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Merge branch 'experiments/arm64-simulator'

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This commit is contained in:
Ali 2020-12-17 16:27:51 +04:00
commit 7a746fa6ec
58 changed files with 564 additions and 10832 deletions
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2
.gitmodules vendored
View File

@ -7,7 +7,7 @@
url=https://github.com/bazelbuild/rules_apple.git
[submodule "build-system/bazel-rules/rules_swift"]
path = build-system/bazel-rules/rules_swift
url = https://github.com/bazelbuild/rules_swift.git
url=https://github.com/ali-fareed/rules_swift.git
[submodule "build-system/bazel-rules/apple_support"]
path = build-system/bazel-rules/apple_support
url = https://github.com/bazelbuild/apple_support.git

10
Makefile
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@ -406,6 +406,16 @@ bazel_app_debug_arm64:
--watchos_cpus=armv7k,arm64_32 \
--verbose_failures
bazel_app_debug_sim_arm64:
APP_VERSION="${APP_VERSION}" \
TELEGRAM_DISABLE_EXTENSIONS="0" \
build-system/prepare-build.sh Telegram distribution
"${BAZEL}" build Telegram/Telegram ${BAZEL_CACHE_FLAGS} ${BAZEL_COMMON_FLAGS} ${BAZEL_DEBUG_FLAGS} \
-c dbg \
--ios_multi_cpus=sim_arm64 \
--watchos_cpus=armv7k,arm64_32 \
--verbose_failures
bazel_app_arm64:
APP_VERSION="${APP_VERSION}" \
BAZEL_CACHE_DIR="${BAZEL_CACHE_DIR}" \

4
WORKSPACE
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@ -53,6 +53,6 @@ bazel_skylib_workspace()
http_file(
name = "cmake_tar_gz",
urls = ["https://github.com/Kitware/CMake/releases/download/v3.18.4/cmake-3.18.4-Darwin-x86_64.tar.gz"],
sha256 = "9d27049660474cf134ab46fa0e0db771b263313fcb8ba82ee8b2d1a1a62f8f20",
urls = ["https://github.com/Kitware/CMake/releases/download/v3.19.2/cmake-3.19.2-macos-universal.tar.gz"],
sha256 = "50afa2cb66bea6a0314ef28034f3ff1647325e30cf5940f97906a56fd9640bd8",
)

5
build-system/BUILD
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@ -0,0 +1,5 @@
config_setting(
name = "ios_sim_arm64",
values = {"cpu": "ios_sim_arm64"},
)

2
build-system/bazel-rules/rules_swift

@ -1 +1 @@
Subproject commit eca50db5a95a50b2be8367d644aebf4d1d880c52
Subproject commit bfb54953cee1bc985ba8113dd2e635e1d294abdb

6
build-system/generate-xcode-project.sh
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@ -89,3 +89,9 @@ sed -i "" -e '1h;2,$H;$!d;g' -e 's/\("sourceFilters" : \[\n[ ]*\)"\.\/\.\.\."/\1
--genconfig "$GEN_DIRECTORY/${APP_TARGET}.tulsiproj:${APP_TARGET}" \
--bazel "$BAZEL" \
--outputfolder "$GEN_DIRECTORY" \
--no-open-xcode \
sed -i '' -e '1h;2,$H;$!d;g' -e 's/BUILD_SETTINGS = BazelBuildSettings(/import os\nBUILD_SETTINGS = BazelBuildSettings(/g' "$GEN_DIRECTORY/${APP_TARGET}.xcodeproj/.tulsi/Scripts/bazel_build_settings.py"
sed -i '' -e '1h;2,$H;$!d;g' -e "s/'--cpu=ios_arm64'/'--cpu=ios_arm64'.replace('ios_arm64', 'ios_sim_arm64' if os.environ.get('EFFECTIVE_PLATFORM_NAME') == '-iphonesimulator' else 'ios_arm64')/g" "$GEN_DIRECTORY/${APP_TARGET}.xcodeproj/.tulsi/Scripts/bazel_build_settings.py"
open "$GEN_DIRECTORY/${APP_TARGET}.xcodeproj"

2
build-system/xcode_version
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@ -1 +1 @@
12.1
12.2

41
submodules/Opus/BUCK vendored
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@ -1,41 +0,0 @@
load("//Config:buck_rule_macros.bzl", "static_library")
genrule(
name = "opus_lib_file",
visibility = [
"//submodules/Opus:..."
],
srcs = [
"Sources/opus/lib/libopus.a",
],
bash = "mkdir -p $OUT; cp $SRCS $OUT/",
out = "opus_lib_file",
)
apple_library(
name = "opus_lib",
visibility = [
"//submodules/Opus:..."
],
header_namespace = "opus",
exported_headers = glob([
"Sources/**/*.h",
]),
exported_linker_flags = [
"-lopus",
"-L$(location :opus_lib_file)",
],
)
static_library(
name = "opus",
srcs = glob([
"Sources/**/*.swift",
]),
exported_headers = glob([
"Public/**/*.h",
]),
deps = [
":opus_lib",
],
)

26
submodules/Opus/BUILD vendored
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@ -1,26 +0,0 @@
cc_library(
name = "opus_lib",
srcs = [
"Sources/opus/lib/libopus.a",
],
)
objc_library(
name = "opus",
module_name = "opus",
enable_modules = True,
hdrs = glob([
"Public/**/*.h",
]),
includes = [
"Public",
"Public/opus",
],
deps = [
":opus_lib",
],
visibility = [
"//visibility:public",
]
)

981
submodules/Opus/Public/opus/opus.h
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@ -1,981 +0,0 @@
/* 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/opus_types.h>
#include <opus/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 (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@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 (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/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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submodules/Opus/Public/opus/opus_defines.h
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@ -1,788 +0,0 @@
/* 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/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
/** 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>Enable inband FEC.</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.</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)
/**@}*/
/** @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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@ -1,660 +0,0 @@
/* 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/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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submodules/Opus/Public/opus/opus_projection.h
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@ -1,568 +0,0 @@
/* 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/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 */

166
submodules/Opus/Public/opus/opus_types.h
View File

@ -1,166 +0,0 @@
/* (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 */

BIN
submodules/Opus/Sources/opus/lib/libopus.a
View File

Binary file not shown.

2
submodules/OpusBinding/BUILD vendored
View File

@ -20,7 +20,7 @@ objc_library(
"-I{}/PublicHeaders/OpusBinding".format(package_name()),
],
deps = [
"//submodules/Opus:opus",
"//third-party/opus:opus",
],
sdk_frameworks = [
"Foundation",

4
submodules/RMIntro/Sources/platform/ios/RMIntroViewController.m
View File

@ -236,6 +236,10 @@ static void TGDispatchOnMainThread(dispatch_block_t block) {
- (void)loadGL
{
#if TARGET_OS_SIMULATOR && defined(__aarch64__)
return;
#endif
if (/*[[UIApplication sharedApplication] applicationState] != UIApplicationStateBackground*/true && !_isOpenGLLoaded)
{
context = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];

2
submodules/TelegramUI/BUILD
View File

@ -164,7 +164,7 @@ swift_library(
"//submodules/DeviceProximity:DeviceProximity",
"//submodules/RaiseToListen:RaiseToListen",
"//submodules/OpusBinding:OpusBinding",
"//submodules/Opus:opus",
"//third-party/opus:opus",
"//submodules/WatchBridgeAudio:WatchBridgeAudio",
"//submodules/WatchBridge:WatchBridge",
"//submodules/LegacyDataImport:LegacyDataImport",

54
submodules/TgVoip/BUILD
View File

@ -7,6 +7,7 @@ copts_arm = [
"-DWEBRTC_NS_FLOAT",
"-DWEBRTC_IOS",
"-DWEBRTC_HAS_NEON",
"-DTGVOIP_NO_DSP",
]
copts_x86 = [
@ -14,52 +15,11 @@ copts_x86 = [
"-DWEBRTC_APM_DEBUG_DUMP=0",
"-DWEBRTC_POSIX",
"-DTGVOIP_HAVE_TGLOG",
"-DTGVOIP_NO_DSP",
"-DWEBRTC_NS_FLOAT",
"-DWEBRTC_IOS",
]
replace_symbols = [
"WebRtcAgc_Process",
"rtc_FatalMessage",
"WebRtcAgc_UpdateAgcThresholds",
"WebRtcAgc_Init",
"WebRtcAgc_GetAddFarendError",
"WebRtcAgc_ZeroCtrl",
"WebRtcAgc_SaturationCtrl",
"WebRtcAgc_SpeakerInactiveCtrl",
"WebRtcAgc_ProcessAnalog",
"WebRtcAgc_set_config",
"WebRtcAgc_get_config",
"WebRtcAgc_ExpCurve",
"WebRtcAgc_Create",
"WebRtcAgc_Free",
"WebRtcAgc_AddFarend",
"WebRtcAgc_VirtualMic",
"WebRtcAgc_AddMic",
"WebRtcAgc_InitDigital",
"WebRtcAgc_AddFarendToDigital",
"WebRtcAgc_CalculateGainTable",
"WebRtcAgc_InitVad",
"WebRtcAgc_ProcessVad",
"TimeDiff",
"TimeAfter",
"TimeMicros",
"TimeUTCMicros",
"SystemTimeNanos",
"TimeNanos",
"SystemTimeMillis",
"TimeMillis",
"TimeUTCMillis",
"GetClockForTesting",
"TimestampWrapAroundHandler",
"Time32",
"TmToSeconds",
"TimeDiff32",
"TimestampWrapAroundHandler",
"g_clock",
"SetClockForTesting",
]
objc_library(
name = "TgVoip",
enable_modules = True,
@ -77,9 +37,6 @@ objc_library(
"libtgvoip/os/darwin/*.mm",
"libtgvoip/os/darwin/*.cpp",
"libtgvoip/os/posix/*.cpp",
"libtgvoip/webrtc_dsp/**/*.c",
"libtgvoip/webrtc_dsp/**/*.cc",
"libtgvoip/webrtc_dsp/**/*.cpp",
], exclude = ["libtgvoip/os/darwin/*OSX*"]),
hdrs = glob([
"PublicHeaders/**/*.h",
@ -87,14 +44,15 @@ objc_library(
copts = [
"-I{}/PublicHeaders/TgVoip".format(package_name()),
"-I{}/libtgvoip".format(package_name()),
"-I{}/libtgvoip/webrtc_dsp".format(package_name()),
"-I{}/third-party/webrtc/webrtc-ios/src".format(package_name()),
"-Isubmodules/Opus/Public/opus",
"-DTGVOIP_USE_INSTALLED_OPUS",
"-Drtc=rtc1",
"-Dwebrtc=webrtc1",
] + ["-D{symbol}={symbol}1".format(symbol=symbol) for symbol in replace_symbols] + select({
] + select({
"@build_bazel_rules_apple//apple:ios_armv7": copts_arm,
"@build_bazel_rules_apple//apple:ios_arm64": copts_arm,
"//build-system:ios_sim_arm64": copts_arm,
"@build_bazel_rules_apple//apple:ios_x86_64": copts_x86,
}),
includes = [
@ -102,7 +60,7 @@ objc_library(
],
deps = [
"//submodules/MtProtoKit:MtProtoKit",
"//submodules/Opus:opus",
"//third-party/opus:opus",
],
sdk_frameworks = [
"Foundation",

