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Diffstat (limited to 'src/main/jni/opus/include/opus.h')
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diff --git a/src/main/jni/opus/include/opus.h b/src/main/jni/opus/include/opus.h deleted file mode 100644 index 93a53a2ff..000000000 --- a/src/main/jni/opus/include/opus.h +++ /dev/null @@ -1,978 +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 1 byte, 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(4); - -/** 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 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 */ |