void *datasource);
int (*seek_func) (void *datasource, ogg_int64_t offset, int whence);
int (*close_func) (void *datasource);
long (*tell_func) (void *datasource);
} ov_callbacks;
#define NOTOPEN 0
#define PARTOPEN 1
#define OPENED 2
#define STREAMSET 3
#define INITSET 4
typedef struct OggVorbis_File {
void *datasource; /* Pointer to a FILE *, etc. */
int seekable;
ogg_int64_t offset;
ogg_int64_t end;
ogg_sync_state oy;
/* If the FILE handle isn't seekable (eg, a pipe), only the current
stream appears */
int links;
ogg_int64_t *offsets;
ogg_int64_t *dataoffsets;
ogg_uint32_t *serialnos;
ogg_int64_t *pcmlengths;
vorbis_info *vi;
vorbis_comment *vc;
/* Decoding working state local storage */
ogg_int64_t pcm_offset;
int ready_state;
ogg_uint32_t current_serialno;
int current_link;
ogg_int64_t bittrack;
ogg_int64_t samptrack;
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
ov_callbacks callbacks;
} OggVorbis_File;
extern int ov_clear(OggVorbis_File *vf);
extern int ov_open(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes);
extern int ov_open_callbacks(void *datasource, OggVorbis_File *vf,
const char *initial, long ibytes, ov_callbacks callbacks);
extern int ov_test(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes);
extern int ov_test_callbacks(void *datasource, OggVorbis_File *vf,
const char *initial, long ibytes, ov_callbacks callbacks);
extern int ov_test_open(OggVorbis_File *vf);
extern long ov_bitrate(OggVorbis_File *vf,int i);
extern long ov_bitrate_instant(OggVorbis_File *vf);
extern long ov_streams(OggVorbis_File *vf);
extern long ov_seekable(OggVorbis_File *vf);
extern long ov_serialnumber(OggVorbis_File *vf,int i);
extern ogg_int64_t ov_raw_total(OggVorbis_File *vf,int i);
extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i);
extern ogg_int64_t ov_time_total(OggVorbis_File *vf,int i);
extern int ov_raw_seek(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_pcm_seek(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_pcm_seek_page(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_time_seek(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_time_seek_page(OggVorbis_File *vf,ogg_int64_t pos);
extern ogg_int64_t ov_raw_tell(OggVorbis_File *vf);
extern ogg_int64_t ov_pcm_tell(OggVorbis_File *vf);
extern ogg_int64_t ov_time_tell(OggVorbis_File *vf);
extern vorbis_info *ov_info(OggVorbis_File *vf,int link);
extern vorbis_comment *ov_comment(OggVorbis_File *vf,int link);
extern long ov_read(OggVorbis_File *vf,char *buffer,int length,
int *bitstream);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif

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