52
submodules/TgVoip/Sources/OngoingCallThreadLocalContext.mm
View File

@ -127,7 +127,7 @@ static void withContext(int32_t contextId, void (^f)(OngoingCallThreadLocalConte
NSTimeInterval _callConnectTimeout;
NSTimeInterval _callPacketTimeout;
TgVoip *_tgVoip;
std::unique_ptr<TgVoip> _tgVoip;
OngoingCallState _state;
int32_t _signalBars;
@ -262,45 +262,32 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
endpoints.push_back(endpoint);
}
TgVoipConfig config = {
.initializationTimeout = _callConnectTimeout,
.receiveTimeout = _callPacketTimeout,
.dataSaving = callControllerDataSavingForType(dataSaving),
.enableP2P = static_cast<bool>(allowP2P),
.enableAEC = false,
.enableNS = true,
.enableAGC = true,
.enableCallUpgrade = false,
.logPath = logPath.length == 0 ? "" : std::string(logPath.UTF8String),
.maxApiLayer = [OngoingCallThreadLocalContext maxLayer]
};
TgVoipConfig config;
config.initializationTimeout = _callConnectTimeout;
config.receiveTimeout = _callPacketTimeout;
config.dataSaving = callControllerDataSavingForType(dataSaving);
config.enableP2P = static_cast<bool>(allowP2P);
config.enableAEC = false;
config.enableNS = true;
config.enableAGC = true;
config.enableVolumeControl = false;
config.enableCallUpgrade = false;
config.logPath = logPath.length == 0 ? "" : std::string(logPath.UTF8String);
config.maxApiLayer = [OngoingCallThreadLocalContext maxLayer];
std::vector<uint8_t> encryptionKeyValue;
encryptionKeyValue.resize(key.length);
memcpy(encryptionKeyValue.data(), key.bytes, key.length);
TgVoipEncryptionKey encryptionKey = {
.value = encryptionKeyValue,
.isOutgoing = isOutgoing,
};
/*
TgVoipConfig const &config,
TgVoipPersistentState const &persistentState,
std::vector<TgVoipEndpoint> const &endpoints,
std::unique_ptr<TgVoipProxy> const &proxy,
TgVoipNetworkType initialNetworkType,
TgVoipEncryptionKey const &encryptionKey
#ifdef TGVOIP_USE_CUSTOM_CRYPTO
,
TgVoipCrypto const &crypto
*/
TgVoipEncryptionKey encryptionKey;
encryptionKey.value = encryptionKeyValue;
encryptionKey.isOutgoing = isOutgoing;
_tgVoip = TgVoip::makeInstance(
config,
{ derivedStateValue },
endpoints,
proxyValue,
proxyValue.get(),
callControllerNetworkTypeForType(networkType),
encryptionKey,
crypto
@ -341,8 +328,7 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
NSString *debugLog = [NSString stringWithUTF8String:finalState.debugLog.c_str()];
_lastDerivedState = [[NSData alloc] initWithBytes:finalState.persistentState.value.data() length:finalState.persistentState.value.size()];
delete _tgVoip;
_tgVoip = NULL;
_tgVoip.reset();
if (completion) {
completion(debugLog, finalState.trafficStats.bytesSentWifi, finalState.trafficStats.bytesReceivedWifi, finalState.trafficStats.bytesSentMobile, finalState.trafficStats.bytesReceivedMobile);
@ -381,7 +367,7 @@ static void (*InternalVoipLoggingFunction)(NSString *) = NULL;
- (void)controllerStateChanged:(TgVoipState)state {
OngoingCallState callState = OngoingCallStateInitializing;
switch (state) {
case TgVoipState::Estabilished:
case TgVoipState::Established:
callState = OngoingCallStateConnected;
break;
case TgVoipState::Failed:

2
submodules/TgVoip/libtgvoip

@ -1 +1 @@
Subproject commit dc4e9ec48207388e41db1c2ef1cccf9899d9765f
Subproject commit 37d98e984fd6fa389262307db826d52ab86c8241

2
submodules/WebPBinding/BUCK
View File

@ -12,7 +12,7 @@ static_library(
"PublicHeaders/**/*.h",
]),
deps = [
"//submodules/webp:webp",
"//third-party/webp:webp",
],
frameworks = [
"$SDKROOT/System/Library/Frameworks/Foundation.framework",

2
submodules/WebPBinding/BUILD
View File

@ -14,7 +14,7 @@ objc_library(
"PublicHeaders",
],
deps = [
"//submodules/webp:webp",
"//third-party/webp:webp",
],
sdk_frameworks = [
"Foundation",

38
submodules/ffmpeg/BUILD vendored
View File

@ -120,15 +120,29 @@ filegroup(
srcs = source_files,
)
opus_headers = [
"opus.h",
"opus_defines.h",
"opus_multistream.h",
"opus_projection.h",
"opus_types.h",
]
opus_libs = [
"opus",
]
genrule(
name = "libffmpeg_build",
srcs = [
":FFMpegSources",
":FFMpegSources"
] + [
"//third-party/opus:Public/opus/{}".format(x) for x in opus_headers
] + [
"//third-party/opus:Public/opus/lib/lib{}.a".format(x) for x in opus_libs
],
cmd_bash =
"""
#random:2
set -x
ABS_RULEDIR="$$(pwd)/$(RULEDIR)"
@ -139,6 +153,20 @@ genrule(
cp -R "submodules/ffmpeg/Sources/FFMpeg" "$$SOURCE_PATH"
mkdir "$$SOURCE_PATH/libopus"
mkdir -p "$$SOURCE_PATH/libopus/include/opus"
mkdir -p "$$SOURCE_PATH/libopus/lib"
""" +
"\n" +
"\n".join([
"cp $(location //third-party/opus:Public/opus/{}) $$SOURCE_PATH/libopus/include/opus/".format(x) for x in opus_headers
]) +
"\n" +
"\n".join([
"cp $(location //third-party/opus:Public/opus/lib/lib{}.a) $$SOURCE_PATH/libopus/lib/".format(x) for x in opus_libs
]) +
"\n" +
"""
BUILD_DIR="$$ABS_RULEDIR/build"
rm -rf "$$BUILD_DIR"
mkdir -p "$$BUILD_DIR"
@ -151,6 +179,8 @@ genrule(
BUILD_ARCH="armv7"
elif [ "$(TARGET_CPU)" == "ios_arm64" ]; then
BUILD_ARCH="arm64"
elif [ "$(TARGET_CPU)" == "ios_sim_arm64" ]; then
BUILD_ARCH="sim_arm64"
elif [ "$(TARGET_CPU)" == "ios_x86_64" ]; then
BUILD_ARCH="x86_64"
else
@ -197,7 +227,7 @@ objc_library(
],
deps = [
":ffmpeg_lib",
"//submodules/Opus:opus",
"//third-party/opus:opus",
],
visibility = [
"//visibility:public",

42
submodules/ffmpeg/Sources/FFMpeg/build-ffmpeg-bazel.sh
View File

@ -1,21 +1,13 @@
#!/bin/sh
set -x
RAW_ARCHS="$2"
ARCHS=""
for RAW_ARCH in $RAW_ARCHS; do
ARCH_NAME="$RAW_ARCH"
if [ "$ARCH_NAME" == "iphoneos-arm64" ]; then
ARCH_NAME="arm64"
elif [ "$ARCH_NAME" == "iphoneos-armv7" ]; then
ARCH_NAME="armv7"
elif [ "$ARCH_NAME" == "iphonesimulator-x86_64" ]; then
ARCH_NAME="x86_64"
elif [ "$ARCH_NAME" == "iphonesimulator-i386" ]; then
ARCH_NAME="i386"
fi
if [ "$ARCH_NAME" = "i386" -o "$ARCH_NAME" = "x86_64" -o "$ARCH_NAME" = "arm64" -o "$ARCH_NAME" = "armv7" ]
if [ "$ARCH_NAME" = "i386" -o "$ARCH_NAME" = "x86_64" -o "$ARCH_NAME" = "arm64" -o "$ARCH_NAME" = "armv7" -o "$ARCH_NAME" = "sim_arm64" ]
then
ARCHS="$ARCHS $ARCH_NAME"
else
@ -78,7 +70,7 @@ fi
COMPILE="y"
DEPLOYMENT_TARGET="8.0"
DEPLOYMENT_TARGET="9.0"
LIBS_HASH=""
for ARCH in $ARCHS
@ -116,19 +108,27 @@ then
exit 1
fi
for ARCH in $ARCHS
for RAW_ARCH in $ARCHS
do
echo "building $ARCH..."
mkdir -p "$SCRATCH/$ARCH"
pushd "$SCRATCH/$ARCH"
ARCH="$RAW_ARCH"
if [ "$RAW_ARCH" == "sim_arm64" ]; then
ARCH="arm64"
fi
echo "building $RAW_ARCH..."
mkdir -p "$SCRATCH/$RAW_ARCH"
pushd "$SCRATCH/$RAW_ARCH"
LIBOPUS_PATH="$SOURCE_DIR/libopus"
CFLAGS="-arch $ARCH"
if [ "$ARCH" = "i386" -o "$ARCH" = "x86_64" ]
if [ "$RAW_ARCH" = "i386" -o "$RAW_ARCH" = "x86_64" ]
then
PLATFORM="iPhoneSimulator"
CFLAGS="$CFLAGS -mios-simulator-version-min=$DEPLOYMENT_TARGET"
elif [ "$RAW_ARCH" = "sim_arm64" ]; then
PLATFORM="iPhoneSimulator"
CFLAGS="$CFLAGS -mios-simulator-version-min=$DEPLOYMENT_TARGET --target=arm64-apple-ios$DEPLOYMENT_TARGET-simulator"
else
PLATFORM="iPhoneOS"
CFLAGS="$CFLAGS -mios-version-min=$DEPLOYMENT_TARGET"
@ -141,7 +141,7 @@ then
XCRUN_SDK=`echo $PLATFORM | tr '[:upper:]' '[:lower:]'`
CC="xcrun -sdk $XCRUN_SDK clang"
if [ "$ARCH" = "arm64" ]
if [ "$RAW_ARCH" = "arm64" ] || [ "$RAW_ARCH" = "sim_arm64" ]
then
AS="$GAS_PREPROCESSOR_PATH -arch aarch64 -- $CC"
else
@ -151,7 +151,7 @@ then
CXXFLAGS="$CFLAGS"
LDFLAGS="$CFLAGS"
CONFIGURED_MARKER="$THIN/$ARCH/configured_marker"
CONFIGURED_MARKER="$THIN/$RAW_ARCH/configured_marker"
CONFIGURED_MARKER_CONTENTS=""
if [ -r "$CONFIGURED_MARKER" ]
then
@ -161,7 +161,7 @@ then
then
echo "1" >/dev/null
else
mkdir -p "$THIN/$ARCH"
mkdir -p "$THIN/$RAW_ARCH"
TMPDIR=${TMPDIR/%\/} "$SOURCE/configure" \
--target-os=darwin \
--arch=$ARCH \
@ -170,7 +170,7 @@ then
$CONFIGURE_FLAGS \
--extra-cflags="$CFLAGS" \
--extra-ldflags="$LDFLAGS" \
--prefix="$THIN/$ARCH" \
--prefix="$THIN/$RAW_ARCH" \
--pkg-config="$PKG_CONFIG" \
--pkg-config-flags="--libopus_path $LIBOPUS_PATH" \
|| exit 1

981
submodules/ffmpeg/Sources/FFMpeg/libopus/include/opus/opus.h
View File

@ -1,981 +0,0 @@
/* 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 (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@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 (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/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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submodules/ffmpeg/Sources/FFMpeg/libopus/include/opus/opus_defines.h
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@ -1,788 +0,0 @@
/* 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
/** 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>Enable inband FEC.</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.</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)
/**@}*/
/** @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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submodules/ffmpeg/Sources/FFMpeg/libopus/include/opus/opus_multistream.h
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@ -1,660 +0,0 @@
/* 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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submodules/ffmpeg/Sources/FFMpeg/libopus/include/opus/opus_projection.h
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@ -1,568 +0,0 @@
/* 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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/* 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 (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@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 (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/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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submodules/ffmpeg/Sources/opus/include/opus/opus_defines.h
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@ -1,788 +0,0 @@
/* 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
/** 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>Enable inband FEC.</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.</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)
/**@}*/
/** @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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@ -1,660 +0,0 @@
/* 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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submodules/ffmpeg/Sources/opus/include/opus/opus_projection.h
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@ -1,568 +0,0 @@
/* 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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submodules/ffmpeg/Sources/opus/include/opus/opus_types.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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submodules/ffmpeg/Sources/opus/lib/libopus.a
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26
submodules/webp/.gitignore vendored
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fastlane/README.md
fastlane/report.xml
fastlane/test_output/*
build/*
*.pbxuser
!default.pbxuser
*.mode1v3
!default.mode1v3
*.mode2v3
!default.mode2v3
*.perspectivev3
!default.perspectivev3
xcuserdata
*.xccheckout
*.xcscmblueprint
*.moved-aside
DerivedData
*.hmap
*.ipa
*.xcuserstate
.DS_Store
*.dSYM
*.dSYM.zip
*.ipa
*/xcuserdata/*
webp.xcodeproj/*

26
submodules/webp/BUCK
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@ -1,26 +0,0 @@
load("//Config:buck_rule_macros.bzl", "static_library", "framework")
genrule(
name = "WebP_lib_file",
srcs = [
"lib/libwebp.a",
],
bash = "mkdir -p $OUT; cp $SRCS $OUT/",
out = "WebP_libs",
visibility = ["PUBLIC"]
)
apple_library(
name = "webp",
header_namespace = "webp",
exported_headers = {
"webp/encode.h": "include/webp/encode.h",
"webp/decode.h": "include/webp/decode.h",
"webp/types.h": "include/webp/types.h",
},
exported_linker_flags = [
"-lwebp",
"-L$(location :WebP_lib_file)",
],
visibility = ["PUBLIC"],
)

25
submodules/webp/BUILD
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@ -1,25 +0,0 @@
cc_library(
name = "webp_lib",
srcs = [
"lib/libwebp.a",
],
)
objc_library(
name = "webp",
module_name = "webp",
enable_modules = True,
hdrs = glob([
"include/**/*.h",
]),
includes = [
"include",
],
deps = [
":webp_lib",
],
visibility = [
"//visibility:public",
]
)

