diff --git a/engine/audio/private/voice.cpp b/engine/audio/private/voice.cpp index ce506d1f..b50d1362 100644 --- a/engine/audio/private/voice.cpp +++ b/engine/audio/private/voice.cpp @@ -37,7 +37,7 @@ void VoiceTweak_EndVoiceTweakMode(); void EngineTool_OverrideSampleRate( int& rate ); // A fallback codec that should be the most likely to work for local/offline use -#define VOICE_FALLBACK_CODEC "vaudio_celt" +#define VOICE_FALLBACK_CODEC "vaudio_opus" // Special entity index used for tweak mode. #define TWEAKMODE_ENTITYINDEX -500 @@ -583,12 +583,13 @@ bool Voice_Init( const char *pCodecName, int nSampleRate ) bool bSpeex = Q_stricmp( pCodecName, "vaudio_speex" ) == 0; bool bCelt = Q_stricmp( pCodecName, "vaudio_celt" ) == 0; + bool bOpus = Q_stricmp( pCodecName, "vaudio_opus" ) == 0; bool bSteam = Q_stricmp( pCodecName, "steam" ) == 0; // Miles has not been in use for voice in a long long time. Not worth the surface to support ancient demos that may // use it (and probably do not work for other reasons) // "vaudio_miles" - if ( !( bSpeex || bCelt || bSteam ) ) + if ( !( bSpeex || bCelt || bOpus || bSteam ) ) { Msg( "Voice_Init Failed: invalid voice codec %s.\n", pCodecName ); return false; @@ -671,6 +672,12 @@ bool Voice_Init( const char *pCodecName, int nSampleRate ) CreateInterfaceFn createCodecFn = NULL; g_hVoiceCodecDLL = FileSystem_LoadModule(pCodecName); + if( !g_hVoiceCodecDLL || (createCodecFn = Sys_GetFactory(g_hVoiceCodecDLL)) == NULL ) + { + g_hVoiceCodecDLL = FileSystem_LoadModule( VOICE_FALLBACK_CODEC ); + pCodecName = VOICE_FALLBACK_CODEC; + } + if ( !g_hVoiceCodecDLL || (createCodecFn = Sys_GetFactory(g_hVoiceCodecDLL)) == NULL || (g_pEncodeCodec = (IVoiceCodec*)createCodecFn(pCodecName, NULL)) == NULL || !g_pEncodeCodec->Init( quality ) ) { @@ -1075,7 +1082,7 @@ int Voice_GetCompressedData(char *pchDest, int nCount, bool bFinal) { // Check g_bVoiceRecordStopping in case g_bUsingSteamVoice changes on us // while waiting for the end of voice data. - if ( g_bUsingSteamVoice || g_bVoiceRecordStopping ) + if ( g_bUsingSteamVoice && g_bVoiceRecordStopping ) { uint32 cbCompressedWritten = 0; uint32 cbUncompressedWritten = 0; diff --git a/engine/voice_codecs/celt/opus/opus.h b/engine/voice_codecs/celt/opus/opus.h new file mode 100644 index 00000000..d282f21d --- /dev/null +++ b/engine/voice_codecs/celt/opus/opus.h @@ -0,0 +1,981 @@ +/* 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 + * + * + * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet. + * The return value can be negative, 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 OpusEncoder structure. + * @param[in] channels int: 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 opus_int32: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) in input signal + * @param [in] application int: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @param [out] error int*: @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 OpusEncoder*: Encoder state + * @param [in] Fs opus_int32: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) in input signal + * @param [in] application int: 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 OpusEncoder*: Encoder state + * @param [in] pcm opus_int16*: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusEncoder*: Encoder state + * @param [in] pcm float*: 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 int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusEncoder allocated by opus_encoder_create(). + * @param[in] st OpusEncoder*: 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 OpusEncoder*: 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 OpusDecoder structure. + * @param [in] channels int: 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 opus_int32: Sample rate to decode at (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) to decode + * @param [out] error int*: #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 OpusDecoder*: Decoder state. + * @param [in] Fs opus_int32: Sampling rate to decode to (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: 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 OpusDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len opus_int32: Number of bytes in payload* + * @param [out] pcm opus_int16*: 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 must be a multiple of 2.5 ms. + * @param [in] decode_fec int: 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 OpusDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len opus_int32: Number of bytes in payload + * @param [out] pcm float*: 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 must be a multiple of 2.5 ms. + * @param [in] decode_fec int: 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 OpusDecoder*: 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 OpusDecoder allocated by opus_decoder_create(). + * @param[in] st OpusDecoder*: 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 char*: Opus packet to be parsed + * @param [in] len opus_int32: size of data + * @param [out] out_toc char*: TOC pointer + * @param [out] frames char*[48] encapsulated frames + * @param [out] size opus_int16[48] sizes of the encapsulated frames + * @param [out] payload_offset int*: 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 char*: 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 char*: Opus packet. + * This must contain at least one byte of + * data. + * @param [in] Fs opus_int32: 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 char*: 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 char*: Opus packet + * @param [in] len opus_int32: 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 char*: Opus packet + * @param [in] len opus_int32: Length of packet + * @param [in] Fs opus_int32: 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 OpusDecoder*: Decoder state + * @param [in] packet char*: Opus packet + * @param [in] len opus_int32: 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 float*: Input PCM and modified PCM + * @param [in] frame_size int Number of samples per channel to process + * @param [in] channels int: Number of channels + * @param [in,out] softclip_mem float*: 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 TARGET_DURATION_MS 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 OpusRepacketizer 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 OpusRepacketizer*: 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 OpusRepacketizer allocated by + * opus_repacketizer_create(). + * @param[in] rp OpusRepacketizer*: 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 OpusRepacketizer*: The repacketizer state to which to + * add the packet. + * @param[in] data const unsigned char*: 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 opus_int32: 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 OpusRepacketizer*: The repacketizer state from which to + * construct the new packet. + * @param begin int: The index of the first frame in the current + * repacketizer state to include in the output. + * @param end int: One past the index of the last frame in the + * current repacketizer state to include in the + * output. + * @param[out] data const unsigned char*: The buffer in which to + * store the output packet. + * @param maxlen opus_int32: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * 1276 for a single frame, + * or for multiple frames, + * 1277*(end-begin). + * However, 1*(end-begin) 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 [begin,end) 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 OpusRepacketizer*: 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 OpusRepacketizer*: The repacketizer state from which to + * construct the new packet. + * @param[out] data const unsigned char*: The buffer in which to + * store the output packet. + * @param maxlen opus_int32: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * 1277*opus_repacketizer_get_nb_frames(rp). + * However, + * 1*opus_repacketizer_get_nb_frames(rp) + * 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 const unsigned char*: The buffer containing the + * packet to pad. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param new_len opus_int32: 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 const unsigned char*: The buffer containing the + * packet to strip. + * @param len opus_int32: 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 const unsigned char*: The buffer containing the + * packet to pad. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param new_len opus_int32: The desired size of the packet after padding. + * This must be at least 1. + * @param nb_streams opus_int32: 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 const unsigned char*: The buffer containing the + * packet to strip. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param nb_streams opus_int32: 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 */ diff --git a/engine/voice_codecs/celt/opus/opus_custom.h b/engine/voice_codecs/celt/opus/opus_custom.h new file mode 100644 index 00000000..41f36bf2 --- /dev/null +++ b/engine/voice_codecs/celt/opus/opus_custom.h @@ -0,0 +1,342 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2012 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + @file opus_custom.h + @brief Opus-Custom reference implementation API + */ + +#ifndef OPUS_CUSTOM_H +#define OPUS_CUSTOM_H + +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#ifdef CUSTOM_MODES +# define OPUS_CUSTOM_EXPORT OPUS_EXPORT +# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT +#else +# define OPUS_CUSTOM_EXPORT +# ifdef OPUS_BUILD +# define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE +# else +# define OPUS_CUSTOM_EXPORT_STATIC +# endif +#endif + +/** @defgroup opus_custom Opus Custom + * @{ + * Opus Custom is an optional part of the Opus specification and + * reference implementation which uses a distinct API from the regular + * API and supports frame sizes that are not normally supported.\ Use + * of Opus Custom is discouraged for all but very special applications + * for which a frame size different from 2.5, 5, 10, or 20 ms is needed + * (for either complexity or latency reasons) and where interoperability + * is less important. + * + * In addition to the interoperability limitations the use of Opus custom + * disables a substantial chunk of the codec and generally lowers the + * quality available at a given bitrate. Normally when an application needs + * a different frame size from the codec it should buffer to match the + * sizes but this adds a small amount of delay which may be important + * in some very low latency applications. Some transports (especially + * constant rate RF transports) may also work best with frames of + * particular durations. + * + * Libopus only supports custom modes if they are enabled at compile time. + * + * The Opus Custom API is similar to the regular API but the + * @ref opus_encoder_create and @ref opus_decoder_create calls take + * an additional mode parameter which is a structure produced by + * a call to @ref opus_custom_mode_create. Both the encoder and decoder + * must create a mode using the same sample rate (fs) and frame size + * (frame size) so these parameters must either be signaled out of band + * or fixed in a particular implementation. + * + * Similar to regular Opus the custom modes support on the fly frame size + * switching, but the sizes available depend on the particular frame size in + * use. For some initial frame sizes on a single on the fly size is available. + */ + +/** Contains the state of an encoder. One encoder state is needed + for each stream. It is initialized once at the beginning of the + stream. Do *not* re-initialize the state for every frame. + @brief Encoder state + */ +typedef struct OpusCustomEncoder OpusCustomEncoder; + +/** State of the decoder. One decoder state is needed for each stream. + It is initialized once at the beginning of the stream. Do *not* + re-initialize the state for every frame. + @brief Decoder state + */ +typedef struct OpusCustomDecoder OpusCustomDecoder; + +/** The mode contains all the information necessary to create an + encoder. Both the encoder and decoder need to be initialized + with exactly the same mode, otherwise the output will be + corrupted. + @brief Mode configuration + */ +typedef struct OpusCustomMode OpusCustomMode; + +/** Creates a new mode struct. This will be passed to an encoder or + * decoder. The mode MUST NOT BE DESTROYED until the encoders and + * decoders that use it are destroyed as well. + * @param [in] Fs int: Sampling rate (8000 to 96000 Hz) + * @param [in] frame_size int: Number of samples (per channel) to encode in each + * packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes) + * @param [out] error int*: Returned error code (if NULL, no error will be returned) + * @return A newly created mode + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error); + +/** Destroys a mode struct. Only call this after all encoders and + * decoders using this mode are destroyed as well. + * @param [in] mode OpusCustomMode*: Mode to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode); + + +#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C) + +/* Encoder */ +/** Gets the size of an OpusCustomEncoder structure. + * @param [in] mode OpusCustomMode *: Mode configuration + * @param [in] channels int: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +# ifdef CUSTOM_MODES +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_encoder_create(),opus_custom_encoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] mode OpusCustomMode *: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels int: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_init( + OpusCustomEncoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); +# endif +#endif + + +/** Creates a new encoder state. Each stream needs its own encoder + * state (can't be shared across simultaneous streams). + * @param [in] mode OpusCustomMode*: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels int: Number of channels + * @param [out] error int*: Returns an error code + * @return Newly created encoder state. +*/ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + + +/** Destroys a an encoder state. + * @param[in] st OpusCustomEncoder*: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st); + +/** Encodes a frame of audio. