| /* 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 */ |