491
submodules/webp/include/webp/decode.h
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@ -1,491 +0,0 @@
// Copyright 2010 Google Inc. All Rights Reserved.
//
// Use of this source code is governed by a BSD-style license
// that can be found in the COPYING file in the root of the source
// tree. An additional intellectual property rights grant can be found
// in the file PATENTS. All contributing project authors may
// be found in the AUTHORS file in the root of the source tree.
// -----------------------------------------------------------------------------
//
// Main decoding functions for WebP images.
//
// Author: Skal (pascal.massimino@gmail.com)
#ifndef WEBP_WEBP_DECODE_H_
#define WEBP_WEBP_DECODE_H_
#include "./types.h"
#ifdef __cplusplus
extern "C" {
#endif
#define WEBP_DECODER_ABI_VERSION 0x0208 // MAJOR(8b) + MINOR(8b)
// Note: forward declaring enumerations is not allowed in (strict) C and C++,
// the types are left here for reference.
// typedef enum VP8StatusCode VP8StatusCode;
// typedef enum WEBP_CSP_MODE WEBP_CSP_MODE;
typedef struct WebPRGBABuffer WebPRGBABuffer;
typedef struct WebPYUVABuffer WebPYUVABuffer;
typedef struct WebPDecBuffer WebPDecBuffer;
typedef struct WebPIDecoder WebPIDecoder;
typedef struct WebPBitstreamFeatures WebPBitstreamFeatures;
typedef struct WebPDecoderOptions WebPDecoderOptions;
typedef struct WebPDecoderConfig WebPDecoderConfig;
// Return the decoder's version number, packed in hexadecimal using 8bits for
// each of major/minor/revision. E.g: v2.5.7 is 0x020507.
WEBP_EXTERN(int) WebPGetDecoderVersion(void);
// Retrieve basic header information: width, height.
// This function will also validate the header and return 0 in
// case of formatting error.
// Pointers 'width' and 'height' can be passed NULL if deemed irrelevant.
WEBP_EXTERN(int) WebPGetInfo(const uint8_t* data, size_t data_size,
int* width, int* height);
// Decodes WebP images pointed to by 'data' and returns RGBA samples, along
// with the dimensions in *width and *height. The ordering of samples in
// memory is R, G, B, A, R, G, B, A... in scan order (endian-independent).
// The returned pointer should be deleted calling WebPFree().
// Returns NULL in case of error.
WEBP_EXTERN(uint8_t*) WebPDecodeRGBA(const uint8_t* data, size_t data_size,
int* width, int* height);
// Same as WebPDecodeRGBA, but returning A, R, G, B, A, R, G, B... ordered data.
WEBP_EXTERN(uint8_t*) WebPDecodeARGB(const uint8_t* data, size_t data_size,
int* width, int* height);
// Same as WebPDecodeRGBA, but returning B, G, R, A, B, G, R, A... ordered data.
WEBP_EXTERN(uint8_t*) WebPDecodeBGRA(const uint8_t* data, size_t data_size,
int* width, int* height);
// Same as WebPDecodeRGBA, but returning R, G, B, R, G, B... ordered data.
// If the bitstream contains transparency, it is ignored.
WEBP_EXTERN(uint8_t*) WebPDecodeRGB(const uint8_t* data, size_t data_size,
int* width, int* height);
// Same as WebPDecodeRGB, but returning B, G, R, B, G, R... ordered data.
WEBP_EXTERN(uint8_t*) WebPDecodeBGR(const uint8_t* data, size_t data_size,
int* width, int* height);
// Decode WebP images pointed to by 'data' to Y'UV format(*). The pointer
// returned is the Y samples buffer. Upon return, *u and *v will point to
// the U and V chroma data. These U and V buffers need NOT be passed to
// WebPFree(), unlike the returned Y luma one. The dimension of the U and V
// planes are both (*width + 1) / 2 and (*height + 1)/ 2.
// Upon return, the Y buffer has a stride returned as '*stride', while U and V
// have a common stride returned as '*uv_stride'.
// Return NULL in case of error.
// (*) Also named Y'CbCr. See: http://en.wikipedia.org/wiki/YCbCr
WEBP_EXTERN(uint8_t*) WebPDecodeYUV(const uint8_t* data, size_t data_size,
int* width, int* height,
uint8_t** u, uint8_t** v,
int* stride, int* uv_stride);
// Releases memory returned by the WebPDecode*() functions above.
WEBP_EXTERN(void) WebPFree(void* ptr);
// These five functions are variants of the above ones, that decode the image
// directly into a pre-allocated buffer 'output_buffer'. The maximum storage
// available in this buffer is indicated by 'output_buffer_size'. If this
// storage is not sufficient (or an error occurred), NULL is returned.
// Otherwise, output_buffer is returned, for convenience.
// The parameter 'output_stride' specifies the distance (in bytes)
// between scanlines. Hence, output_buffer_size is expected to be at least
// output_stride x picture-height.
WEBP_EXTERN(uint8_t*) WebPDecodeRGBAInto(
const uint8_t* data, size_t data_size,
uint8_t* output_buffer, size_t output_buffer_size, int output_stride);
WEBP_EXTERN(uint8_t*) WebPDecodeARGBInto(
const uint8_t* data, size_t data_size,
uint8_t* output_buffer, size_t output_buffer_size, int output_stride);
WEBP_EXTERN(uint8_t*) WebPDecodeBGRAInto(
const uint8_t* data, size_t data_size,
uint8_t* output_buffer, size_t output_buffer_size, int output_stride);
// RGB and BGR variants. Here too the transparency information, if present,
// will be dropped and ignored.
WEBP_EXTERN(uint8_t*) WebPDecodeRGBInto(
const uint8_t* data, size_t data_size,
uint8_t* output_buffer, size_t output_buffer_size, int output_stride);
WEBP_EXTERN(uint8_t*) WebPDecodeBGRInto(
const uint8_t* data, size_t data_size,
uint8_t* output_buffer, size_t output_buffer_size, int output_stride);
// WebPDecodeYUVInto() is a variant of WebPDecodeYUV() that operates directly
// into pre-allocated luma/chroma plane buffers. This function requires the
// strides to be passed: one for the luma plane and one for each of the
// chroma ones. The size of each plane buffer is passed as 'luma_size',
// 'u_size' and 'v_size' respectively.
// Pointer to the luma plane ('*luma') is returned or NULL if an error occurred
// during decoding (or because some buffers were found to be too small).
WEBP_EXTERN(uint8_t*) WebPDecodeYUVInto(
const uint8_t* data, size_t data_size,
uint8_t* luma, size_t luma_size, int luma_stride,
uint8_t* u, size_t u_size, int u_stride,
uint8_t* v, size_t v_size, int v_stride);
//------------------------------------------------------------------------------
// Output colorspaces and buffer
// Colorspaces
// Note: the naming describes the byte-ordering of packed samples in memory.
// For instance, MODE_BGRA relates to samples ordered as B,G,R,A,B,G,R,A,...
// Non-capital names (e.g.:MODE_Argb) relates to pre-multiplied RGB channels.
// RGBA-4444 and RGB-565 colorspaces are represented by following byte-order:
// RGBA-4444: [r3 r2 r1 r0 g3 g2 g1 g0], [b3 b2 b1 b0 a3 a2 a1 a0], ...
// RGB-565: [r4 r3 r2 r1 r0 g5 g4 g3], [g2 g1 g0 b4 b3 b2 b1 b0], ...
// In the case WEBP_SWAP_16BITS_CSP is defined, the bytes are swapped for
// these two modes:
// RGBA-4444: [b3 b2 b1 b0 a3 a2 a1 a0], [r3 r2 r1 r0 g3 g2 g1 g0], ...
// RGB-565: [g2 g1 g0 b4 b3 b2 b1 b0], [r4 r3 r2 r1 r0 g5 g4 g3], ...
typedef enum WEBP_CSP_MODE {
MODE_RGB = 0, MODE_RGBA = 1,
MODE_BGR = 2, MODE_BGRA = 3,
MODE_ARGB = 4, MODE_RGBA_4444 = 5,
MODE_RGB_565 = 6,
// RGB-premultiplied transparent modes (alpha value is preserved)
MODE_rgbA = 7,
MODE_bgrA = 8,
MODE_Argb = 9,
MODE_rgbA_4444 = 10,
// YUV modes must come after RGB ones.
MODE_YUV = 11, MODE_YUVA = 12, // yuv 4:2:0
MODE_LAST = 13
} WEBP_CSP_MODE;
// Some useful macros:
static WEBP_INLINE int WebPIsPremultipliedMode(WEBP_CSP_MODE mode) {
return (mode == MODE_rgbA || mode == MODE_bgrA || mode == MODE_Argb ||
mode == MODE_rgbA_4444);
}
static WEBP_INLINE int WebPIsAlphaMode(WEBP_CSP_MODE mode) {
return (mode == MODE_RGBA || mode == MODE_BGRA || mode == MODE_ARGB ||
mode == MODE_RGBA_4444 || mode == MODE_YUVA ||
WebPIsPremultipliedMode(mode));
}
static WEBP_INLINE int WebPIsRGBMode(WEBP_CSP_MODE mode) {
return (mode < MODE_YUV);
}
//------------------------------------------------------------------------------
// WebPDecBuffer: Generic structure for describing the output sample buffer.
struct WebPRGBABuffer { // view as RGBA
uint8_t* rgba; // pointer to RGBA samples
int stride; // stride in bytes from one scanline to the next.
size_t size; // total size of the *rgba buffer.
};
struct WebPYUVABuffer { // view as YUVA
uint8_t* y, *u, *v, *a; // pointer to luma, chroma U/V, alpha samples
int y_stride; // luma stride
int u_stride, v_stride; // chroma strides
int a_stride; // alpha stride
size_t y_size; // luma plane size
size_t u_size, v_size; // chroma planes size
size_t a_size; // alpha-plane size
};
// Output buffer
struct WebPDecBuffer {
WEBP_CSP_MODE colorspace; // Colorspace.
int width, height; // Dimensions.
int is_external_memory; // If non-zero, 'internal_memory' pointer is not
// used. If value is '2' or more, the external
// memory is considered 'slow' and multiple
// read/write will be avoided.
union {
WebPRGBABuffer RGBA;
WebPYUVABuffer YUVA;
} u; // Nameless union of buffer parameters.
uint32_t pad[4]; // padding for later use
uint8_t* private_memory; // Internally allocated memory (only when
// is_external_memory is 0). Should not be used
// externally, but accessed via the buffer union.
};
// Internal, version-checked, entry point
WEBP_EXTERN(int) WebPInitDecBufferInternal(WebPDecBuffer*, int);
// Initialize the structure as empty. Must be called before any other use.
// Returns false in case of version mismatch
static WEBP_INLINE int WebPInitDecBuffer(WebPDecBuffer* buffer) {
return WebPInitDecBufferInternal(buffer, WEBP_DECODER_ABI_VERSION);
}
// Free any memory associated with the buffer. Must always be called last.
// Note: doesn't free the 'buffer' structure itself.
WEBP_EXTERN(void) WebPFreeDecBuffer(WebPDecBuffer* buffer);
//------------------------------------------------------------------------------
// Enumeration of the status codes
typedef enum VP8StatusCode {
VP8_STATUS_OK = 0,
VP8_STATUS_OUT_OF_MEMORY,
VP8_STATUS_INVALID_PARAM,
VP8_STATUS_BITSTREAM_ERROR,
VP8_STATUS_UNSUPPORTED_FEATURE,
VP8_STATUS_SUSPENDED,
VP8_STATUS_USER_ABORT,
VP8_STATUS_NOT_ENOUGH_DATA
} VP8StatusCode;
//------------------------------------------------------------------------------
// Incremental decoding
//
// This API allows streamlined decoding of partial data.
// Picture can be incrementally decoded as data become available thanks to the
// WebPIDecoder object. This object can be left in a SUSPENDED state if the
// picture is only partially decoded, pending additional input.
// Code example:
//
// WebPInitDecBuffer(&buffer);
// buffer.colorspace = mode;
// ...
// WebPIDecoder* idec = WebPINewDecoder(&buffer);
// while (has_more_data) {
// // ... (get additional data)
// status = WebPIAppend(idec, new_data, new_data_size);
// if (status != VP8_STATUS_SUSPENDED ||
// break;
// }
//
// // The above call decodes the current available buffer.
// // Part of the image can now be refreshed by calling to
// // WebPIDecGetRGB()/WebPIDecGetYUVA() etc.
// }
// WebPIDelete(idec);
// Creates a new incremental decoder with the supplied buffer parameter.
// This output_buffer can be passed NULL, in which case a default output buffer
// is used (with MODE_RGB). Otherwise, an internal reference to 'output_buffer'
// is kept, which means that the lifespan of 'output_buffer' must be larger than
// that of the returned WebPIDecoder object.
// The supplied 'output_buffer' content MUST NOT be changed between calls to
// WebPIAppend() or WebPIUpdate() unless 'output_buffer.is_external_memory' is
// not set to 0. In such a case, it is allowed to modify the pointers, size and
// stride of output_buffer.u.RGBA or output_buffer.u.YUVA, provided they remain
// within valid bounds.
// All other fields of WebPDecBuffer MUST remain constant between calls.
// Returns NULL if the allocation failed.
WEBP_EXTERN(WebPIDecoder*) WebPINewDecoder(WebPDecBuffer* output_buffer);
// This function allocates and initializes an incremental-decoder object, which
// will output the RGB/A samples specified by 'csp' into a preallocated
// buffer 'output_buffer'. The size of this buffer is at least
// 'output_buffer_size' and the stride (distance in bytes between two scanlines)
// is specified by 'output_stride'.
// Additionally, output_buffer can be passed NULL in which case the output
// buffer will be allocated automatically when the decoding starts. The
// colorspace 'csp' is taken into account for allocating this buffer. All other
// parameters are ignored.
// Returns NULL if the allocation failed, or if some parameters are invalid.
WEBP_EXTERN(WebPIDecoder*) WebPINewRGB(
WEBP_CSP_MODE csp,
uint8_t* output_buffer, size_t output_buffer_size, int output_stride);
// This function allocates and initializes an incremental-decoder object, which
// will output the raw luma/chroma samples into a preallocated planes if
// supplied. The luma plane is specified by its pointer 'luma', its size
// 'luma_size' and its stride 'luma_stride'. Similarly, the chroma-u plane
// is specified by the 'u', 'u_size' and 'u_stride' parameters, and the chroma-v
// plane by 'v' and 'v_size'. And same for the alpha-plane. The 'a' pointer
// can be pass NULL in case one is not interested in the transparency plane.
// Conversely, 'luma' can be passed NULL if no preallocated planes are supplied.
// In this case, the output buffer will be automatically allocated (using
// MODE_YUVA) when decoding starts. All parameters are then ignored.
// Returns NULL if the allocation failed or if a parameter is invalid.
WEBP_EXTERN(WebPIDecoder*) WebPINewYUVA(
uint8_t* luma, size_t luma_size, int luma_stride,
uint8_t* u, size_t u_size, int u_stride,
uint8_t* v, size_t v_size, int v_stride,
uint8_t* a, size_t a_size, int a_stride);
// Deprecated version of the above, without the alpha plane.
// Kept for backward compatibility.
WEBP_EXTERN(WebPIDecoder*) WebPINewYUV(
uint8_t* luma, size_t luma_size, int luma_stride,
uint8_t* u, size_t u_size, int u_stride,
uint8_t* v, size_t v_size, int v_stride);
// Deletes the WebPIDecoder object and associated memory. Must always be called
// if WebPINewDecoder, WebPINewRGB or WebPINewYUV succeeded.
WEBP_EXTERN(void) WebPIDelete(WebPIDecoder* idec);
// Copies and decodes the next available data. Returns VP8_STATUS_OK when
// the image is successfully decoded. Returns VP8_STATUS_SUSPENDED when more
// data is expected. Returns error in other cases.
WEBP_EXTERN(VP8StatusCode) WebPIAppend(
WebPIDecoder* idec, const uint8_t* data, size_t data_size);
// A variant of the above function to be used when data buffer contains
// partial data from the beginning. In this case data buffer is not copied
// to the internal memory.
// Note that the value of the 'data' pointer can change between calls to
// WebPIUpdate, for instance when the data buffer is resized to fit larger data.
WEBP_EXTERN(VP8StatusCode) WebPIUpdate(
WebPIDecoder* idec, const uint8_t* data, size_t data_size);
// Returns the RGB/A image decoded so far. Returns NULL if output params
// are not initialized yet. The RGB/A output type corresponds to the colorspace
// specified during call to WebPINewDecoder() or WebPINewRGB().
// *last_y is the index of last decoded row in raster scan order. Some pointers
// (*last_y, *width etc.) can be NULL if corresponding information is not
// needed.
WEBP_EXTERN(uint8_t*) WebPIDecGetRGB(
const WebPIDecoder* idec, int* last_y,
int* width, int* height, int* stride);
// Same as above function to get a YUVA image. Returns pointer to the luma
// plane or NULL in case of error. If there is no alpha information
// the alpha pointer '*a' will be returned NULL.
WEBP_EXTERN(uint8_t*) WebPIDecGetYUVA(
const WebPIDecoder* idec, int* last_y,
uint8_t** u, uint8_t** v, uint8_t** a,
int* width, int* height, int* stride, int* uv_stride, int* a_stride);
// Deprecated alpha-less version of WebPIDecGetYUVA(): it will ignore the
// alpha information (if present). Kept for backward compatibility.
static WEBP_INLINE uint8_t* WebPIDecGetYUV(
const WebPIDecoder* idec, int* last_y, uint8_t** u, uint8_t** v,
int* width, int* height, int* stride, int* uv_stride) {
return WebPIDecGetYUVA(idec, last_y, u, v, NULL, width, height,
stride, uv_stride, NULL);
}
// Generic call to retrieve information about the displayable area.
// If non NULL, the left/right/width/height pointers are filled with the visible
// rectangular area so far.
// Returns NULL in case the incremental decoder object is in an invalid state.
// Otherwise returns the pointer to the internal representation. This structure
// is read-only, tied to WebPIDecoder's lifespan and should not be modified.
WEBP_EXTERN(const WebPDecBuffer*) WebPIDecodedArea(
const WebPIDecoder* idec, int* left, int* top, int* width, int* height);
//------------------------------------------------------------------------------
// Advanced decoding parametrization
//
// Code sample for using the advanced decoding API
/*
// A) Init a configuration object
WebPDecoderConfig config;
CHECK(WebPInitDecoderConfig(&config));
// B) optional: retrieve the bitstream's features.
CHECK(WebPGetFeatures(data, data_size, &config.input) == VP8_STATUS_OK);
// C) Adjust 'config', if needed
config.no_fancy_upsampling = 1;
config.output.colorspace = MODE_BGRA;
// etc.
// Note that you can also make config.output point to an externally
// supplied memory buffer, provided it's big enough to store the decoded
// picture. Otherwise, config.output will just be used to allocate memory
// and store the decoded picture.
// D) Decode!
CHECK(WebPDecode(data, data_size, &config) == VP8_STATUS_OK);
// E) Decoded image is now in config.output (and config.output.u.RGBA)
// F) Reclaim memory allocated in config's object. It's safe to call
// this function even if the memory is external and wasn't allocated
// by WebPDecode().
WebPFreeDecBuffer(&config.output);
*/
// Features gathered from the bitstream
struct WebPBitstreamFeatures {
int width; // Width in pixels, as read from the bitstream.
int height; // Height in pixels, as read from the bitstream.
int has_alpha; // True if the bitstream contains an alpha channel.
int has_animation; // True if the bitstream is an animation.
int format; // 0 = undefined (/mixed), 1 = lossy, 2 = lossless
uint32_t pad[5]; // padding for later use
};
// Internal, version-checked, entry point
WEBP_EXTERN(VP8StatusCode) WebPGetFeaturesInternal(
const uint8_t*, size_t, WebPBitstreamFeatures*, int);
// Retrieve features from the bitstream. The *features structure is filled
// with information gathered from the bitstream.
// Returns VP8_STATUS_OK when the features are successfully retrieved. Returns
// VP8_STATUS_NOT_ENOUGH_DATA when more data is needed to retrieve the
// features from headers. Returns error in other cases.
static WEBP_INLINE VP8StatusCode WebPGetFeatures(
const uint8_t* data, size_t data_size,
WebPBitstreamFeatures* features) {
return WebPGetFeaturesInternal(data, data_size, features,
WEBP_DECODER_ABI_VERSION);
}
// Decoding options
struct WebPDecoderOptions {
int bypass_filtering; // if true, skip the in-loop filtering
int no_fancy_upsampling; // if true, use faster pointwise upsampler
int use_cropping; // if true, cropping is applied _first_
int crop_left, crop_top; // top-left position for cropping.
// Will be snapped to even values.
int crop_width, crop_height; // dimension of the cropping area
int use_scaling; // if true, scaling is applied _afterward_
int scaled_width, scaled_height; // final resolution
int use_threads; // if true, use multi-threaded decoding
int dithering_strength; // dithering strength (0=Off, 100=full)
int flip; // flip output vertically
int alpha_dithering_strength; // alpha dithering strength in [0..100]
uint32_t pad[5]; // padding for later use
};
// Main object storing the configuration for advanced decoding.
struct WebPDecoderConfig {
WebPBitstreamFeatures input; // Immutable bitstream features (optional)
WebPDecBuffer output; // Output buffer (can point to external mem)
WebPDecoderOptions options; // Decoding options
};
// Internal, version-checked, entry point
WEBP_EXTERN(int) WebPInitDecoderConfigInternal(WebPDecoderConfig*, int);
// Initialize the configuration as empty. This function must always be
// called first, unless WebPGetFeatures() is to be called.
// Returns false in case of mismatched version.
static WEBP_INLINE int WebPInitDecoderConfig(WebPDecoderConfig* config) {
return WebPInitDecoderConfigInternal(config, WEBP_DECODER_ABI_VERSION);
}
// Instantiate a new incremental decoder object with the requested
// configuration. The bitstream can be passed using 'data' and 'data_size'
// parameter, in which case the features will be parsed and stored into
// config->input. Otherwise, 'data' can be NULL and no parsing will occur.
// Note that 'config' can be NULL too, in which case a default configuration
// is used.
// The return WebPIDecoder object must always be deleted calling WebPIDelete().
// Returns NULL in case of error (and config->status will then reflect
// the error condition).
WEBP_EXTERN(WebPIDecoder*) WebPIDecode(const uint8_t* data, size_t data_size,
WebPDecoderConfig* config);
// Non-incremental version. This version decodes the full data at once, taking
// 'config' into account. Returns decoding status (which should be VP8_STATUS_OK
// if the decoding was successful).
WEBP_EXTERN(VP8StatusCode) WebPDecode(const uint8_t* data, size_t data_size,
WebPDecoderConfig* config);
#ifdef __cplusplus
} // extern "C"
#endif
#endif /* WEBP_WEBP_DECODE_H_ */