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] pcm float*: PCM audio in float format, with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. There must be exactly + * frame_size samples per channel. + * @param [in] frame_size int: Number of samples per frame of input signal + * @param [out] compressed char *: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes int: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float( + OpusCustomEncoder *st, + const float *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes a frame of audio. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] pcm opus_int16*: PCM audio in signed 16-bit format (native endian). + * There must be exactly frame_size samples per channel. + * @param [in] frame_size int: Number of samples per frame of input signal + * @param [out] compressed char *: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes int: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode( + OpusCustomEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_encoderctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + + +#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C) +/* Decoder */ + +/** Gets the size of an OpusCustomDecoder structure. + * @param [in] mode OpusCustomMode *: Mode configuration + * @param [in] channels int: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +/** Initializes a previously allocated decoder state + * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_decoder_create(),opus_custom_decoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] mode OpusCustomMode *: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * encoder) + * @param [in] channels int: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init( + OpusCustomDecoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +#endif + + +/** Creates a new decoder state. Each stream needs its own decoder state (can't + * be shared across simultaneous streams). + * @param [in] mode OpusCustomMode: Contains all the information about the characteristics of the + * stream (must be the same characteristics as used for the encoder) + * @param [in] channels int: Number of channels + * @param [out] error int*: Returns an error code + * @return Newly created decoder state. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + +/** Destroys a an decoder state. + * @param[in] st OpusCustomDecoder*: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st); + +/** Decode an opus custom frame with floating point output + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len int: Number of bytes in payload + * @param [out] pcm float*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + float *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an opus custom frame + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len int: Number of bytes in payload + * @param [out] pcm opus_int16*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + opus_int16 *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_genericctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_CUSTOM_H */ diff --git a/engine/voice_codecs/celt/opus/opus_defines.h b/engine/voice_codecs/celt/opus/opus_defines.h new file mode 100644 index 00000000..d141418b --- /dev/null +++ b/engine/voice_codecs/celt/opus/opus_defines.h @@ -0,0 +1,799 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_defines.h + * @brief Opus reference implementation constants + */ + +#ifndef OPUS_DEFINES_H +#define OPUS_DEFINES_H + +#include "opus_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @defgroup opus_errorcodes Error codes + * @{ + */ +/** No error @hideinitializer*/ +#define OPUS_OK 0 +/** One or more invalid/out of range arguments @hideinitializer*/ +#define OPUS_BAD_ARG -1 +/** Not enough bytes allocated in the buffer @hideinitializer*/ +#define OPUS_BUFFER_TOO_SMALL -2 +/** An internal error was detected @hideinitializer*/ +#define OPUS_INTERNAL_ERROR -3 +/** The compressed data passed is corrupted @hideinitializer*/ +#define OPUS_INVALID_PACKET -4 +/** Invalid/unsupported request number @hideinitializer*/ +#define OPUS_UNIMPLEMENTED -5 +/** An encoder or decoder structure is invalid or already freed @hideinitializer*/ +#define OPUS_INVALID_STATE -6 +/** Memory allocation has failed @hideinitializer*/ +#define OPUS_ALLOC_FAIL -7 +/**@}*/ + +/** @cond OPUS_INTERNAL_DOC */ +/**Export control for opus functions */ + +#ifndef OPUS_EXPORT +# if defined(WIN32) +# if defined(OPUS_BUILD) && defined(DLL_EXPORT) +# define OPUS_EXPORT __declspec(dllexport) +# else +# define OPUS_EXPORT +# endif +# elif defined(__GNUC__) && defined(OPUS_BUILD) +# define OPUS_EXPORT __attribute__ ((visibility ("default"))) +# else +# define OPUS_EXPORT +# endif +#endif + +# if !defined(OPUS_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OPUS_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OPUS_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(3,0) +# define OPUS_RESTRICT __restrict__ +# elif (defined(_MSC_VER) && _MSC_VER >= 1400) +# define OPUS_RESTRICT __restrict +# else +# define OPUS_RESTRICT +# endif +#else +# define OPUS_RESTRICT restrict +#endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(2,7) +# define OPUS_INLINE __inline__ +# elif (defined(_MSC_VER)) +# define OPUS_INLINE __inline +# else +# define OPUS_INLINE +# endif +#else +# define OPUS_INLINE inline +#endif + +/**Warning attributes for opus functions + * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out + * some paranoid null checks. */ +#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__)) +#else +# define OPUS_WARN_UNUSED_RESULT +#endif +#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x))) +#else +# define OPUS_ARG_NONNULL(_x) +#endif + +/** These are the actual Encoder CTL ID numbers. + * They should not be used directly by applications. + * In general, SETs should be even and GETs should be odd.*/ +#define OPUS_SET_APPLICATION_REQUEST 4000 +#define OPUS_GET_APPLICATION_REQUEST 4001 +#define OPUS_SET_BITRATE_REQUEST 4002 +#define OPUS_GET_BITRATE_REQUEST 4003 +#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004 +#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005 +#define OPUS_SET_VBR_REQUEST 4006 +#define OPUS_GET_VBR_REQUEST 4007 +#define OPUS_SET_BANDWIDTH_REQUEST 4008 +#define OPUS_GET_BANDWIDTH_REQUEST 4009 +#define OPUS_SET_COMPLEXITY_REQUEST 4010 +#define OPUS_GET_COMPLEXITY_REQUEST 4011 +#define OPUS_SET_INBAND_FEC_REQUEST 4012 +#define OPUS_GET_INBAND_FEC_REQUEST 4013 +#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014 +#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015 +#define OPUS_SET_DTX_REQUEST 4016 +#define OPUS_GET_DTX_REQUEST 4017 +#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020 +#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021 +#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022 +#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023 +#define OPUS_SET_SIGNAL_REQUEST 4024 +#define OPUS_GET_SIGNAL_REQUEST 4025 +#define OPUS_GET_LOOKAHEAD_REQUEST 4027 +/* #define OPUS_RESET_STATE 4028 */ +#define OPUS_GET_SAMPLE_RATE_REQUEST 4029 +#define OPUS_GET_FINAL_RANGE_REQUEST 4031 +#define OPUS_GET_PITCH_REQUEST 4033 +#define OPUS_SET_GAIN_REQUEST 4034 +#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */ +#define OPUS_SET_LSB_DEPTH_REQUEST 4036 +#define OPUS_GET_LSB_DEPTH_REQUEST 4037 +#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039 +#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040 +#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041 +#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042 +#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043 +/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */ +#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046 +#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047 +#define OPUS_GET_IN_DTX_REQUEST 4049 + +/** Defines for the presence of extended APIs. */ +#define OPUS_HAVE_OPUS_PROJECTION_H + +/* Macros to trigger compilation errors when the wrong types are provided to a CTL */ +#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x)) +#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr))) +#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr))) +#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr))) +/** @endcond */ + +/** @defgroup opus_ctlvalues Pre-defined values for CTL interface + * @see opus_genericctls, opus_encoderctls + * @{ + */ +/* Values for the various encoder CTLs */ +#define OPUS_AUTO -1000 /**opus_int32: 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 opus_int32 *: 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 opus_int32: 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 opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Hard CBR. For LPC/hybrid modes at very low bit-rate, this can + * cause noticeable quality degradation.
+ *
1
VBR (default). The exact type of VBR is controlled by + * #OPUS_SET_VBR_CONSTRAINT.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Hard CBR.
+ *
1
VBR (default). The exact type of VBR may be retrieved via + * #OPUS_GET_VBR_CONSTRAINT.
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
0
Unconstrained VBR.
+ *
1
Constrained VBR (default). This creates a maximum of one + * frame of buffering delay assuming a transport with a + * serialization speed of the nominal bitrate.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Unconstrained VBR.
+ *
1
Constrained VBR (default).
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
#OPUS_AUTO
Not forced (default)
+ *
1
Forced mono
+ *
2
Forced stereo
+ *
+ * @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 opus_int32 *: + *
+ *
#OPUS_AUTO
Not forced (default)
+ *
1
Forced mono
+ *
2
Forced stereo
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
OPUS_BANDWIDTH_FULLBAND
20 kHz passband (default)
+ *
+ * @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 opus_int32 *: Allowed values: + *
+ *
#OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
#OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
#OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
#OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
#OPUS_BANDWIDTH_FULLBAND
20 kHz passband (default)
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
#OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
#OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
#OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
#OPUS_BANDWIDTH_FULLBAND
20 kHz passband
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_SIGNAL_VOICE
Bias thresholds towards choosing LPC or Hybrid modes.
+ *
#OPUS_SIGNAL_MUSIC
Bias thresholds towards choosing MDCT modes.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_SIGNAL_VOICE
Bias thresholds towards choosing LPC or Hybrid modes.
+ *
#OPUS_SIGNAL_MUSIC
Bias thresholds towards choosing MDCT modes.
+ *
+ * @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 opus_int32: Returns one of the following values: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Disable inband FEC (default).
+ *
1
Enable inband FEC.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Inband FEC disabled (default).
+ *
1
Inband FEC enabled.
+ *
+ * @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 opus_int32: 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 opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Disable DTX (default).
+ *
1
Enabled DTX.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
DTX disabled (default).
+ *
1
DTX enabled.
+ *
+ * @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 opus_int32: 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 opus_int32 *: 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 really + * know what you are doing. + * @see OPUS_GET_EXPERT_FRAME_DURATION + * @param[in] x opus_int32: Allowed values: + *
+ *
OPUS_FRAMESIZE_ARG
Select frame size from the argument (default).
+ *
OPUS_FRAMESIZE_2_5_MS
Use 2.5 ms frames.
+ *
OPUS_FRAMESIZE_5_MS
Use 5 ms frames.
+ *
OPUS_FRAMESIZE_10_MS
Use 10 ms frames.
+ *
OPUS_FRAMESIZE_20_MS
Use 20 ms frames.
+ *
OPUS_FRAMESIZE_40_MS
Use 40 ms frames.
+ *
OPUS_FRAMESIZE_60_MS
Use 60 ms frames.
+ *
OPUS_FRAMESIZE_80_MS
Use 80 ms frames.