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@ -1,527 +0,0 @@
// Copyright 2011 Google Inc. All Rights Reserved.
//
// Use of this source code is governed by a BSD-style license
// that can be found in the COPYING file in the root of the source
// tree. An additional intellectual property rights grant can be found
// in the file PATENTS. All contributing project authors may
// be found in the AUTHORS file in the root of the source tree.
// -----------------------------------------------------------------------------
//
// WebP encoder: main interface
//
// Author: Skal (pascal.massimino@gmail.com)
#ifndef WEBP_WEBP_ENCODE_H_
#define WEBP_WEBP_ENCODE_H_
#include "./types.h"
#ifdef __cplusplus
extern "C" {
#endif
#define WEBP_ENCODER_ABI_VERSION 0x0209 // MAJOR(8b) + MINOR(8b)
// Note: forward declaring enumerations is not allowed in (strict) C and C++,
// the types are left here for reference.
// typedef enum WebPImageHint WebPImageHint;
// typedef enum WebPEncCSP WebPEncCSP;
// typedef enum WebPPreset WebPPreset;
// typedef enum WebPEncodingError WebPEncodingError;
typedef struct WebPConfig WebPConfig;
typedef struct WebPPicture WebPPicture; // main structure for I/O
typedef struct WebPAuxStats WebPAuxStats;
typedef struct WebPMemoryWriter WebPMemoryWriter;
// Return the encoder's version number, packed in hexadecimal using 8bits for
// each of major/minor/revision. E.g: v2.5.7 is 0x020507.
WEBP_EXTERN(int) WebPGetEncoderVersion(void);
//------------------------------------------------------------------------------
// One-stop-shop call! No questions asked:
// Returns the size of the compressed data (pointed to by *output), or 0 if
// an error occurred. The compressed data must be released by the caller
// using the call 'WebPFree(*output)'.
// These functions compress using the lossy format, and the quality_factor
// can go from 0 (smaller output, lower quality) to 100 (best quality,
// larger output).
WEBP_EXTERN(size_t) WebPEncodeRGB(const uint8_t* rgb,
int width, int height, int stride,
float quality_factor, uint8_t** output);
WEBP_EXTERN(size_t) WebPEncodeBGR(const uint8_t* bgr,
int width, int height, int stride,
float quality_factor, uint8_t** output);
WEBP_EXTERN(size_t) WebPEncodeRGBA(const uint8_t* rgba,
int width, int height, int stride,
float quality_factor, uint8_t** output);
WEBP_EXTERN(size_t) WebPEncodeBGRA(const uint8_t* bgra,
int width, int height, int stride,
float quality_factor, uint8_t** output);
// These functions are the equivalent of the above, but compressing in a
// lossless manner. Files are usually larger than lossy format, but will
// not suffer any compression loss.
WEBP_EXTERN(size_t) WebPEncodeLosslessRGB(const uint8_t* rgb,
int width, int height, int stride,
uint8_t** output);
WEBP_EXTERN(size_t) WebPEncodeLosslessBGR(const uint8_t* bgr,
int width, int height, int stride,
uint8_t** output);
WEBP_EXTERN(size_t) WebPEncodeLosslessRGBA(const uint8_t* rgba,
int width, int height, int stride,
uint8_t** output);
WEBP_EXTERN(size_t) WebPEncodeLosslessBGRA(const uint8_t* bgra,
int width, int height, int stride,
uint8_t** output);
// Releases memory returned by the WebPEncode*() functions above.
WEBP_EXTERN(void) WebPFree(void* ptr);
//------------------------------------------------------------------------------
// Coding parameters
// Image characteristics hint for the underlying encoder.
typedef enum WebPImageHint {
WEBP_HINT_DEFAULT = 0, // default preset.
WEBP_HINT_PICTURE, // digital picture, like portrait, inner shot
WEBP_HINT_PHOTO, // outdoor photograph, with natural lighting
WEBP_HINT_GRAPH, // Discrete tone image (graph, map-tile etc).
WEBP_HINT_LAST
} WebPImageHint;
// Compression parameters.
struct WebPConfig {
int lossless; // Lossless encoding (0=lossy(default), 1=lossless).
float quality; // between 0 (smallest file) and 100 (biggest)
int method; // quality/speed trade-off (0=fast, 6=slower-better)
WebPImageHint image_hint; // Hint for image type (lossless only for now).
// Parameters related to lossy compression only:
int target_size; // if non-zero, set the desired target size in bytes.
// Takes precedence over the 'compression' parameter.
float target_PSNR; // if non-zero, specifies the minimal distortion to
// try to achieve. Takes precedence over target_size.
int segments; // maximum number of segments to use, in [1..4]
int sns_strength; // Spatial Noise Shaping. 0=off, 100=maximum.
int filter_strength; // range: [0 = off .. 100 = strongest]
int filter_sharpness; // range: [0 = off .. 7 = least sharp]
int filter_type; // filtering type: 0 = simple, 1 = strong (only used
// if filter_strength > 0 or autofilter > 0)
int autofilter; // Auto adjust filter's strength [0 = off, 1 = on]
int alpha_compression; // Algorithm for encoding the alpha plane (0 = none,
// 1 = compressed with WebP lossless). Default is 1.
int alpha_filtering; // Predictive filtering method for alpha plane.
// 0: none, 1: fast, 2: best. Default if 1.
int alpha_quality; // Between 0 (smallest size) and 100 (lossless).
// Default is 100.
int pass; // number of entropy-analysis passes (in [1..10]).
int show_compressed; // if true, export the compressed picture back.
// In-loop filtering is not applied.
int preprocessing; // preprocessing filter:
// 0=none, 1=segment-smooth, 2=pseudo-random dithering
int partitions; // log2(number of token partitions) in [0..3]. Default
// is set to 0 for easier progressive decoding.
int partition_limit; // quality degradation allowed to fit the 512k limit
// on prediction modes coding (0: no degradation,
// 100: maximum possible degradation).
int emulate_jpeg_size; // If true, compression parameters will be remapped
// to better match the expected output size from
// JPEG compression. Generally, the output size will
// be similar but the degradation will be lower.
int thread_level; // If non-zero, try and use multi-threaded encoding.
int low_memory; // If set, reduce memory usage (but increase CPU use).
int near_lossless; // Near lossless encoding [0 = max loss .. 100 = off
// (default)].
int exact; // if non-zero, preserve the exact RGB values under
// transparent area. Otherwise, discard this invisible
// RGB information for better compression. The default
// value is 0.
#ifdef WEBP_EXPERIMENTAL_FEATURES
int delta_palettization;
uint32_t pad[2]; // padding for later use
#else
uint32_t pad[3]; // padding for later use
#endif // WEBP_EXPERIMENTAL_FEATURES
};
// Enumerate some predefined settings for WebPConfig, depending on the type
// of source picture. These presets are used when calling WebPConfigPreset().
typedef enum WebPPreset {
WEBP_PRESET_DEFAULT = 0, // default preset.
WEBP_PRESET_PICTURE, // digital picture, like portrait, inner shot
WEBP_PRESET_PHOTO, // outdoor photograph, with natural lighting
WEBP_PRESET_DRAWING, // hand or line drawing, with high-contrast details
WEBP_PRESET_ICON, // small-sized colorful images
WEBP_PRESET_TEXT // text-like
} WebPPreset;
// Internal, version-checked, entry point
WEBP_EXTERN(int) WebPConfigInitInternal(WebPConfig*, WebPPreset, float, int);
// Should always be called, to initialize a fresh WebPConfig structure before
// modification. Returns false in case of version mismatch. WebPConfigInit()
// must have succeeded before using the 'config' object.
// Note that the default values are lossless=0 and quality=75.
static WEBP_INLINE int WebPConfigInit(WebPConfig* config) {
return WebPConfigInitInternal(config, WEBP_PRESET_DEFAULT, 75.f,
WEBP_ENCODER_ABI_VERSION);
}
// This function will initialize the configuration according to a predefined
// set of parameters (referred to by 'preset') and a given quality factor.
// This function can be called as a replacement to WebPConfigInit(). Will
// return false in case of error.
static WEBP_INLINE int WebPConfigPreset(WebPConfig* config,
WebPPreset preset, float quality) {
return WebPConfigInitInternal(config, preset, quality,
WEBP_ENCODER_ABI_VERSION);
}
// Activate the lossless compression mode with the desired efficiency level
// between 0 (fastest, lowest compression) and 9 (slower, best compression).
// A good default level is '6', providing a fair tradeoff between compression
// speed and final compressed size.
// This function will overwrite several fields from config: 'method', 'quality'
// and 'lossless'. Returns false in case of parameter error.
WEBP_EXTERN(int) WebPConfigLosslessPreset(WebPConfig* config, int level);
// Returns true if 'config' is non-NULL and all configuration parameters are
// within their valid ranges.
WEBP_EXTERN(int) WebPValidateConfig(const WebPConfig* config);
//------------------------------------------------------------------------------
// Input / Output
// Structure for storing auxiliary statistics (mostly for lossy encoding).
struct WebPAuxStats {
int coded_size; // final size
float PSNR[5]; // peak-signal-to-noise ratio for Y/U/V/All/Alpha
int block_count[3]; // number of intra4/intra16/skipped macroblocks
int header_bytes[2]; // approximate number of bytes spent for header
// and mode-partition #0
int residual_bytes[3][4]; // approximate number of bytes spent for
// DC/AC/uv coefficients for each (0..3) segments.
int segment_size[4]; // number of macroblocks in each segments
int segment_quant[4]; // quantizer values for each segments
int segment_level[4]; // filtering strength for each segments [0..63]
int alpha_data_size; // size of the transparency data
int layer_data_size; // size of the enhancement layer data
// lossless encoder statistics
uint32_t lossless_features; // bit0:predictor bit1:cross-color transform
// bit2:subtract-green bit3:color indexing
int histogram_bits; // number of precision bits of histogram
int transform_bits; // precision bits for transform
int cache_bits; // number of bits for color cache lookup
int palette_size; // number of color in palette, if used
int lossless_size; // final lossless size
int lossless_hdr_size; // lossless header (transform, huffman etc) size
int lossless_data_size; // lossless image data size
uint32_t pad[2]; // padding for later use
};
// Signature for output function. Should return true if writing was successful.
// data/data_size is the segment of data to write, and 'picture' is for
// reference (and so one can make use of picture->custom_ptr).
typedef int (*WebPWriterFunction)(const uint8_t* data, size_t data_size,
const WebPPicture* picture);
// WebPMemoryWrite: a special WebPWriterFunction that writes to memory using
// the following WebPMemoryWriter object (to be set as a custom_ptr).
struct WebPMemoryWriter {
uint8_t* mem; // final buffer (of size 'max_size', larger than 'size').
size_t size; // final size
size_t max_size; // total capacity
uint32_t pad[1]; // padding for later use
};
// The following must be called first before any use.
WEBP_EXTERN(void) WebPMemoryWriterInit(WebPMemoryWriter* writer);
// The following must be called to deallocate writer->mem memory. The 'writer'
// object itself is not deallocated.
WEBP_EXTERN(void) WebPMemoryWriterClear(WebPMemoryWriter* writer);
// The custom writer to be used with WebPMemoryWriter as custom_ptr. Upon
// completion, writer.mem and writer.size will hold the coded data.
// writer.mem must be freed by calling WebPMemoryWriterClear.
WEBP_EXTERN(int) WebPMemoryWrite(const uint8_t* data, size_t data_size,
const WebPPicture* picture);
// Progress hook, called from time to time to report progress. It can return
// false to request an abort of the encoding process, or true otherwise if
// everything is OK.
typedef int (*WebPProgressHook)(int percent, const WebPPicture* picture);
// Color spaces.
typedef enum WebPEncCSP {
// chroma sampling
WEBP_YUV420 = 0, // 4:2:0
WEBP_YUV420A = 4, // alpha channel variant
WEBP_CSP_UV_MASK = 3, // bit-mask to get the UV sampling factors
WEBP_CSP_ALPHA_BIT = 4 // bit that is set if alpha is present
} WebPEncCSP;
// Encoding error conditions.
typedef enum WebPEncodingError {
VP8_ENC_OK = 0,
VP8_ENC_ERROR_OUT_OF_MEMORY, // memory error allocating objects
VP8_ENC_ERROR_BITSTREAM_OUT_OF_MEMORY, // memory error while flushing bits
VP8_ENC_ERROR_NULL_PARAMETER, // a pointer parameter is NULL
VP8_ENC_ERROR_INVALID_CONFIGURATION, // configuration is invalid
VP8_ENC_ERROR_BAD_DIMENSION, // picture has invalid width/height
VP8_ENC_ERROR_PARTITION0_OVERFLOW, // partition is bigger than 512k
VP8_ENC_ERROR_PARTITION_OVERFLOW, // partition is bigger than 16M
VP8_ENC_ERROR_BAD_WRITE, // error while flushing bytes
VP8_ENC_ERROR_FILE_TOO_BIG, // file is bigger than 4G
VP8_ENC_ERROR_USER_ABORT, // abort request by user
VP8_ENC_ERROR_LAST // list terminator. always last.
} WebPEncodingError;
// maximum width/height allowed (inclusive), in pixels
#define WEBP_MAX_DIMENSION 16383
// Main exchange structure (input samples, output bytes, statistics)
struct WebPPicture {
// INPUT
//////////////
// Main flag for encoder selecting between ARGB or YUV input.
// It is recommended to use ARGB input (*argb, argb_stride) for lossless
// compression, and YUV input (*y, *u, *v, etc.) for lossy compression
// since these are the respective native colorspace for these formats.
int use_argb;
// YUV input (mostly used for input to lossy compression)
WebPEncCSP colorspace; // colorspace: should be YUV420 for now (=Y'CbCr).
int width, height; // dimensions (less or equal to WEBP_MAX_DIMENSION)
uint8_t *y, *u, *v; // pointers to luma/chroma planes.
int y_stride, uv_stride; // luma/chroma strides.
uint8_t* a; // pointer to the alpha plane
int a_stride; // stride of the alpha plane
uint32_t pad1[2]; // padding for later use
// ARGB input (mostly used for input to lossless compression)
uint32_t* argb; // Pointer to argb (32 bit) plane.
int argb_stride; // This is stride in pixels units, not bytes.
uint32_t pad2[3]; // padding for later use
// OUTPUT
///////////////
// Byte-emission hook, to store compressed bytes as they are ready.
WebPWriterFunction writer; // can be NULL
void* custom_ptr; // can be used by the writer.
// map for extra information (only for lossy compression mode)
int extra_info_type; // 1: intra type, 2: segment, 3: quant
// 4: intra-16 prediction mode,
// 5: chroma prediction mode,
// 6: bit cost, 7: distortion
uint8_t* extra_info; // if not NULL, points to an array of size
// ((width + 15) / 16) * ((height + 15) / 16) that
// will be filled with a macroblock map, depending
// on extra_info_type.
// STATS AND REPORTS
///////////////////////////
// Pointer to side statistics (updated only if not NULL)
WebPAuxStats* stats;
// Error code for the latest error encountered during encoding
WebPEncodingError error_code;
// If not NULL, report progress during encoding.
WebPProgressHook progress_hook;
void* user_data; // this field is free to be set to any value and
// used during callbacks (like progress-report e.g.).
uint32_t pad3[3]; // padding for later use
// Unused for now
uint8_t *pad4, *pad5;
uint32_t pad6[8]; // padding for later use
// PRIVATE FIELDS
////////////////////
void* memory_; // row chunk of memory for yuva planes
void* memory_argb_; // and for argb too.
void* pad7[2]; // padding for later use
};
// Internal, version-checked, entry point
WEBP_EXTERN(int) WebPPictureInitInternal(WebPPicture*, int);
// Should always be called, to initialize the structure. Returns false in case
// of version mismatch. WebPPictureInit() must have succeeded before using the
// 'picture' object.
// Note that, by default, use_argb is false and colorspace is WEBP_YUV420.
static WEBP_INLINE int WebPPictureInit(WebPPicture* picture) {
return WebPPictureInitInternal(picture, WEBP_ENCODER_ABI_VERSION);
}
//------------------------------------------------------------------------------
// WebPPicture utils
// Convenience allocation / deallocation based on picture->width/height:
// Allocate y/u/v buffers as per colorspace/width/height specification.
// Note! This function will free the previous buffer if needed.
// Returns false in case of memory error.
WEBP_EXTERN(int) WebPPictureAlloc(WebPPicture* picture);
// Release the memory allocated by WebPPictureAlloc() or WebPPictureImport*().
// Note that this function does _not_ free the memory used by the 'picture'
// object itself.
// Besides memory (which is reclaimed) all other fields of 'picture' are
// preserved.
WEBP_EXTERN(void) WebPPictureFree(WebPPicture* picture);
// Copy the pixels of *src into *dst, using WebPPictureAlloc. Upon return, *dst
// will fully own the copied pixels (this is not a view). The 'dst' picture need
// not be initialized as its content is overwritten.
// Returns false in case of memory allocation error.
WEBP_EXTERN(int) WebPPictureCopy(const WebPPicture* src, WebPPicture* dst);
// Compute PSNR, SSIM or LSIM distortion metric between two pictures. Results
// are in dB, stored in result[] in the Y/U/V/Alpha/All or B/G/R/A/All order.
// Returns false in case of error (src and ref don't have same dimension, ...)
// Warning: this function is rather CPU-intensive.
WEBP_EXTERN(int) WebPPictureDistortion(
const WebPPicture* src, const WebPPicture* ref,
int metric_type, // 0 = PSNR, 1 = SSIM, 2 = LSIM
float result[5]);
// self-crops a picture to the rectangle defined by top/left/width/height.
// Returns false in case of memory allocation error, or if the rectangle is
// outside of the source picture.
// The rectangle for the view is defined by the top-left corner pixel
// coordinates (left, top) as well as its width and height. This rectangle
// must be fully be comprised inside the 'src' source picture. If the source
// picture uses the YUV420 colorspace, the top and left coordinates will be
// snapped to even values.
WEBP_EXTERN(int) WebPPictureCrop(WebPPicture* picture,
int left, int top, int width, int height);
// Extracts a view from 'src' picture into 'dst'. The rectangle for the view
// is defined by the top-left corner pixel coordinates (left, top) as well
// as its width and height. This rectangle must be fully be comprised inside
// the 'src' source picture. If the source picture uses the YUV420 colorspace,
// the top and left coordinates will be snapped to even values.
// Picture 'src' must out-live 'dst' picture. Self-extraction of view is allowed
// ('src' equal to 'dst') as a mean of fast-cropping (but note that doing so,
// the original dimension will be lost). Picture 'dst' need not be initialized
// with WebPPictureInit() if it is different from 'src', since its content will
// be overwritten.
// Returns false in case of memory allocation error or invalid parameters.
WEBP_EXTERN(int) WebPPictureView(const WebPPicture* src,
int left, int top, int width, int height,
WebPPicture* dst);
// Returns true if the 'picture' is actually a view and therefore does
// not own the memory for pixels.
WEBP_EXTERN(int) WebPPictureIsView(const WebPPicture* picture);
// Rescale a picture to new dimension width x height.
// If either 'width' or 'height' (but not both) is 0 the corresponding
// dimension will be calculated preserving the aspect ratio.
// No gamma correction is applied.
// Returns false in case of error (invalid parameter or insufficient memory).
WEBP_EXTERN(int) WebPPictureRescale(WebPPicture* pic, int width, int height);
// Colorspace conversion function to import RGB samples.
// Previous buffer will be free'd, if any.
// *rgb buffer should have a size of at least height * rgb_stride.
// Returns false in case of memory error.
WEBP_EXTERN(int) WebPPictureImportRGB(
WebPPicture* picture, const uint8_t* rgb, int rgb_stride);
// Same, but for RGBA buffer.
WEBP_EXTERN(int) WebPPictureImportRGBA(
WebPPicture* picture, const uint8_t* rgba, int rgba_stride);
// Same, but for RGBA buffer. Imports the RGB direct from the 32-bit format
// input buffer ignoring the alpha channel. Avoids needing to copy the data
// to a temporary 24-bit RGB buffer to import the RGB only.
WEBP_EXTERN(int) WebPPictureImportRGBX(
WebPPicture* picture, const uint8_t* rgbx, int rgbx_stride);
// Variants of the above, but taking BGR(A|X) input.
WEBP_EXTERN(int) WebPPictureImportBGR(
WebPPicture* picture, const uint8_t* bgr, int bgr_stride);
WEBP_EXTERN(int) WebPPictureImportBGRA(
WebPPicture* picture, const uint8_t* bgra, int bgra_stride);
WEBP_EXTERN(int) WebPPictureImportBGRX(
WebPPicture* picture, const uint8_t* bgrx, int bgrx_stride);
// Converts picture->argb data to the YUV420A format. The 'colorspace'
// parameter is deprecated and should be equal to WEBP_YUV420.
// Upon return, picture->use_argb is set to false. The presence of real
// non-opaque transparent values is detected, and 'colorspace' will be
// adjusted accordingly. Note that this method is lossy.
// Returns false in case of error.
WEBP_EXTERN(int) WebPPictureARGBToYUVA(WebPPicture* picture,
WebPEncCSP /*colorspace = WEBP_YUV420*/);
// Same as WebPPictureARGBToYUVA(), but the conversion is done using
// pseudo-random dithering with a strength 'dithering' between
// 0.0 (no dithering) and 1.0 (maximum dithering). This is useful
// for photographic picture.
WEBP_EXTERN(int) WebPPictureARGBToYUVADithered(
WebPPicture* picture, WebPEncCSP colorspace, float dithering);
// Performs 'smart' RGBA->YUVA420 downsampling and colorspace conversion.
// Downsampling is handled with extra care in case of color clipping. This
// method is roughly 2x slower than WebPPictureARGBToYUVA() but produces better
// YUV representation.
// Returns false in case of error.
WEBP_EXTERN(int) WebPPictureSmartARGBToYUVA(WebPPicture* picture);
// Converts picture->yuv to picture->argb and sets picture->use_argb to true.
// The input format must be YUV_420 or YUV_420A.
// Note that the use of this method is discouraged if one has access to the
// raw ARGB samples, since using YUV420 is comparatively lossy. Also, the
// conversion from YUV420 to ARGB incurs a small loss too.
// Returns false in case of error.
WEBP_EXTERN(int) WebPPictureYUVAToARGB(WebPPicture* picture);
// Helper function: given a width x height plane of RGBA or YUV(A) samples
// clean-up the YUV or RGB samples under fully transparent area, to help
// compressibility (no guarantee, though).
WEBP_EXTERN(void) WebPCleanupTransparentArea(WebPPicture* picture);
// Scan the picture 'picture' for the presence of non fully opaque alpha values.
// Returns true in such case. Otherwise returns false (indicating that the
// alpha plane can be ignored altogether e.g.).
WEBP_EXTERN(int) WebPPictureHasTransparency(const WebPPicture* picture);
// Remove the transparency information (if present) by blending the color with
// the background color 'background_rgb' (specified as 24bit RGB triplet).
// After this call, all alpha values are reset to 0xff.
WEBP_EXTERN(void) WebPBlendAlpha(WebPPicture* pic, uint32_t background_rgb);
//------------------------------------------------------------------------------
// Main call
// Main encoding call, after config and picture have been initialized.
// 'picture' must be less than 16384x16384 in dimension (cf WEBP_MAX_DIMENSION),
// and the 'config' object must be a valid one.
// Returns false in case of error, true otherwise.
// In case of error, picture->error_code is updated accordingly.
// 'picture' can hold the source samples in both YUV(A) or ARGB input, depending
// on the value of 'picture->use_argb'. It is highly recommended to use
// the former for lossy encoding, and the latter for lossless encoding
// (when config.lossless is true). Automatic conversion from one format to
// another is provided but they both incur some loss.
WEBP_EXTERN(int) WebPEncode(const WebPConfig* config, WebPPicture* picture);
//------------------------------------------------------------------------------
#ifdef __cplusplus
} // extern "C"
#endif
#endif /* WEBP_WEBP_ENCODE_H_ */