+ *
OPUS_FRAMESIZE_100_MS
Use 100 ms frames.
+ *
OPUS_FRAMESIZE_120_MS
Use 120 ms frames.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
OPUS_FRAMESIZE_ARG
Select frame size from the argument (default).
+ *
OPUS_FRAMESIZE_2_5_MS
Use 2.5 ms frames.
+ *
OPUS_FRAMESIZE_5_MS
Use 5 ms frames.
+ *
OPUS_FRAMESIZE_10_MS
Use 10 ms frames.
+ *
OPUS_FRAMESIZE_20_MS
Use 20 ms frames.
+ *
OPUS_FRAMESIZE_40_MS
Use 40 ms frames.
+ *
OPUS_FRAMESIZE_60_MS
Use 60 ms frames.
+ *
OPUS_FRAMESIZE_80_MS
Use 80 ms frames.
+ *
OPUS_FRAMESIZE_100_MS
Use 100 ms frames.
+ *
OPUS_FRAMESIZE_120_MS
Use 120 ms frames.
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
0
Enable prediction (default).
+ *
1
Disable prediction.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Prediction enabled (default).
+ *
1
Prediction disabled.
+ *
+ * @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 opus_uint32 *: 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 opus_int32 *: Returns one of the following values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
#OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
#OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
#OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
#OPUS_BANDWIDTH_FULLBAND
20 kHz passband
+ *
+ * @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 Fs value passed to opus_encoder_init() + * or opus_decoder_init(). + * @param[out] x opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Enable phase inversion (default).
+ *
1
Disable phase inversion.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Stereo phase inversion enabled (default).
+ *
1
Stereo phase inversion disabled.
+ *
+ * @hideinitializer */ +#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x) +/** Gets the DTX state of the encoder. + * Returns whether the last encoded frame was either a comfort noise update + * during DTX or not encoded because of DTX. + * @param[out] x opus_int32 *: Returns one of the following values: + *
+ *
0
The encoder is not in DTX.
+ *
1
The encoder is in DTX.
+ *
+ * @hideinitializer */ +#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_decoderctls Decoder related CTLs + * @see opus_genericctls, opus_encoderctls, opus_decoder + * @{ + */ + +/** Configures decoder gain adjustment. + * Scales the decoded output by a factor specified in Q8 dB units. + * This has a maximum range of -32768 to 32767 inclusive, and returns + * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment. + * This setting survives decoder reset. + * + * gain = pow(10, x/(20.0*256)) + * + * @param[in] x opus_int32: 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 opus_int32 *: 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 opus_int32 *: 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 opus_int32 *: 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 int: 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 */ diff --git a/engine/voice_codecs/celt/opus/opus_multistream.h b/engine/voice_codecs/celt/opus/opus_multistream.h new file mode 100644 index 00000000..babcee69 --- /dev/null +++ b/engine/voice_codecs/celt/opus/opus_multistream.h @@ -0,0 +1,660 @@ +/* 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 opus_int32: The index of the stream whose encoder you + * wish to retrieve. + * This must be non-negative and less than + * the streams parameter used + * to initialize the encoder. + * @param[out] y OpusEncoder**: 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 opus_int32: The index of the stream whose decoder you + * wish to retrieve. + * This must be non-negative and less than + * the streams parameter used + * to initialize the decoder. + * @param[out] y OpusDecoder**: 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 + * RFC 7845 + * and is based on the self-delimited Opus framing described in Appendix B of + * RFC 6716. + * Normal Opus packets are just a degenerate case of multistream Opus packets, + * and can be encoded or decoded with the multistream API by setting + * streams to 1 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 mapping table defines which decoded channel i + * should be used for each input/output (I/O) channel j. This table is + * typically provided as an unsigned char array. + * Let i = mapping[j] be the index for I/O channel j. + * If i < 2*coupled_streams, then I/O channel j is + * encoded as the left channel of stream (i/2) if i + * is even, or as the right channel of stream (i/2) if + * i is odd. Otherwise, I/O channel j is encoded as + * mono in stream (i - coupled_streams), 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 i must either + * be the special value 255 or be less than streams + coupled_streams. + * + * The output channels specified by the encoder + * should use the + * Vorbis + * channel ordering. 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 int: The total number of streams to encode from the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) 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 opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param[in] mapping const unsigned char[channels]: 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 int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @param[out] error int *: 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 OpusMSEncoder*: Multistream encoder state to initialize. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param[in] mapping const unsigned char[channels]: 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 int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 OpusMSEncoder*: Multistream encoder state. + * @param[in] pcm const opus_int16*: The input signal as interleaved + * samples. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusMSEncoder*: Multistream encoder state. + * @param[in] pcm const float*: 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 + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusMSEncoder allocated by + * opus_multistream_encoder_create(). + * @param st OpusMSEncoder*: 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 OpusMSEncoder*: 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 OpusMSDecoder structure. + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) 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 opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] mapping const unsigned char[channels]: Mapping from + * coded channels to output channels, as described in + * @ref opus_multistream. + * @param[out] error int *: 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 OpusMSEncoder*: Multistream encoder state to initialize. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] mapping const unsigned char[channels]: 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 OpusMSDecoder*: Multistream decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusMSDecoder*: Multistream decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusMSDecoder*: 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 OpusMSDecoder allocated by + * opus_multistream_decoder_create(). + * @param st OpusMSDecoder: Multistream decoder state to be freed. + */ +OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st); + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_MULTISTREAM_H */ diff --git a/engine/voice_codecs/celt/opus/opus_projection.h b/engine/voice_codecs/celt/opus/opus_projection.h new file mode 100644 index 00000000..9dabf4e8 --- /dev/null +++ b/engine/voice_codecs/celt/opus/opus_projection.h @@ -0,0 +1,568 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_projection.h + * @brief Opus projection reference API + */ + +#ifndef OPUS_PROJECTION_H +#define OPUS_PROJECTION_H + +#include "opus_multistream.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @cond OPUS_INTERNAL_DOC */ + +/** These are the actual encoder and decoder CTL ID numbers. + * They should not be used directly by applications.c + * In general, SETs should be even and GETs should be odd.*/ +/**@{*/ +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST 6001 +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST 6003 +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST 6005 +/**@}*/ + + +/** @endcond */ + +/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs + * + * These are convenience macros that are specific to the + * opus_projection_encoder_ctl() and opus_projection_decoder_ctl() + * interface. + * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, + * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a + * projection encoder or decoder as well. + */ +/**@{*/ + +/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder. + * @param[out] x opus_int32 *: 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 opus_int32 *: 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 unsigned char *: Returns the demixing matrix to the + * supplied pointer location. + * @param y opus_int32: 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 int: The total number of input channels to encode. + * This must be no more than 255. + * @param mapping_family int: 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 opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param mapping_family int: The mapping family to use for selecting + * the appropriate projection. + * @param[out] streams int *: The total number of streams that will + * be encoded from the input. + * @param[out] coupled_streams int *: Number of coupled (2 channel) + * streams that will be encoded from the input. + * @param application int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @param[out] error int *: 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 OpusProjectionEncoder*: Projection encoder state to initialize. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param application int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 OpusProjectionEncoder*: Projection encoder state. + * @param[in] pcm const opus_int16*: The input signal as interleaved + * samples. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusProjectionEncoder*: Projection encoder state. + * @param[in] pcm const float*: 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 + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusProjectionEncoder allocated by + * opus_projection_ambisonics_encoder_create(). + * @param st OpusProjectionEncoder*: 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 OpusProjectionEncoder*: 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 OpusProjectionDecoder structure. + * @param channels int: The total number of output channels. + * This must be no more than 255. + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) 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 opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] demixing_matrix const unsigned char[demixing_matrix_size]: 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 opus_int32: The size in bytes of the + * demixing matrix, as + * described in @ref + * opus_projection_ctls. + * @param[out] error int *: 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 OpusProjectionDecoder*: Projection encoder state to initialize. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] demixing_matrix const unsigned char[demixing_matrix_size]: 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 opus_int32: 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 OpusProjectionDecoder*: Projection decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusProjectionDecoder*: Projection decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusProjectionDecoder*: 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 OpusProjectionDecoder allocated by + * opus_projection_decoder_create(). + * @param st OpusProjectionDecoder: Projection decoder state to be freed. + */ +OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st); + + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_PROJECTION_H */ diff --git a/engine/voice_codecs/celt/opus/opus_types.h b/engine/voice_codecs/celt/opus/opus_types.h new file mode 100644 index 00000000..7cf67558 --- /dev/null +++ b/engine/voice_codecs/celt/opus/opus_types.h @@ -0,0 +1,166 @@ +/* (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 +# 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 + 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 + 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 + 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 */ diff --git a/engine/voice_codecs/celt/opus/opusfile.h b/engine/voice_codecs/celt/opus/opusfile.h new file mode 100644 index 00000000..7c1c89e0 --- /dev/null +++ b/engine/voice_codecs/celt/opus/opusfile.h @@ -0,0 +1,2164 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2012 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.h 17182 2010-04-29 03:48:32Z xiphmont $ + + ********************************************************************/ +#if !defined(_opusfile_h) +# define _opusfile_h (1) + +/**\mainpage + \section Introduction + + This is the documentation for the libopusfile C API. + + The libopusfile package provides a convenient high-level API for + decoding and basic manipulation of all Ogg Opus audio streams. + libopusfile is implemented as a layer on top of Xiph.Org's + reference + libogg + and + libopus + libraries. + + libopusfile provides several sets of built-in routines for + file/stream access, and may also use custom stream I/O routines provided by + the embedded environment. + There are built-in I/O routines provided for ANSI-compliant + stdio (FILE *), memory buffers, and URLs + (including URLs, plus optionally and URLs). + + \section Organization + + The main API is divided into several sections: + - \ref stream_open_close + - \ref stream_info + - \ref stream_decoding + - \ref stream_seeking + + Several additional sections are not tied to the main API. + - \ref stream_callbacks + - \ref header_info + - \ref error_codes + + \section Overview + + The libopusfile API always decodes files to 48 kHz. + The original sample rate is not preserved by the lossy compression, though + it is stored in the header to allow you to resample to it after decoding + (the libopusfile API does not currently provide a resampler, + but the + the + Speex resampler is a good choice if you need one). + In general, if you are playing back the audio, you should leave it at + 48 kHz, provided your audio hardware supports it. + When decoding to a file, it may be worth resampling back to the original + sample rate, so as not to surprise users who might not expect the sample + rate to change after encoding to Opus and decoding. + + Opus files can contain anywhere from 1 to 255 channels of audio. + The channel mappings for up to 8 channels are the same as the + Vorbis + mappings. + A special stereo API can convert everything to 2 channels, making it simple + to support multichannel files in an application which only has stereo + output. + Although the libopusfile ABI provides support for the theoretical + maximum number of channels, the current implementation does not support + files with more than 8 channels, as they do not have well-defined channel + mappings. + + Like all Ogg files, Opus files may be "chained". + That is, multiple Opus files may be combined into a single, longer file just + by concatenating the original files. + This is commonly done in internet radio streaming, as it allows the title + and artist to be updated each time the song changes, since each link in the + chain includes its own set of metadata. + + libopusfile fully supports chained files. + It will decode the first Opus stream found in each link of a chained file + (ignoring any other streams that might be concurrently multiplexed with it, + such as a video stream). + + The channel count can also change between links. + If your application is not prepared to deal with this, it can use the stereo + API to ensure the audio from all links will always get decoded into a + common format. + Since libopusfile always decodes to 48 kHz, you do not have to + worry about the sample rate changing between links (as was possible with + Vorbis). + This makes application support for chained files with libopusfile + very easy.*/ + +# if defined(__cplusplus) +extern "C" { +# endif + +# include +# include +# include +# include + +/**@cond PRIVATE*/ + +/*Enable special features for gcc and gcc-compatible compilers.*/ +# if !defined(OP_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OP_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OP_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility push(default) +# endif + +typedef struct OpusHead OpusHead; +typedef struct OpusTags OpusTags; +typedef struct OpusPictureTag OpusPictureTag; +typedef struct OpusServerInfo OpusServerInfo; +typedef struct OpusFileCallbacks OpusFileCallbacks; +typedef struct OggOpusFile OggOpusFile; + +/*Warning attributes for libopusfile functions.*/ +# if OP_GNUC_PREREQ(3,4) +# define OP_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__)) +# else +# define OP_WARN_UNUSED_RESULT +# endif +# if OP_GNUC_PREREQ(3,4) +# define OP_ARG_NONNULL(_x) __attribute__((__nonnull__(_x))) +# else +# define OP_ARG_NONNULL(_x) +# endif + +/**@endcond*/ + +/**\defgroup error_codes Error Codes*/ +/*@{*/ +/**\name List of possible error codes + Many of the functions in this library return a negative error code when a + function fails. + This list provides a brief explanation of the common errors. + See each individual function for more details on what a specific error code + means in that context.*/ +/*@{*/ + +/**A request did not succeed.*/ +#define OP_FALSE (-1) +/*Currently not used externally.*/ +#define OP_EOF (-2) +/**There was a hole in the page sequence numbers (e.g., a page was corrupt or + missing).*/ +#define OP_HOLE (-3) +/**An underlying read, seek, or tell operation failed when it should have + succeeded.*/ +#define OP_EREAD (-128) +/**A NULL pointer was passed where one was unexpected, or an + internal memory allocation failed, or an internal library error was + encountered.*/ +#define OP_EFAULT (-129) +/**The stream used a feature that is not implemented, such as an unsupported + channel family.*/ +#define OP_EIMPL (-130) +/**One or more parameters to a function were invalid.*/ +#define OP_EINVAL (-131) +/**A purported Ogg Opus stream did not begin with an Ogg page, a purported + header packet did not start with one of the required strings, "OpusHead" or + "OpusTags", or a link in a chained file was encountered that did not + contain any logical Opus streams.*/ +#define OP_ENOTFORMAT (-132) +/**A required header packet was not properly formatted, contained illegal + values, or was missing altogether.*/ +#define OP_EBADHEADER (-133) +/**The ID header contained an unrecognized version number.*/ +#define OP_EVERSION (-134) +/*Currently not used at all.*/ +#define OP_ENOTAUDIO (-135) +/**An audio packet failed to decode properly. + This is usually caused by a multistream Ogg packet where the durations of + the individual Opus packets contained in it are not all the same.*/ +#define OP_EBADPACKET (-136) +/**We failed to find data we had seen before, or the bitstream structure was + sufficiently malformed that seeking to the target destination was + impossible.*/ +#define OP_EBADLINK (-137) +/**An operation that requires seeking was requested on an unseekable stream.*/ +#define OP_ENOSEEK (-138) +/**The first or last granule position of a link failed basic validity checks.*/ +#define OP_EBADTIMESTAMP (-139) + +/*@}*/ +/*@}*/ + +/**\defgroup header_info Header Information*/ +/*@{*/ + +/**The maximum number of channels in an Ogg Opus stream.*/ +#define OPUS_CHANNEL_COUNT_MAX (255) + +/**Ogg Opus bitstream information. + This contains the basic playback parameters for a stream, and corresponds to + the initial ID header packet of an Ogg Opus stream.*/ +struct OpusHead{ + /**The Ogg Opus format version, in the range 0...255. + The top 4 bits represent a "major" version, and the bottom four bits + represent backwards-compatible "minor" revisions. + The current specification describes version 1. + This library will recognize versions up through 15 as backwards compatible + with the current specification. + An earlier draft of the specification described a version 0, but the only + difference between version 1 and version 0 is that version 0 did + not specify the semantics for handling the version field.*/ + int version; + /**The number of channels, in the range 1...255.*/ + int channel_count; + /**The number of samples that should be discarded from the beginning of the + stream.*/ + unsigned pre_skip; + /**The sampling rate of the original input. + All Opus audio is coded at 48 kHz, and should also be decoded at 48 kHz + for playback (unless the target hardware does not support this sampling + rate). + However, this field may be used to resample the audio back to the original + sampling rate, for example, when saving the output to a file.*/ + opus_uint32 input_sample_rate; + /**The gain to apply to the decoded output, in dB, as a Q8 value in the range + -32768...32767. + The libopusfile API will automatically apply this gain to the + decoded output before returning it, scaling it by + pow(10,output_gain/(20.0*256)). + You can adjust this behavior with op_set_gain_offset().*/ + int output_gain; + /**The channel mapping family, in the range 0...255. + Channel mapping family 0 covers mono or stereo in a single stream. + Channel mapping family 1 covers 1 to 8 channels in one or more streams, + using the Vorbis speaker assignments. + Channel mapping family 255 covers 1 to 255 channels in one or more + streams, but without any defined speaker assignment.*/ + int mapping_family; + /**The number of Opus streams in each Ogg packet, in the range 1...255.*/ + int stream_count; + /**The number of coupled Opus streams in each Ogg packet, in the range + 0...127. + This must satisfy 0 <= coupled_count <= stream_count and + coupled_count + stream_count <= 255. + The coupled streams appear first, before all uncoupled streams, in an Ogg + Opus packet.*/ + int coupled_count; + /**The mapping from coded stream channels to output channels. + Let index=mapping[k] be the value for channel k. + If index<2*coupled_count, then it refers to the left channel + from stream (index/2) if even, and the right channel from + stream (index/2) if odd. + Otherwise, it refers to the output of the uncoupled stream + (index-coupled_count).*/ + unsigned char mapping[OPUS_CHANNEL_COUNT_MAX]; +}; + +/**The metadata from an Ogg Opus stream. + + This structure holds the in-stream metadata corresponding to the 'comment' + header packet of an Ogg Opus stream. + The comment header is meant to be used much like someone jotting a quick + note on the label of a CD. + It should be a short, to the point text note that can be more than a couple + words, but not more than a short paragraph. + + The metadata is stored as a series of (tag, value) pairs, in length-encoded + string vectors, using the same format as Vorbis (without the final "framing + bit"), Theora, and Speex, except for the packet header. + The first occurrence of the '=' character delimits the tag and value. + A particular tag may occur more than once, and order is significant. + The character set encoding for the strings is always UTF-8, but the tag + names are limited to ASCII, and treated as case-insensitive. + See the Vorbis + comment header specification for details. + + In filling in this structure, libopusfile will null-terminate the + #user_comments strings for safety. + However, the bitstream format itself treats them as 8-bit clean vectors, + possibly containing NUL characters, so the #comment_lengths array should be + treated as their authoritative length. + + This structure is binary and source-compatible with a + vorbis_comment, and pointers to it may be freely cast to + vorbis_comment pointers, and vice versa. + It is provided as a separate type to avoid introducing a compile-time + dependency on the libvorbis headers.*/ +struct OpusTags{ + /**The array of comment string vectors.*/ + char **user_comments; + /**An array of the corresponding length of each vector, in bytes.*/ + int *comment_lengths; + /**The total number of comment streams.*/ + int comments; + /**The null-terminated vendor string. + This identifies the software used to encode the stream.*/ + char *vendor; +}; + +/**\name Picture tag image formats*/ +/*@{*/ + +/**The MIME type was not recognized, or the image data did not match the + declared MIME type.*/ +#define OP_PIC_FORMAT_UNKNOWN (-1) +/**The MIME type indicates the image data is really a URL.*/ +#define OP_PIC_FORMAT_URL (0) +/**The image is a JPEG.*/ +#define OP_PIC_FORMAT_JPEG (1) +/**The image is a PNG.*/ +#define OP_PIC_FORMAT_PNG (2) +/**The image is a GIF.*/ +#define OP_PIC_FORMAT_GIF (3) + +/*@}*/ + +/**The contents of a METADATA_BLOCK_PICTURE tag.*/ +struct OpusPictureTag{ + /**The picture type according to the ID3v2 APIC frame: +
    +
  1. Other
    2. +
  2. 32x32 pixels 'file icon' (PNG only)
    3. +
  3. Other file icon
    4. +
  4. Cover (front)
    5. +
  5. Cover (back)
    6. +
  6. Leaflet page
    7. +
  7. Media (e.g. label side of CD)
    8. +
  8. Lead artist/lead performer/soloist
    9. +
  9. Artist/performer
    10. +
  10. Conductor
    11. +
  11. Band/Orchestra
    12. +
  12. Composer
    13. +
  13. Lyricist/text writer
    14. +
  14. Recording Location
    15. +
  15. During recording
    16. +
  16. During performance
    17. +
  17. Movie/video screen capture
    18. +
  18. A bright colored fish
    19. +
  19. Illustration
    20. +
  20. Band/artist logotype
    21. +
  21. Publisher/Studio logotype
    22. +
+ Others are reserved and should not be used. + There may only be one each of picture type 1 and 2 in a file.*/ + opus_int32 type; + /**The MIME type of the picture, in printable ASCII characters 0x20-0x7E. + The MIME type may also be "-->" to signify that the data part + is a URL pointing to the picture instead of the picture data itself. + In this case, a terminating NUL is appended to the URL string in #data, + but #data_length is set to the length of the string excluding that + terminating NUL.*/ + char *mime_type; + /**The description of the picture, in UTF-8.*/ + char *description; + /**The width of the picture in pixels.*/ + opus_uint32 width; + /**The height of the picture in pixels.*/ + opus_uint32 height; + /**The color depth of the picture in bits-per-pixel (not + bits-per-channel).*/ + opus_uint32 depth; + /**For indexed-color pictures (e.g., GIF), the number of colors used, or 0 + for non-indexed pictures.*/ + opus_uint32 colors; + /**The length of the picture data in bytes.*/ + opus_uint32 data_length; + /**The binary picture data.*/ + unsigned char *data; + /**The format of the picture data, if known. + One of +
    +
  • #OP_PIC_FORMAT_UNKNOWN,
    • +
  • #OP_PIC_FORMAT_URL,
    • +
  • #OP_PIC_FORMAT_JPEG,
    • +
  • #OP_PIC_FORMAT_PNG, or
    • +
  • #OP_PIC_FORMAT_GIF.