52
submodules/webp/include/webp/types.h
View File

@ -1,52 +0,0 @@
// Copyright 2010 Google Inc. All Rights Reserved.
//
// Use of this source code is governed by a BSD-style license
// that can be found in the COPYING file in the root of the source
// tree. An additional intellectual property rights grant can be found
// in the file PATENTS. All contributing project authors may
// be found in the AUTHORS file in the root of the source tree.
// -----------------------------------------------------------------------------
//
// Common types
//
// Author: Skal (pascal.massimino@gmail.com)
#ifndef WEBP_WEBP_TYPES_H_
#define WEBP_WEBP_TYPES_H_
#include <stddef.h> // for size_t
#ifndef _MSC_VER
#include <inttypes.h>
#if defined(__cplusplus) || !defined(__STRICT_ANSI__) || \
(defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L)
#define WEBP_INLINE inline
#else
#define WEBP_INLINE
#endif
#else
typedef signed char int8_t;
typedef unsigned char uint8_t;
typedef signed short int16_t;
typedef unsigned short uint16_t;
typedef signed int int32_t;
typedef unsigned int uint32_t;
typedef unsigned long long int uint64_t;
typedef long long int int64_t;
#define WEBP_INLINE __forceinline
#endif /* _MSC_VER */
#ifndef WEBP_EXTERN
// This explicitly marks library functions and allows for changing the
// signature for e.g., Windows DLL builds.
# if defined(__GNUC__) && __GNUC__ >= 4
# define WEBP_EXTERN(type) extern __attribute__ ((visibility ("default"))) type
# else
# define WEBP_EXTERN(type) extern type
# endif /* __GNUC__ >= 4 */
#endif /* WEBP_EXTERN */
// Macro to check ABI compatibility (same major revision number)
#define WEBP_ABI_IS_INCOMPATIBLE(a, b) (((a) >> 8) != ((b) >> 8))
#endif /* WEBP_WEBP_TYPES_H_ */

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submodules/webp/lib/libwebp.a
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10
third-party/boringssl/BUILD vendored
View File

@ -61,15 +61,15 @@ posix_copts = [
boringssl_copts = select({
"@build_bazel_rules_apple//apple:ios_armv7": posix_copts,
"@build_bazel_rules_apple//apple:ios_arm64": posix_copts,
"//build-system:ios_sim_arm64": posix_copts,
"@build_bazel_rules_apple//apple:ios_x86_64": posix_copts,
"//conditions:default": ["-DOPENSSL_NO_ASM"],
})
crypto_sources_asm = select({
"@build_bazel_rules_apple//apple:ios_armv7": crypto_sources_ios_arm,
"@build_bazel_rules_apple//apple:ios_arm64": crypto_sources_ios_aarch64,
"//build-system:ios_sim_arm64": crypto_sources_ios_aarch64,
"@build_bazel_rules_apple//apple:ios_x86_64": crypto_sources_mac_x86_64,
"//conditions:default": [],
})
# For C targets only (not C++), compile with C11 support.
@ -83,8 +83,8 @@ posix_copts_c11 = [
boringssl_copts_c11 = boringssl_copts + select({
"@build_bazel_rules_apple//apple:ios_armv7": posix_copts_c11,
"@build_bazel_rules_apple//apple:ios_arm64": posix_copts_c11,
"//build-system:ios_sim_arm64": posix_copts_c11,
"@build_bazel_rules_apple//apple:ios_x86_64": posix_copts_c11,
"//conditions:default": [],
})
# For C++ targets only (not C), compile with C++11 support.
@ -96,8 +96,8 @@ posix_copts_cxx = [
boringssl_copts_cxx = boringssl_copts + select({
"@build_bazel_rules_apple//apple:ios_armv7": posix_copts_cxx,
"@build_bazel_rules_apple//apple:ios_arm64": posix_copts_cxx,
"//build-system:ios_sim_arm64": posix_copts_cxx,
"@build_bazel_rules_apple//apple:ios_x86_64": posix_copts_cxx,
"//conditions:default": [],
})
cc_library(
@ -109,8 +109,8 @@ cc_library(
linkopts = select({
"@build_bazel_rules_apple//apple:ios_armv7": [],
"@build_bazel_rules_apple//apple:ios_arm64": [],
"//build-system:ios_sim_arm64": [],
"@build_bazel_rules_apple//apple:ios_x86_64": [],
"//conditions:default": ["-lpthread"],
}),
linkstatic = 1,
visibility = ["//visibility:public"],

132
third-party/libvpx/0001-Add-support-for-arm64-iphonesimulator-gcc.patch vendored Normal file
View File

@ -0,0 +1,132 @@
From 654c44d26bd86efec4019990beac67900231f868 Mon Sep 17 00:00:00 2001
From: Ali <>
Date: Wed, 16 Dec 2020 17:11:29 +0400
Subject: [PATCH] Add support for arm64-iphonesimulator-gcc
---
build/make/configure.sh | 73 ++++++++++++++++++++++++++++++++++++++++-
configure | 3 +-
2 files changed, 74 insertions(+), 2 deletions(-)
diff --git a/build/make/configure.sh b/build/make/configure.sh
index 206b54f77..1c113e51d 100644
--- a/build/make/configure.sh
+++ b/build/make/configure.sh
@@ -845,6 +845,14 @@ process_common_toolchain() {
# Handle darwin variants. Newer SDKs allow targeting older
# platforms, so use the newest one available.
case ${toolchain} in
+ arm64-iphonesimulator*)
+ add_cflags "-miphonesimulator-version-min=${IOS_VERSION_MIN}"
+ iphonesimulator_sdk_dir="$(show_darwin_sdk_path iphonesimulator)"
+ if [ -d "${iphonesimulator_sdk_dir}" ]; then
+ add_cflags "-isysroot ${iphonesimulator_sdk_dir}"
+ add_ldflags "-isysroot ${iphonesimulator_sdk_dir}"
+ fi
+ ;;
arm*-darwin*)
add_cflags "-miphoneos-version-min=${IOS_VERSION_MIN}"
iphoneos_sdk_dir="$(show_darwin_sdk_path iphoneos)"
@@ -934,7 +942,7 @@ process_common_toolchain() {
# Process ARM architecture variants
case ${toolchain} in
- arm*)
+ arm*|arm64-iphonesimulator-gcc)
# on arm, isa versions are supersets
case ${tgt_isa} in
arm64|armv8)
@@ -1144,6 +1152,69 @@ EOF
asm_conversion_cmd="${source_path}/build/make/ads2gas_apple.pl"
;;
+ iphonesimulator*)
+ if ! enabled external_build; then
+ XCRUN_FIND="xcrun --sdk iphonesimulator --find"
+ CXX="$(${XCRUN_FIND} clang++)"
+ CC="$(${XCRUN_FIND} clang)"
+ AR="$(${XCRUN_FIND} ar)"
+ AS="$(${XCRUN_FIND} as)"
+ STRIP="$(${XCRUN_FIND} strip)"
+ NM="$(${XCRUN_FIND} nm)"
+ RANLIB="$(${XCRUN_FIND} ranlib)"
+ AS_SFX=.S
+ LD="${CXX:-$(${XCRUN_FIND} ld)}"
+
+ # ASFLAGS is written here instead of using check_add_asflags
+ # because we need to overwrite all of ASFLAGS and purge the
+ # options that were put in above
+ ASFLAGS="-arch ${tgt_isa} -g"
+
+ add_cflags -arch ${tgt_isa}
+ add_ldflags -arch ${tgt_isa}
+
+ add_cflags --target=arm64-apple-ios7.0-simulator
+ add_ldflags --target=arm64-apple-ios7.0-simulator
+
+ alt_libc="$(show_darwin_sdk_path iphonesimulator)"
+ if [ -d "${alt_libc}" ]; then
+ add_cflags -isysroot ${alt_libc}
+ fi
+
+ if [ "${LD}" = "${CXX}" ]; then
+ add_ldflags -miphonesimulator-version-min="${IOS_VERSION_MIN}"
+ else
+ add_ldflags -ios_version_min "${IOS_VERSION_MIN}"
+ fi
+
+ for d in lib usr/lib usr/lib/system; do
+ try_dir="${alt_libc}/${d}"
+ [ -d "${try_dir}" ] && add_ldflags -L"${try_dir}"
+ done
+
+ case ${tgt_isa} in
+ armv7|armv7s|armv8|arm64)
+ if enabled neon && ! check_xcode_minimum_version; then
+ soft_disable neon
+ log_echo " neon disabled: upgrade Xcode (need v6.3+)."
+ if enabled neon_asm; then
+ soft_disable neon_asm
+ log_echo " neon_asm disabled: upgrade Xcode (need v6.3+)."
+ fi
+ fi
+ ;;
+ esac
+
+ if [ "$(show_darwin_sdk_major_version iphoneos)" -gt 8 ]; then
+ check_add_cflags -fembed-bitcode
+ check_add_asflags -fembed-bitcode
+ check_add_ldflags -fembed-bitcode
+ fi
+ fi
+
+ asm_conversion_cmd="${source_path}/build/make/ads2gas_apple.pl"
+ ;;
+
linux*)
enable_feature linux
if enabled rvct; then
diff --git a/configure b/configure
index 32272ce36..826769948 100755
--- a/configure
+++ b/configure
@@ -99,6 +99,7 @@ EOF
# alphabetically by architecture, generic-gnu last.
all_platforms="${all_platforms} arm64-android-gcc"
all_platforms="${all_platforms} arm64-darwin-gcc"
+all_platforms="${all_platforms} arm64-iphonesimulator-gcc"
all_platforms="${all_platforms} arm64-linux-gcc"
all_platforms="${all_platforms} arm64-win64-gcc"
all_platforms="${all_platforms} arm64-win64-vs15"
@@ -733,7 +734,7 @@ process_toolchain() {
soft_enable libyuv
# GTestLog must be modified to use Android logging utilities.
;;
- *-darwin-*)
+ *-darwin-*|arm64-iphonesimulator-*)
check_add_cxxflags -std=c++11
# iOS/ARM builds do not work with gtest. This does not match
# x86 targets.
--
2.24.3 (Apple Git-128)

11
third-party/libvpx/BUILD vendored
View File

@ -28,6 +28,7 @@ genrule(
name = "libvpx_build",
srcs = [
"build-libvpx-bazel.sh",
"0001-Add-support-for-arm64-iphonesimulator-gcc.patch",
":libvpx_sources",
],
cmd_bash =
@ -40,6 +41,9 @@ genrule(
elif [ "$(TARGET_CPU)" == "ios_arm64" ]; then
BUILD_ARCH="arm64"
PLATFORM_HEADER_DIR="arm64-darwin-gcc"
elif [ "$(TARGET_CPU)" == "ios_sim_arm64" ]; then
BUILD_ARCH="sim_arm64"
PLATFORM_HEADER_DIR="arm64-iphonesimulator-gcc"
elif [ "$(TARGET_CPU)" == "ios_x86_64" ]; then
BUILD_ARCH="x86_64"
PLATFORM_HEADER_DIR="x86_64-iphonesimulator-gcc"
@ -52,14 +56,19 @@ genrule(
mkdir -p "$$BUILD_DIR"
cp $(location :build-libvpx-bazel.sh) "$$BUILD_DIR/"
cp $(location :0001-Add-support-for-arm64-iphonesimulator-gcc.patch) "$$BUILD_DIR/"
SOURCE_PATH="third-party/libvpx/libvpx"
cp -R "$$SOURCE_PATH" "$$BUILD_DIR/"
pushd "$$BUILD_DIR/libvpx"
patch -p1 < ../0001-Add-support-for-arm64-iphonesimulator-gcc.patch
popd
mkdir -p "$$BUILD_DIR/Public/libvpx"
arch -x86_64 sh $$BUILD_DIR/build-libvpx-bazel.sh $$BUILD_ARCH "$$BUILD_DIR/libvpx" "$$BUILD_DIR"
sh $$BUILD_DIR/build-libvpx-bazel.sh $$BUILD_ARCH "$$BUILD_DIR/libvpx" "$$BUILD_DIR"
""" +
"\n".join([
"cp -f \"$$BUILD_DIR/VPX.framework/Headers/vpx/{}\" \"$(location Public/vpx/{})\"".format(header, header) for header in headers

6
third-party/libvpx/build-libvpx-bazel.sh vendored
View File

@ -34,6 +34,8 @@ if [ "$ARCH" = "armv7" ]; then
TARGETS="armv7-darwin-gcc"
elif [ "$ARCH" = "arm64" ]; then
TARGETS="arm64-darwin-gcc"
elif [ "$ARCH" = "sim_arm64" ]; then
TARGETS="arm64-iphonesimulator-gcc"
elif [ "$ARCH" = "x86_64" ]; then
TARGETS="x86_64-iphonesimulator-gcc"
else
@ -58,9 +60,9 @@ build_target() {
cd "${target}"
eval "${LIBVPX_SOURCE_DIR}/configure" --target="${target}" \
${CONFIGURE_ARGS} ${EXTRA_CONFIGURE_ARGS} ${target_specific_flags} \
${devnull}
export DIST_DIR
eval make dist ${devnull}
eval make dist
cd "${old_pwd}"
}

4
third-party/mozjpeg/BUILD vendored
View File

@ -32,6 +32,8 @@ genrule(
BUILD_ARCH="armv7"
elif [ "$(TARGET_CPU)" == "ios_arm64" ]; then
BUILD_ARCH="arm64"
elif [ "$(TARGET_CPU)" == "ios_sim_arm64" ]; then
BUILD_ARCH="sim_arm64"
elif [ "$(TARGET_CPU)" == "ios_x86_64" ]; then
BUILD_ARCH="x86_64"
else
@ -55,7 +57,7 @@ genrule(
mkdir -p "$$BUILD_DIR/Public/mozjpeg"
PATH="$$PATH:$$CMAKE_DIR/cmake-3.18.4-Darwin-x86_64/CMake.app/Contents/bin" arch -x86_64 sh $$BUILD_DIR/build-mozjpeg-bazel.sh $$BUILD_ARCH "$$BUILD_DIR/mozjpeg" "$$BUILD_DIR"
PATH="$$PATH:$$CMAKE_DIR/cmake-3.19.2-macos-universal/CMake.app/Contents/bin" sh $$BUILD_DIR/build-mozjpeg-bazel.sh $$BUILD_ARCH "$$BUILD_DIR/mozjpeg" "$$BUILD_DIR"
""" +
"\n".join([
"cp -f \"$$BUILD_DIR/mozjpeg/{}\" \"$(location Public/mozjpeg/{})\"".format(header, header) for header in headers

16
third-party/mozjpeg/build-mozjpeg-bazel.sh vendored
View File

@ -38,6 +38,22 @@ elif [ "$ARCH" = "arm64" ]; then
echo "set(CMAKE_SYSTEM_PROCESSOR aarch64)" >> toolchain.cmake
echo "set(CMAKE_C_COMPILER $(xcode-select -p)/Toolchains/XcodeDefault.xctoolchain/usr/bin/clang)" >> toolchain.cmake
cmake -G"Unix Makefiles" -DCMAKE_TOOLCHAIN_FILE=toolchain.cmake -DCMAKE_OSX_SYSROOT=${IOS_SYSROOT[0]} -DPNG_SUPPORTED=FALSE -DENABLE_SHARED=FALSE -DWITH_JPEG8=1 -DBUILD=10000 ../mozjpeg
make
elif [ "$ARCH" = "sim_arm64" ]; then
IOS_PLATFORMDIR="$(xcode-select -p)/Platforms/iPhoneSimulator.platform"
IOS_SYSROOT=($IOS_PLATFORMDIR/Developer/SDKs/iPhoneSimulator*.sdk)
export CFLAGS="-Wall -arch arm64 --target=arm64-apple-ios9.0-simulator -miphonesimulator-version-min=9.0 -funwind-tables"
cd "$BUILD_DIR"
mkdir build
cd build
touch toolchain.cmake
echo "set(CMAKE_SYSTEM_NAME Darwin)" >> toolchain.cmake
echo "set(CMAKE_SYSTEM_PROCESSOR aarch64)" >> toolchain.cmake
echo "set(CMAKE_C_COMPILER $(xcode-select -p)/Toolchains/XcodeDefault.xctoolchain/usr/bin/clang)" >> toolchain.cmake
cmake -G"Unix Makefiles" -DCMAKE_TOOLCHAIN_FILE=toolchain.cmake -DCMAKE_OSX_SYSROOT=${IOS_SYSROOT[0]} -DPNG_SUPPORTED=FALSE -DENABLE_SHARED=FALSE -DWITH_JPEG8=1 -DBUILD=10000 ../mozjpeg
make
elif [ "$ARCH" = "x86_64" ]; then