    • +
*/ + int format; +}; + +/**\name Functions for manipulating header data + + These functions manipulate the #OpusHead and #OpusTags structures, + which describe the audio parameters and tag-value metadata, respectively. + These can be used to query the headers returned by libopusfile, or + to parse Opus headers from sources other than an Ogg Opus stream, provided + they use the same format.*/ +/*@{*/ + +/**Parses the contents of the ID header packet of an Ogg Opus stream. + \param[out] _head Returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be NULL to merely test the header + for validity. + \param[in] _data The contents of the ID header packet. + \param _len The number of bytes of data in the ID header packet. + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusHead" + string. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EIMPL If the channel mapping family was 255, which general + purpose players should not attempt to play. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification: +
    +
  • Insufficient data,
    • +
  • Too much data for the known minor versions,
    • +
  • An unrecognized channel mapping family,
    • +
  • Zero channels or too many channels,
    • +
  • Zero coded streams,
    • +
  • Too many coupled streams, or
    • +
  • An invalid channel mapping index.
    • +
*/ +OP_WARN_UNUSED_RESULT int opus_head_parse(OpusHead *_head, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Converts a granule position to a sample offset for a given Ogg Opus stream. + The sample offset is simply _gp-_head->pre_skip. + Granule position values smaller than OpusHead#pre_skip correspond to audio + that should never be played, and thus have no associated sample offset. + This function returns -1 for such values. + This function also correctly handles extremely large granule positions, + which may have wrapped around to a negative number when stored in a signed + ogg_int64_t value. + \param _head The #OpusHead information from the ID header of the stream. + \param _gp The granule position to convert. + \return The sample offset associated with the given granule position + (counting at a 48 kHz sampling rate), or the special value -1 on + error (i.e., the granule position was smaller than the pre-skip + amount).*/ +ogg_int64_t opus_granule_sample(const OpusHead *_head,ogg_int64_t _gp) + OP_ARG_NONNULL(1); + +/**Parses the contents of the 'comment' header packet of an Ogg Opus stream. + \param[out] _tags An uninitialized #OpusTags structure. + This returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be NULL to merely test the header + for validity. + \param[in] _data The contents of the 'comment' header packet. + \param _len The number of bytes of data in the 'info' header packet. + \retval 0 Success. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusTags" + string. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification. + \retval #OP_EFAULT If there wasn't enough memory to store the tags.*/ +OP_WARN_UNUSED_RESULT int opus_tags_parse(OpusTags *_tags, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Performs a deep copy of an #OpusTags structure. + \param _dst The #OpusTags structure to copy into. + If this function fails, the contents of this structure remain + untouched. + \param _src The #OpusTags structure to copy from. + \retval 0 Success. + \retval #OP_EFAULT If there wasn't enough memory to copy the tags.*/ +int opus_tags_copy(OpusTags *_dst,const OpusTags *_src) OP_ARG_NONNULL(1); + +/**Initializes an #OpusTags structure. + This should be called on a freshly allocated #OpusTags structure before + attempting to use it. + \param _tags The #OpusTags structure to initialize.*/ +void opus_tags_init(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Add a (tag, value) pair to an initialized #OpusTags structure. + \note Neither opus_tags_add() nor opus_tags_add_comment() support values + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the (tag, value) pair to. + \param _tag A NUL-terminated, case-insensitive, ASCII string containing + the tag to add (without an '=' character). + \param _value A NUL-terminated UTF-8 containing the corresponding value. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add(OpusTags *_tags,const char *_tag,const char *_value) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) OP_ARG_NONNULL(3); + +/**Add a comment to an initialized #OpusTags structure. + \note Neither opus_tags_add_comment() nor opus_tags_add() support comments + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the comment to. + \param _comment A NUL-terminated UTF-8 string containing the comment in + "TAG=value" form. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add_comment(OpusTags *_tags,const char *_comment) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Replace the binary suffix data at the end of the packet (if any). + \param _tags An initialized #OpusTags structure. + \param _data A buffer of binary data to append after the encoded user + comments. + The least significant bit of the first byte of this data must + be set (to ensure the data is preserved by other editors). + \param _len The number of bytes of binary data to append. + This may be zero to remove any existing binary suffix data. + \return 0 on success, or a negative value on error. + \retval #OP_EINVAL \a _len was negative, or \a _len was positive but + \a _data was NULL or the least significant + bit of the first byte was not set. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_set_binary_suffix(OpusTags *_tags, + const unsigned char *_data,int _len) OP_ARG_NONNULL(1); + +/**Look up a comment value by its tag. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \param _count The instance of the tag. + The same tag can appear multiple times, each with a distinct + value, so an index is required to retrieve them all. + The order in which these values appear is significant and + should be preserved. + Use opus_tags_query_count() to get the legal range for the + \a _count parameter. + \return A pointer to the queried tag's value. + This points directly to data in the #OpusTags structure. + It should not be modified or freed by the application, and + modifications to the structure may invalidate the pointer. + \retval NULL If no matching tag is found.*/ +const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Look up the number of instances of a tag. + Call this first when querying for a specific tag and then iterate over the + number of instances with separate calls to opus_tags_query() to retrieve + all the values for that tag in order. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \return The number of instances of this particular tag.*/ +int opus_tags_query_count(const OpusTags *_tags,const char *_tag) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Retrieve the binary suffix data at the end of the packet (if any). + \param _tags An initialized #OpusTags structure. + \param[out] _len Returns the number of bytes of binary suffix data returned. + \return A pointer to the binary suffix data, or NULL if none + was present.*/ +const unsigned char *opus_tags_get_binary_suffix(const OpusTags *_tags, + int *_len) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Get the album gain from an R128_ALBUM_GAIN tag, if one was specified. + This searches for the first R128_ALBUM_GAIN tag with a valid signed, + 16-bit decimal integer value and returns the value. + This routine is exposed merely for convenience for applications which wish + to do something special with the album gain (i.e., display it). + If you simply wish to apply the album gain instead of the header gain, you + can use op_set_gain_offset() with an #OP_ALBUM_GAIN type and no offset. + \param _tags An initialized #OpusTags structure. + \param[out] _gain_q8 The album gain, in 1/256ths of a dB. + This will lie in the range [-32768,32767], and should + be applied in addition to the header gain. + On error, no value is returned, and the previous + contents remain unchanged. + \return 0 on success, or a negative value on error. + \retval #OP_FALSE There was no album gain available in the given tags.*/ +int opus_tags_get_album_gain(const OpusTags *_tags,int *_gain_q8) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Get the track gain from an R128_TRACK_GAIN tag, if one was specified. + This searches for the first R128_TRACK_GAIN tag with a valid signed, + 16-bit decimal integer value and returns the value. + This routine is exposed merely for convenience for applications which wish + to do something special with the track gain (i.e., display it). + If you simply wish to apply the track gain instead of the header gain, you + can use op_set_gain_offset() with an #OP_TRACK_GAIN type and no offset. + \param _tags An initialized #OpusTags structure. + \param[out] _gain_q8 The track gain, in 1/256ths of a dB. + This will lie in the range [-32768,32767], and should + be applied in addition to the header gain. + On error, no value is returned, and the previous + contents remain unchanged. + \return 0 on success, or a negative value on error. + \retval #OP_FALSE There was no track gain available in the given tags.*/ +int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Clears the #OpusTags structure. + This should be called on an #OpusTags structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _tags The #OpusTags structure to clear.*/ +void opus_tags_clear(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Check if \a _comment is an instance of a \a _tag_name tag. + \see opus_tagncompare + \param _tag_name A NUL-terminated, case-insensitive, ASCII string containing + the name of the tag to check for (without the terminating + '=' character). + \param _comment The comment string to check. + \return An integer less than, equal to, or greater than zero if \a _comment + is found respectively, to be less than, to match, or be greater + than a "tag=value" string whose tag matches \a _tag_name.*/ +int opus_tagcompare(const char *_tag_name,const char *_comment); + +/**Check if \a _comment is an instance of a \a _tag_name tag. + This version is slightly more efficient than opus_tagcompare() if the length + of the tag name is already known (e.g., because it is a constant). + \see opus_tagcompare + \param _tag_name A case-insensitive ASCII string containing the name of the + tag to check for (without the terminating '=' character). + \param _tag_len The number of characters in the tag name. + This must be non-negative. + \param _comment The comment string to check. + \return An integer less than, equal to, or greater than zero if \a _comment + is found respectively, to be less than, to match, or be greater + than a "tag=value" string whose tag matches the first \a _tag_len + characters of \a _tag_name.*/ +int opus_tagncompare(const char *_tag_name,int _tag_len,const char *_comment); + +/**Parse a single METADATA_BLOCK_PICTURE tag. + This decodes the BASE64-encoded content of the tag and returns a structure + with the MIME type, description, image parameters (if known), and the + compressed image data. + If the MIME type indicates the presence of an image format we recognize + (JPEG, PNG, or GIF) and the actual image data contains the magic signature + associated with that format, then the OpusPictureTag::format field will be + set to the corresponding format. + This is provided as a convenience to avoid requiring applications to parse + the MIME type and/or do their own format detection for the commonly used + formats. + In this case, we also attempt to extract the image parameters directly from + the image data (overriding any that were present in the tag, which the + specification says applications are not meant to rely on). + The application must still provide its own support for actually decoding the + image data and, if applicable, retrieving that data from URLs. + \param[out] _pic Returns the parsed picture data. + No sanitation is done on the type, MIME type, or + description fields, so these might return invalid values. + The contents of this structure are left unmodified on + failure. + \param _tag The METADATA_BLOCK_PICTURE tag contents. + The leading "METADATA_BLOCK_PICTURE=" portion is optional, + to allow the function to be used on either directly on the + values in OpusTags::user_comments or on the return value + of opus_tags_query(). + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT The METADATA_BLOCK_PICTURE contents were not valid. + \retval #OP_EFAULT There was not enough memory to store the picture tag + contents.*/ +OP_WARN_UNUSED_RESULT int opus_picture_tag_parse(OpusPictureTag *_pic, + const char *_tag) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Initializes an #OpusPictureTag structure. + This should be called on a freshly allocated #OpusPictureTag structure + before attempting to use it. + \param _pic The #OpusPictureTag structure to initialize.*/ +void opus_picture_tag_init(OpusPictureTag *_pic) OP_ARG_NONNULL(1); + +/**Clears the #OpusPictureTag structure. + This should be called on an #OpusPictureTag structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _pic The #OpusPictureTag structure to clear.*/ +void opus_picture_tag_clear(OpusPictureTag *_pic) OP_ARG_NONNULL(1); + +/*@}*/ + +/*@}*/ + +/**\defgroup url_options URL Reading Options*/ +/*@{*/ +/**\name URL reading options + Options for op_url_stream_create() and associated functions. + These allow you to provide proxy configuration parameters, skip SSL + certificate checks, etc. + Options are processed in order, and if the same option is passed multiple + times, only the value specified by the last occurrence has an effect + (unless otherwise specified). + They may be expanded in the future.*/ +/*@{*/ + +/**@cond PRIVATE*/ + +/*These are the raw numbers used to define the request codes. + They should not be used directly.*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST (6464) +#define OP_HTTP_PROXY_HOST_REQUEST (6528) +#define OP_HTTP_PROXY_PORT_REQUEST (6592) +#define OP_HTTP_PROXY_USER_REQUEST (6656) +#define OP_HTTP_PROXY_PASS_REQUEST (6720) +#define OP_GET_SERVER_INFO_REQUEST (6784) + +#define OP_URL_OPT(_request) ((char *)(_request)) + +/*These macros trigger compilation errors or warnings if the wrong types are + provided to one of the URL options.*/ +#define OP_CHECK_INT(_x) ((void)((_x)==(opus_int32)0),(opus_int32)(_x)) +#define OP_CHECK_CONST_CHAR_PTR(_x) ((_x)+((_x)-(const char *)(_x))) +#define OP_CHECK_SERVER_INFO_PTR(_x) ((_x)+((_x)-(OpusServerInfo *)(_x))) + +/**@endcond*/ + +/**HTTP/Shoutcast/Icecast server information associated with a URL.*/ +struct OpusServerInfo{ + /**The name of the server (icy-name/ice-name). + This is NULL if there was no icy-name or + ice-name header.*/ + char *name; + /**A short description of the server (icy-description/ice-description). + This is NULL if there was no icy-description or + ice-description header.*/ + char *description; + /**The genre the server falls under (icy-genre/ice-genre). + This is NULL if there was no icy-genre or + ice-genre header.*/ + char *genre; + /**The homepage for the server (icy-url/ice-url). + This is NULL if there was no icy-url or + ice-url header.*/ + char *url; + /**The software used by the origin server (Server). + This is NULL if there was no Server header.*/ + char *server; + /**The media type of the entity sent to the recepient (Content-Type). + This is NULL if there was no Content-Type + header.*/ + char *content_type; + /**The nominal stream bitrate in kbps (icy-br/ice-bitrate). + This is -1 if there was no icy-br or + ice-bitrate header.*/ + opus_int32 bitrate_kbps; + /**Flag indicating whether the server is public (1) or not + (0) (icy-pub/ice-public). + This is -1 if there was no icy-pub or + ice-public header.*/ + int is_public; + /**Flag indicating whether the server is using HTTPS instead of HTTP. + This is 0 unless HTTPS is being used. + This may not match the protocol used in the original URL if there were + redirections.*/ + int is_ssl; +}; + +/**Initializes an #OpusServerInfo structure. + All fields are set as if the corresponding header was not available. + \param _info The #OpusServerInfo structure to initialize. + \note If you use this function, you must link against libopusurl.*/ +void opus_server_info_init(OpusServerInfo *_info) OP_ARG_NONNULL(1); + +/**Clears the #OpusServerInfo structure. + This should be called on an #OpusServerInfo structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _info The #OpusServerInfo structure to clear. + \note If you use this function, you must link against libopusurl.*/ +void opus_server_info_clear(OpusServerInfo *_info) OP_ARG_NONNULL(1); + +/**Skip the certificate check when connecting via TLS/SSL (https). + \param _b opus_int32: Whether or not to skip the certificate + check. + The check will be skipped if \a _b is non-zero, and will not be + skipped if \a _b is zero. + \hideinitializer*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK(_b) \ + OP_URL_OPT(OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST),OP_CHECK_INT(_b) + +/**Proxy connections through the given host. + If no port is specified via #OP_HTTP_PROXY_PORT, the port number defaults + to 8080 (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _host const char *: The proxy server hostname. + This may be NULL to disable the use of a proxy + server. + \hideinitializer*/ +#define OP_HTTP_PROXY_HOST(_host) \ + OP_URL_OPT(OP_HTTP_PROXY_HOST_REQUEST),OP_CHECK_CONST_CHAR_PTR(_host) + +/**Use the given port when proxying connections. + This option only has an effect if #OP_HTTP_PROXY_HOST is specified with a + non-NULL \a _host. + If this option is not provided, the proxy port number defaults to 8080 + (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _port opus_int32: The proxy server port. + This must be in the range 0...65535 (inclusive), or the + URL function this is passed to will fail. + \hideinitializer*/ +#define OP_HTTP_PROXY_PORT(_port) \ + OP_URL_OPT(OP_HTTP_PROXY_PORT_REQUEST),OP_CHECK_INT(_port) + +/**Use the given user name for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _user const char *: The proxy server user name. + This may be NULL to disable proxy + authentication. + A non-NULL value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_PASS + are also specified with non-NULL + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_USER(_user) \ + OP_URL_OPT(OP_HTTP_PROXY_USER_REQUEST),OP_CHECK_CONST_CHAR_PTR(_user) + +/**Use the given password for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _pass const char *: The proxy server password. + This may be NULL to disable proxy + authentication. + A non-NULL value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_USER + are also specified with non-NULL + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_PASS(_pass) \ + OP_URL_OPT(OP_HTTP_PROXY_PASS_REQUEST),OP_CHECK_CONST_CHAR_PTR(_pass) + +/**Parse information about the streaming server (if any) and return it. + Very little validation is done. + In particular, OpusServerInfo::url may not be a valid URL, + OpusServerInfo::bitrate_kbps may not really be in kbps, and + OpusServerInfo::content_type may not be a valid MIME type. + The character set of the string fields is not specified anywhere, and should + not be assumed to be valid UTF-8. + \param _info OpusServerInfo *: Returns information about the server. + If there is any error opening the stream, the + contents of this structure remain + unmodified. + On success, fills in the structure with the + server information that was available, if + any. + After a successful return, the contents of + this structure should be freed by calling + opus_server_info_clear(). + \hideinitializer*/ +#define OP_GET_SERVER_INFO(_info) \ + OP_URL_OPT(OP_GET_SERVER_INFO_REQUEST),OP_CHECK_SERVER_INFO_PTR(_info) + +/*@}*/ +/*@}*/ + +/**\defgroup stream_callbacks Abstract Stream Reading Interface*/ +/*@{*/ +/**\name Functions for reading from streams + These functions define the interface used to read from and seek in a stream + of data. + A stream does not need to implement seeking, but the decoder will not be + able to seek if it does not do so. + These functions also include some convenience routines for working with + standard FILE pointers, complete streams stored in a single + block of memory, or URLs.*/ +/*@{*/ + +/**Reads up to \a _nbytes bytes of data from \a _stream. + \param _stream The stream to read from. + \param[out] _ptr The buffer to store the data in. + \param _nbytes The maximum number of bytes to read. + This function may return fewer, though it will not + return zero unless it reaches end-of-file. + \return The number of bytes successfully read, or a negative value on + error.*/ +typedef int (*op_read_func)(void *_stream,unsigned char *_ptr,int _nbytes); + +/**Sets the position indicator for \a _stream. + The new position, measured in bytes, is obtained by adding \a _offset + bytes to the position specified by \a _whence. + If \a _whence is set to SEEK_SET, SEEK_CUR, or + SEEK_END, the offset is relative to the start of the stream, + the current position indicator, or end-of-file, respectively. + \retval 0 Success. + \retval -1 Seeking is not supported or an error occurred. + errno need not be set.*/ +typedef int (*op_seek_func)(void *_stream,opus_int64 _offset,int _whence); + +/**Obtains the current value of the position indicator for \a _stream. + \return The current position indicator.*/ +typedef opus_int64 (*op_tell_func)(void *_stream); + +/**Closes the underlying stream. + \retval 0 Success. + \retval EOF An error occurred. + errno need not be set.*/ +typedef int (*op_close_func)(void *_stream); + +/**The callbacks used to access non-FILE stream resources. + The function prototypes are basically the same as for the stdio functions + fread(), fseek(), ftell(), and + fclose(). + The differences are that the FILE * arguments have been + replaced with a void *, which is to be used as a pointer to + whatever internal data these functions might need, that #seek and #tell + take and return 64-bit offsets, and that #seek must return -1 if + the stream is unseekable.*/ +struct OpusFileCallbacks{ + /**Used to read data from the stream. + This must not be NULL.*/ + op_read_func read; + /**Used to seek in the stream. + This may be NULL if seeking is not implemented.*/ + op_seek_func seek; + /**Used to return the current read position in the stream. + This may be NULL if seeking is not implemented.*/ + op_tell_func tell; + /**Used to close the stream when the decoder is freed. + This may be NULL to leave the stream open.*/ + op_close_func close; +}; + +/**Opens a stream with fopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + On Windows, this string must be UTF-8 (to allow access to + files whose names cannot be represented in the current + MBCS code page). + All other systems use the native character encoding. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) + OP_ARG_NONNULL(3); + +/**Opens a stream with fdopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _fd The file descriptor to open. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fdopen(OpusFileCallbacks *_cb, + int _fd,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(3); + +/**Opens a stream with freopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + On Windows, this string must be UTF-8 (to allow access + to files whose names cannot be represented in the + current MBCS code page). + All other systems use the native character encoding. + \param _mode The mode to open the file in. + \param _stream A stream previously returned by op_fopen(), op_fdopen(), + or op_freopen(). + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_freopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode,void *_stream) OP_ARG_NONNULL(1) + OP_ARG_NONNULL(2) OP_ARG_NONNULL(3) OP_ARG_NONNULL(4); + +/**Creates a stream that reads from the given block of memory. + This block of memory must contain the complete stream to decode. + This is useful for caching small streams (e.g., sound effects) in RAM. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _data The block of memory to read from. + \param _size The size of the block of memory. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_mem_stream_create(OpusFileCallbacks *_cb, + const unsigned char *_data,size_t _size) OP_ARG_NONNULL(1); + +/**Creates a stream that reads from the given URL. + This function behaves identically to op_url_stream_create(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will + be filled in here. + \param _url The URL to read from. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[in,out] _ap A list of the \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with NULL. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_vcreate(OpusFileCallbacks *_cb, + const char *_url,va_list _ap) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Creates a stream that reads from the given URL. + \note If you use this function, you must link against libopusurl. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _url The URL to read from. + Currently only the , , and schemes + are supported. + Both and may be disabled at compile time, + in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, before + passing them to this function. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + NULL. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_create(OpusFileCallbacks *_cb, + const char *_url,...) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_open_close Opening and Closing*/ +/*@{*/ +/**\name Functions for opening and closing streams + + These functions allow you to test a stream to see if it is Opus, open it, + and close it. + Several flavors are provided for each of the built-in stream types, plus a + more general version which takes a set of application-provided callbacks.*/ +/*@{*/ + +/**Test to see if this is an Opus stream. + For good results, you will need at least 57 bytes (for a pure Opus-only + stream). + Something like 512 bytes will give more reliable results for multiplexed + streams. + This function is meant to be a quick-rejection filter. + Its purpose is not to guarantee that a stream is a valid Opus stream, but to + ensure that it looks enough like Opus that it isn't going to be recognized + as some other format (except possibly an Opus stream that is also + multiplexed with other codecs, such as video). + \param[out] _head The parsed ID header contents. + You may pass NULL if you do not need + this information. + If the function fails, the contents of this structure + remain untouched. + \param _initial_data An initial buffer of data from the start of the + stream. + \param _initial_bytes The number of bytes in \a _initial_data. + \return 0 if the data appears to be Opus, or a negative value on error. + \retval #OP_FALSE There was not enough data to tell if this was an Opus + stream or not. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_ENOTFORMAT If the data did not contain a recognizable ID + header for an Opus stream. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EBADHEADER The ID header was not properly formatted or contained + illegal values.*/ +int op_test(OpusHead *_head, + const unsigned char *_initial_data,size_t _initial_bytes); + +/**Open a stream from the given file path. + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Open a stream from a memory buffer. + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Open a stream from a URL. + This function behaves identically to op_open_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always + fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, + with internationalized domain names encoded in + punycode, before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with NULL. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vopen_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Open a stream from a URL. + \note If you use this function, you must link against libopusurl. + \param _url The URL to open. + Currently only the , , and schemes + are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + NULL. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Open a stream using the given set of callbacks to access it. + \param _stream The stream to read from (e.g., a FILE *). + This value will be passed verbatim as the first + argument to all of the callbacks. + \param _cb The callbacks with which to access the stream. + read() must + be implemented. + seek() and + tell() may + be NULL, or may always return -1 to + indicate a stream is unseekable, but if + seek() is + implemented and succeeds on a particular stream, then + tell() must + also. + close() may + be NULL, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be opened, even if it is unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + tell() + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + The failure code will be one of +
+
#OP_EREAD
+
An underlying read, seek, or tell operation + failed when it should have succeeded, or we failed + to find data in the stream we had seen before.