81
third-party/opus/BUILD vendored Normal file
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@ -0,0 +1,81 @@
headers = [
"opus.h",
"opus_defines.h",
"opus_multistream.h",
"opus_projection.h",
"opus_types.h",
]
libs = [
"opus",
]
genrule(
name = "opus_build",
srcs = [
"build-opus-bazel.sh",
"opus-1.3.1.tar.gz",
],
cmd_bash =
"""
set -ex
if [ "$(TARGET_CPU)" == "ios_armv7" ]; then
BUILD_ARCH="armv7"
elif [ "$(TARGET_CPU)" == "ios_arm64" ]; then
BUILD_ARCH="arm64"
elif [ "$(TARGET_CPU)" == "ios_sim_arm64" ]; then
BUILD_ARCH="sim_arm64"
elif [ "$(TARGET_CPU)" == "ios_x86_64" ]; then
BUILD_ARCH="x86_64"
else
echo "Unsupported architecture $(TARGET_CPU)"
fi
BUILD_DIR="$(RULEDIR)/build_$${BUILD_ARCH}"
rm -rf "$$BUILD_DIR"
mkdir -p "$$BUILD_DIR"
cp $(location :build-opus-bazel.sh) "$$BUILD_DIR/"
cp $(location :opus-1.3.1.tar.gz) "$$BUILD_DIR/"
mkdir -p "$$BUILD_DIR/Public/opus"
sh $$BUILD_DIR/build-opus-bazel.sh $$BUILD_ARCH "$$BUILD_DIR" opus-1.3.1.tar.gz
""" +
"\n".join([
"cp -f \"$$BUILD_DIR/built/include/opus/{}\" \"$(location Public/opus/{})\"".format(header, header) for header in headers
]) +
"\n" +
"\n".join([
"cp -f \"$$BUILD_DIR/built/lib/lib{}.a\" \"$(location Public/opus/lib/lib{}.a)\"".format(lib, lib) for lib in libs
]),
outs = ["Public/opus/" + x for x in headers] +
["Public/opus/lib/lib{}.a".format(x) for x in libs],
visibility = [
"//visibility:public",
]
)
cc_library(
name = "opus_lib",
srcs = [":Public/opus/lib/lib" + x + ".a" for x in libs],
)
objc_library(
name = "opus",
module_name = "opus",
enable_modules = True,
hdrs = [":Public/opus/" + x for x in headers],
includes = [
"Public",
"Public/opus",
],
deps = [
":opus_lib",
],
visibility = [
"//visibility:public",
],
)

55
third-party/opus/build-opus-bazel.sh vendored Executable file
View File

@ -0,0 +1,55 @@
#! /bin/sh
set -ex
ARCH="$1"
BUILD_DIR=$(echo "$(cd "$(dirname "$2")"; pwd -P)/$(basename "$2")")
SOURCE_CODE_ARCHIVE="$3"
MINIOSVERSION="9.0"
OPT_CFLAGS="-Os -g"
OPT_LDFLAGS=""
OPT_CONFIG_ARGS=""
DEVELOPER=`xcode-select -print-path`
OUTPUTDIR="$BUILD_DIR/Public"
# where we will keep our sources and build from.
SRCDIR="${BUILD_DIR}/src"
mkdir -p $SRCDIR
# where we will store intermediary builds
INTERDIR="${BUILD_DIR}/built"
mkdir -p $INTERDIR
########################################
tar zxf "$BUILD_DIR/$SOURCE_CODE_ARCHIVE" -C $SRCDIR
cd "${SRCDIR}/opus-"*
if [ "${ARCH}" == "i386" ] || [ "${ARCH}" == "x86_64" ]; then
PLATFORM="iphonesimulator"
EXTRA_CFLAGS="-arch ${ARCH}"
EXTRA_CONFIG="--host=x86_64-apple-darwin"
elif [ "${ARCH}" == "sim_arm64" ]; then
PLATFORM="iphonesimulator"
EXTRA_CFLAGS="-arch arm64 --target=arm64-apple-ios$MINIOSVERSION-simulator"
EXTRA_CONFIG="--host=arm-apple-darwin20"
else
PLATFORM="iphoneos"
EXTRA_CFLAGS="-arch ${ARCH}"
EXTRA_CONFIG="--host=arm-apple-darwin"
fi
SDK_PATH="$(xcrun --sdk $PLATFORM --show-sdk-path 2>/dev/null)"
mkdir -p "${INTERDIR}"
./configure --enable-float-approx --disable-shared --enable-static --with-pic --disable-extra-programs --disable-doc ${EXTRA_CONFIG} \
--prefix="${INTERDIR}" \
LDFLAGS="$LDFLAGS ${OPT_LDFLAGS} -fPIE -miphoneos-version-min=${MINIOSVERSION} -L${OUTPUTDIR}/lib" \
CFLAGS="$CFLAGS ${EXTRA_CFLAGS} ${OPT_CFLAGS} -fPIE -miphoneos-version-min=${MINIOSVERSION} -I${OUTPUTDIR}/include -isysroot ${SDK_PATH}" \
make -j
make install

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third-party/opus/opus-1.3.1.tar.gz vendored Normal file
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81
third-party/webp/BUILD vendored Normal file
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@ -0,0 +1,81 @@
headers = [
"decode.h",
"encode.h",
"types.h",
#"demux.h",
#"mux_types.h",
]
libs = [
"webp",
]
genrule(
name = "webp_build",
srcs = [
"build-webp-bazel.sh",
"libwebp-1.1.0.tar.gz",
],
cmd_bash =
"""
set -ex
if [ "$(TARGET_CPU)" == "ios_armv7" ]; then
BUILD_ARCH="armv7"
elif [ "$(TARGET_CPU)" == "ios_arm64" ]; then
BUILD_ARCH="arm64"
elif [ "$(TARGET_CPU)" == "ios_sim_arm64" ]; then
BUILD_ARCH="sim_arm64"
elif [ "$(TARGET_CPU)" == "ios_x86_64" ]; then
BUILD_ARCH="x86_64"
else
echo "Unsupported architecture $(TARGET_CPU)"
fi
BUILD_DIR="$(RULEDIR)/build_$${BUILD_ARCH}"
rm -rf "$$BUILD_DIR"
mkdir -p "$$BUILD_DIR"
cp $(location :build-webp-bazel.sh) "$$BUILD_DIR/"
cp $(location :libwebp-1.1.0.tar.gz) "$$BUILD_DIR/"
mkdir -p "$$BUILD_DIR/Public/webp"
sh $$BUILD_DIR/build-webp-bazel.sh $$BUILD_ARCH "$$BUILD_DIR" libwebp-1.1.0.tar.gz
""" +
"\n".join([
"cp -f \"$$BUILD_DIR/src/libwebp-1.1.0/build-output/include/webp/{}\" \"$(location Public/webp/{})\"".format(header, header) for header in headers
]) +
"\n" +
"\n".join([
"cp -f \"$$BUILD_DIR/src/libwebp-1.1.0/build-output/lib/lib{}.a\" \"$(location Public/webp/lib/lib{}.a)\"".format(lib, lib) for lib in libs
]),
outs = ["Public/webp/" + x for x in headers] +
["Public/webp/lib/lib{}.a".format(x) for x in libs],
visibility = [
"//visibility:public",
]
)
cc_library(
name = "webp_lib",
srcs = [":Public/webp/lib/lib" + x + ".a" for x in libs],
)
objc_library(
name = "webp",
module_name = "webp",
enable_modules = True,
hdrs = [":Public/webp/" + x for x in headers],
includes = [
"Public",
"Public/webp",
],
deps = [
":webp_lib",
],
visibility = [
"//visibility:public",
],
)

56
third-party/webp/build-webp-bazel.sh vendored Executable file
View File

@ -0,0 +1,56 @@
#! /bin/sh
set -ex
ARCH="$1"
BUILD_DIR=$(echo "$(cd "$(dirname "$2")"; pwd -P)/$(basename "$2")")
SOURCE_CODE_ARCHIVE="$3"
MINIOSVERSION="9.0"
OPT_CFLAGS="-Os -g"
OPT_LDFLAGS=""
OPT_CONFIG_ARGS=""
DEVELOPER=`xcode-select -print-path`
OUTPUTDIR="$BUILD_DIR/Public"
SRCDIR="${BUILD_DIR}/src"
mkdir -p $SRCDIR
tar zxf "$BUILD_DIR/$SOURCE_CODE_ARCHIVE" -C $SRCDIR
cd "${SRCDIR}/libwebp-"*
PREFIX="$(pwd)/build-output"
if [ "${ARCH}" == "i386" ] || [ "${ARCH}" == "x86_64" ]; then
PLATFORM="iphonesimulator"
EXTRA_CFLAGS="-arch ${ARCH}"
EXTRA_CONFIG="--host=x86_64-apple-darwin"
elif [ "${ARCH}" == "sim_arm64" ]; then
PLATFORM="iphonesimulator"
EXTRA_CFLAGS="-arch arm64 --target=arm64-apple-ios$MINIOSVERSION-simulator"
EXTRA_CONFIG="--host=arm-apple-darwin20"
else
PLATFORM="iphoneos"
EXTRA_CFLAGS="-arch ${ARCH}"
EXTRA_CONFIG="--host=arm-apple-darwin"
fi
SDKROOT="$(xcrun --sdk $PLATFORM --show-sdk-path 2>/dev/null)"
DEVROOT="${DEVELOPER}/Toolchains/XcodeDefault.xctoolchain"
CFLAGS="-pipe -isysroot ${SDKROOT} -O3 -DNDEBUG $EXTRA_CFLAGS"
CFLAGS+=" -miphoneos-version-min=9.0"
PATH="${DEVROOT}/usr/bin:${PATH}" ./configure \
${EXTRA_CONFIG} \
--prefix=${PREFIX} \
--build=$(./config.guess) \
--disable-shared --enable-static \
--disable-libwebpdecoder --enable-swap-16bit-csp \
CFLAGS="${CFLAGS}"
make V=0
make install

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6
third-party/webrtc/BUILD vendored
View File

@ -1766,6 +1766,7 @@ x86_specific_sources = ["webrtc-ios/src/" + path for path in [
arch_specific_sources = select({
"@build_bazel_rules_apple//apple:ios_armv7": common_arm_specific_sources + armv7_specific_sources,
"@build_bazel_rules_apple//apple:ios_arm64": common_arm_specific_sources + arm64_specific_sources,
"//build-system:ios_sim_arm64": common_arm_specific_sources + arm64_specific_sources,
"@build_bazel_rules_apple//apple:ios_x86_64": x86_specific_sources,
})
@ -1795,6 +1796,7 @@ x86_64_specific_flags = [
arch_specific_cflags = select({
"@build_bazel_rules_apple//apple:ios_armv7": common_flags + arm_specific_flags,
"@build_bazel_rules_apple//apple:ios_arm64": common_flags + arm64_specific_flags,
"//build-system:ios_sim_arm64": common_flags + arm64_specific_flags,
"@build_bazel_rules_apple//apple:ios_x86_64": common_flags + x86_64_specific_flags,
})
@ -2134,7 +2136,7 @@ objc_library(
deps = [
"//third-party/boringssl:crypto",
"//third-party/boringssl:ssl",
"//submodules/Opus:opus",
"//third-party/opus:opus",
":usrsctp",
":libsrtp",
":libevent",
@ -2176,7 +2178,7 @@ objc_library(
deps = [
"//third-party/boringssl:crypto",
"//third-party/boringssl:ssl",
"//submodules/Opus:opus",
"//third-party/opus:opus",
":usrsctp",
":libsrtp",
":libevent",

4
third-party/yasm/BUILD vendored
View File

@ -23,8 +23,8 @@ set -x
pushd "$$BUILD_DIR/yasm-1.3.0"
mkdir build
cd build
PATH="$$PATH:$$CMAKE_DIR/cmake-3.18.4-Darwin-x86_64/CMake.app/Contents/bin" arch -x86_64 cmake .. -DYASM_BUILD_TESTS=OFF -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=OFF
arch -x86_64 make -j $$core_count
PATH="$$PATH:$$CMAKE_DIR/cmake-3.19.2-macos-universal/CMake.app/Contents/bin" cmake .. -DYASM_BUILD_TESTS=OFF -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=OFF
make -j $$core_count
popd
tar -cf "$(location yasm.tar)" -C "$$BUILD_DIR/yasm-1.3.0/build" .