+
#OP_EFAULT
+
There was a memory allocation failure, or an + internal library error.
+
#OP_EIMPL
+
The stream used a feature that is not + implemented, such as an unsupported channel + family.
+
#OP_EINVAL
+
seek() + was implemented and succeeded on this source, but + tell() + did not, or the starting position indicator was + not equal to \a _initial_bytes.
+
#OP_ENOTFORMAT
+
The stream contained a link that did not have + any logical Opus streams in it.
+
#OP_EBADHEADER
+
A required header packet was not properly + formatted, contained illegal values, or was missing + altogether.
+
#OP_EVERSION
+
An ID header contained an unrecognized version + number.
+
#OP_EBADLINK
+
We failed to find data we had seen before after + seeking.
+
#OP_EBADTIMESTAMP
+
The first or last timestamp in a link failed + basic validity checks.
+
+ \return A freshly opened \c OggOpusFile, or NULL on error. + libopusfile does not take ownership of the stream + if the call fails. + The calling application is responsible for closing the stream if + this call returns an error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_stream, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Partially open a stream from the given file path. + \see op_test_callbacks + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Partially open a stream from a memory buffer. + \see op_test_callbacks + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Partially open a stream from a URL. + This function behaves identically to op_test_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \see op_test_url + \see op_test_callbacks + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always + fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, + with internationalized domain names encoded in + punycode, before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with NULL. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vtest_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Partially open a stream from a URL. + \note If you use this function, you must link against libopusurl. + \see op_test_callbacks + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure + codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with NULL. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Partially open a stream using the given set of callbacks to access it. + This tests for Opusness and loads the headers for the first link. + It does not seek (although it tests for seekability). + You can query a partially open stream for the few pieces of basic + information returned by op_serialno(), op_channel_count(), op_head(), and + op_tags() (but only for the first link). + You may also determine if it is seekable via a call to op_seekable(). + You cannot read audio from the stream, seek, get the size or duration, + get information from links other than the first one, or even get the total + number of links until you finish opening the stream with op_test_open(). + If you do not need to do any of these things, you can dispose of it with + op_free() instead. + + This function is provided mostly to simplify porting existing code that used + libvorbisfile. + For new code, you are likely better off using op_test() instead, which + is less resource-intensive, requires less data to succeed, and imposes a + hard limit on the amount of data it examines (important for unseekable + streams, where all such data must be buffered until you are sure of the + stream type). + \param _stream The stream to read from (e.g., a FILE *). + This value will be passed verbatim as the first + argument to all of the callbacks. + \param _cb The callbacks with which to access the stream. + read() must + be implemented. + seek() and + tell() may + be NULL, or may always return -1 to + indicate a stream is unseekable, but if + seek() is + implemented and succeeds on a particular stream, then + tell() must + also. + close() may + be NULL, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be tested more thoroughly, even if it is + unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + tell() + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \return A partially opened \c OggOpusFile, or NULL on error. + libopusfile does not take ownership of the stream + if the call fails. + The calling application is responsible for closing the stream if + this call returns an error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_stream, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Finish opening a stream partially opened with op_test_callbacks() or one of + the associated convenience functions. + If this function fails, you are still responsible for freeing the + \c OggOpusFile with op_free(). + \param _of The \c OggOpusFile to finish opening. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read, seek, or tell operation failed + when it should have succeeded. + \retval #OP_EFAULT There was a memory allocation failure, or an + internal library error. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_EINVAL The stream was not partially opened with + op_test_callbacks() or one of the associated + convenience functions. + \retval #OP_ENOTFORMAT The stream contained a link that did not have any + logical Opus streams in it. + \retval #OP_EBADHEADER A required header packet was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An ID header contained an unrecognized version + number. + \retval #OP_EBADLINK We failed to find data we had seen before after + seeking. + \retval #OP_EBADTIMESTAMP The first or last timestamp in a link failed basic + validity checks.*/ +int op_test_open(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Release all memory used by an \c OggOpusFile. + \param _of The \c OggOpusFile to free.*/ +void op_free(OggOpusFile *_of); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_info Stream Information*/ +/*@{*/ +/**\name Functions for obtaining information about streams + + These functions allow you to get basic information about a stream, including + seekability, the number of links (for chained streams), plus the size, + duration, bitrate, header parameters, and meta information for each link + (or, where available, the stream as a whole). + Some of these (size, duration) are only available for seekable streams. + You can also query the current stream position, link, and playback time, + and instantaneous bitrate during playback. + + Some of these functions may be used successfully on the partially open + streams returned by op_test_callbacks() or one of the associated + convenience functions. + Their documention will indicate so explicitly.*/ +/*@{*/ + +/**Returns whether or not the stream being read is seekable. + This is true if +
    +
  1. The seek() and + tell() callbacks are both + non-NULL,
    2. +
  2. The seek() callback was + successfully executed at least once, and
    3. +
  3. The tell() callback was + successfully able to report the position indicator afterwards.
    4. +
+ This function may be called on partially-opened streams. + \param _of The \c OggOpusFile whose seekable status is to be returned. + \return A non-zero value if seekable, and 0 if unseekable.*/ +int op_seekable(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Returns the number of links in this chained stream. + This function may be called on partially-opened streams, but it will always + return 1. + The actual number of links is not known until the stream is fully opened. + \param _of The \c OggOpusFile from which to retrieve the link count. + \return For fully-open seekable streams, this returns the total number of + links in the whole stream, which will be at least 1. + For partially-open or unseekable streams, this always returns 1.*/ +int op_link_count(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Get the serial number of the given link in a (possibly-chained) Ogg Opus + stream. + This function may be called on partially-opened streams, but it will always + return the serial number of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the serial number. + \param _li The index of the link whose serial number should be retrieved. + Use a negative number to get the serial number of the current + link. + \return The serial number of the given link. + If \a _li is greater than the total number of links, this returns + the serial number of the last link. + If the stream is not seekable, this always returns the serial number + of the current link.*/ +opus_uint32 op_serialno(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the channel count of the given link in a (possibly-chained) Ogg Opus + stream. + This is equivalent to op_head(_of,_li)->channel_count, but + is provided for convenience. + This function may be called on partially-opened streams, but it will always + return the channel count of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the channel count. + \param _li The index of the link whose channel count should be retrieved. + Use a negative number to get the channel count of the current + link. + \return The channel count of the given link. + If \a _li is greater than the total number of links, this returns + the channel count of the last link. + If the stream is not seekable, this always returns the channel count + of the current link.*/ +int op_channel_count(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total (compressed) size of the stream, or of an individual link in + a (possibly-chained) Ogg Opus stream, including all headers and Ogg muxing + overhead. + \warning If the Opus stream (or link) is concurrently multiplexed with other + logical streams (e.g., video), this returns the size of the entire stream + (or link), not just the number of bytes in the first logical Opus stream. + Returning the latter would require scanning the entire file. + \param _of The \c OggOpusFile from which to retrieve the compressed size. + \param _li The index of the link whose compressed size should be computed. + Use a negative number to get the compressed size of the entire + stream. + \return The compressed size of the entire stream if \a _li is negative, the + compressed size of link \a _li if it is non-negative, or a negative + value on error. + The compressed size of the entire stream may be smaller than that + of the underlying stream if trailing garbage was detected in the + file. + \retval #OP_EINVAL The stream is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +opus_int64 op_raw_total(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total PCM length (number of samples at 48 kHz) of the stream, or of + an individual link in a (possibly-chained) Ogg Opus stream. + Users looking for op_time_total() should use op_pcm_total() + instead. + Because timestamps in Opus are fixed at 48 kHz, there is no need for a + separate function to convert this to seconds (and leaving it out avoids + introducing floating point to the API, for those that wish to avoid it). + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \param _li The index of the link whose PCM length should be computed. + Use a negative number to get the PCM length of the entire stream. + \return The PCM length of the entire stream if \a _li is negative, the PCM + length of link \a _li if it is non-negative, or a negative value on + error. + \retval #OP_EINVAL The stream is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +ogg_int64_t op_pcm_total(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the ID header information for the given link in a (possibly chained) Ogg + Opus stream. + This function may be called on partially-opened streams, but it will always + return the ID header information of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the ID header + information. + \param _li The index of the link whose ID header information should be + retrieved. + Use a negative number to get the ID header information of the + current link. + For an unseekable stream, \a _li is ignored, and the ID header + information for the current link is always returned, if + available. + \return The contents of the ID header for the given link.*/ +const OpusHead *op_head(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the comment header information for the given link in a (possibly + chained) Ogg Opus stream. + This function may be called on partially-opened streams, but it will always + return the tags from the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the comment header + information. + \param _li The index of the link whose comment header information should be + retrieved. + Use a negative number to get the comment header information of + the current link. + For an unseekable stream, \a _li is ignored, and the comment + header information for the current link is always returned, if + available. + \return The contents of the comment header for the given link, or + NULL if this is an unseekable stream that encountered + an invalid link.*/ +const OpusTags *op_tags(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Retrieve the index of the current link. + This is the link that produced the data most recently read by + op_read_float() or its associated functions, or, after a seek, the link + that the seek target landed in. + Reading more data may advance the link index (even on the first read after a + seek). + \param _of The \c OggOpusFile from which to retrieve the current link index. + \return The index of the current link on success, or a negative value on + failure. + For seekable streams, this is a number between 0 (inclusive) and the + value returned by op_link_count() (exclusive). + For unseekable streams, this value starts at 0 and increments by one + each time a new link is encountered (even though op_link_count() + always returns 1). + \retval #OP_EINVAL The stream was only partially open.*/ +int op_current_link(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Computes the bitrate of the stream, or of an individual link in a + (possibly-chained) Ogg Opus stream. + The stream must be seekable to compute the bitrate. + For unseekable streams, use op_bitrate_instant() to get periodic estimates. + \warning If the Opus stream (or link) is concurrently multiplexed with other + logical streams (e.g., video), this uses the size of the entire stream (or + link) to compute the bitrate, not just the number of bytes in the first + logical Opus stream. + Returning the latter requires scanning the entire file, but this may be done + by decoding the whole file and calling op_bitrate_instant() once at the + end. + Install a trivial decoding callback with op_set_decode_callback() if you + wish to skip actual decoding during this process. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \param _li The index of the link whose bitrate should be computed. + Use a negative number to get the bitrate of the whole stream. + \return The bitrate on success, or a negative value on error. + \retval #OP_EINVAL The stream was only partially open, the stream was not + seekable, or \a _li was larger than the number of + links.*/ +opus_int32 op_bitrate(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Compute the instantaneous bitrate, measured as the ratio of bits to playable + samples decoded since a) the last call to op_bitrate_instant(), b) the last + seek, or c) the start of playback, whichever was most recent. + This will spike somewhat after a seek or at the start/end of a chain + boundary, as pre-skip, pre-roll, and end-trimming causes samples to be + decoded but not played. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \return The bitrate, in bits per second, or a negative value on error. + \retval #OP_FALSE No data has been decoded since any of the events + described above. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int32 op_bitrate_instant(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the current value of the position indicator for \a _of. + \param _of The \c OggOpusFile from which to retrieve the position indicator. + \return The byte position that is currently being read from. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int64 op_raw_tell(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the PCM offset of the next sample to be read. + If the stream is not properly timestamped, this might not increment by the + proper amount between reads, or even return monotonically increasing + values. + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \return The PCM offset of the next sample to be read. + \retval #OP_EINVAL The stream was only partially open.*/ +ogg_int64_t op_pcm_tell(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_seeking Seeking*/ +/*@{*/ +/**\name Functions for seeking in Opus streams + + These functions let you seek in Opus streams, if the underlying stream + support it. + Seeking is implemented for all built-in stream I/O routines, though some + individual streams may not be seekable (pipes, live HTTP streams, or HTTP + streams from a server that does not support Range requests). + + op_raw_seek() is the fastest: it is guaranteed to perform at most one + physical seek, but, since the target is a byte position, makes no guarantee + how close to a given time it will come. + op_pcm_seek() provides sample-accurate seeking. + The number of physical seeks it requires is still quite small (often 1 or + 2, even in highly variable bitrate streams). + + Seeking in Opus requires decoding some pre-roll amount before playback to + allow the internal state to converge (as if recovering from packet loss). + This is handled internally by libopusfile, but means there is + little extra overhead for decoding up to the exact position requested + (since it must decode some amount of audio anyway). + It also means that decoding after seeking may not return exactly the same + values as would be obtained by decoding the stream straight through. + However, such differences are expected to be smaller than the loss + introduced by Opus's lossy compression.*/ +/*@{*/ + +/**Seek to a byte offset relative to the compressed data. + This also scans packets to update the PCM cursor. + It will cross a logical bitstream boundary, but only if it can't get any + packets out of the tail of the link to which it seeks. + \param _of The \c OggOpusFile in which to seek. + \param _byte_offset The byte position to seek to. + This must be between 0 and #op_raw_total(\a _of,\c -1) + (inclusive). + \return 0 on success, or a negative error code on failure. + \retval #OP_EREAD The underlying seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK Failed to initialize a decoder for a stream for an + unknown reason.*/ +int op_raw_seek(OggOpusFile *_of,opus_int64 _byte_offset) OP_ARG_NONNULL(1); + +/**Seek to the specified PCM offset, such that decoding will begin at exactly + the requested position. + \param _of The \c OggOpusFile in which to seek. + \param _pcm_offset The PCM offset to seek to. + This is in samples at 48 kHz relative to the start of the + stream. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read or seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK We failed to find data we had seen before, or the + bitstream structure was sufficiently malformed that + seeking to the target destination was impossible.*/ +int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_decoding Decoding*/ +/*@{*/ +/**\name Functions for decoding audio data + + These functions retrieve actual decoded audio data from the stream. + The general functions, op_read() and op_read_float() return 16-bit or + floating-point output, both using native endian ordering. + The number of channels returned can change from link to link in a chained + stream. + There are special functions, op_read_stereo() and op_read_float_stereo(), + which always output two channels, to simplify applications which do not + wish to handle multichannel audio. + These downmix multichannel files to two channels, so they can always return + samples in the same format for every link in a chained file. + + If the rest of your audio processing chain can handle floating point, the + floating-point routines should be preferred, as they prevent clipping and + other issues which might be avoided entirely if, e.g., you scale down the + volume at some other stage. + However, if you intend to consume 16-bit samples directly, the conversion in + libopusfile provides noise-shaping dithering and, if compiled + against libopus 1.1 or later, soft-clipping prevention. + + libopusfile can also be configured at compile time to use the + fixed-point libopus API. + If so, libopusfile's floating-point API may also be disabled. + In that configuration, nothing in libopusfile will use any + floating-point operations, to simplify support on devices without an + adequate FPU. + + \warning HTTPS streams may be be vulnerable to truncation attacks if you do + not check the error return code from op_read_float() or its associated + functions. + If the remote peer does not close the connection gracefully (with a TLS + "close notify" message), these functions will return #OP_EREAD instead of 0 + when they reach the end of the file. + If you are reading from an URL (particularly if seeking is not + supported), you should make sure to check for this error and warn the user + appropriately.*/ +/*@{*/ + +/**Indicates that the decoding callback should produce signed 16-bit + native-endian output samples.*/ +#define OP_DEC_FORMAT_SHORT (7008) +/**Indicates that the decoding callback should produce 32-bit native-endian + float samples.*/ +#define OP_DEC_FORMAT_FLOAT (7040) + +/**Indicates that the decoding callback did not decode anything, and that + libopusfile should decode normally instead.*/ +#define OP_DEC_USE_DEFAULT (6720) + +/**Called to decode an Opus packet. + This should invoke the functional equivalent of opus_multistream_decode() or + opus_multistream_decode_float(), except that it returns 0 on success + instead of the number of decoded samples (which is known a priori). + \param _ctx The application-provided callback context. + \param _decoder The decoder to use to decode the packet. + \param[out] _pcm The buffer to decode into. + This will always have enough room for \a _nchannels of + \a _nsamples samples, which should be placed into this + buffer interleaved. + \param _op The packet to decode. + This will always have its granule position set to a valid + value. + \param _nsamples The number of samples expected from the packet. + \param _nchannels The number of channels expected from the packet. + \param _format The desired sample output format. + This is either #OP_DEC_FORMAT_SHORT or + #OP_DEC_FORMAT_FLOAT. + \param _li The index of the link from which this packet was decoded. + \return A non-negative value on success, or a negative value on error. + Any error codes should be the same as those returned by + opus_multistream_decode() or opus_multistream_decode_float(). + Success codes are as follows: + \retval 0 Decoding was successful. + The application has filled the buffer with + exactly \a _nsamples*\a + _nchannels samples in the requested + format. + \retval #OP_DEC_USE_DEFAULT No decoding was done. + libopusfile should do the decoding + by itself instead.*/ +typedef int (*op_decode_cb_func)(void *_ctx,OpusMSDecoder *_decoder,void *_pcm, + const ogg_packet *_op,int _nsamples,int _nchannels,int _format,int _li); + +/**Sets the packet decode callback function. + If set, this is called once for each packet that needs to be decoded. + This can be used by advanced applications to do additional processing on the + compressed or uncompressed data. + For example, an application might save the final entropy coder state for + debugging and testing purposes, or it might apply additional filters + before the downmixing, dithering, or soft-clipping performed by + libopusfile, so long as these filters do not introduce any + latency. + + A call to this function is no guarantee that the audio will eventually be + delivered to the application. + libopusfile may discard some or all of the decoded audio data + (i.e., at the beginning or end of a link, or after a seek), however the + callback is still required to provide all of it. + \param _of The \c OggOpusFile on which to set the decode callback. + \param _decode_cb The callback function to call. + This may be NULL to disable calling the + callback. + \param _ctx The application-provided context pointer to pass to the + callback on each call.*/ +void op_set_decode_callback(OggOpusFile *_of, + op_decode_cb_func _decode_cb,void *_ctx) OP_ARG_NONNULL(1); + +/**Gain offset type that indicates that the provided offset is relative to the + header gain. + This is the default.*/ +#define OP_HEADER_GAIN (0) + +/**Gain offset type that indicates that the provided offset is relative to the + R128_ALBUM_GAIN value (if any), in addition to the header gain.*/ +#define OP_ALBUM_GAIN (3007) + +/**Gain offset type that indicates that the provided offset is relative to the + R128_TRACK_GAIN value (if any), in addition to the header gain.*/ +#define OP_TRACK_GAIN (3008) + +/**Gain offset type that indicates that the provided offset should be used as + the gain directly, without applying any the header or track gains.*/ +#define OP_ABSOLUTE_GAIN (3009) + +/**Sets the gain to be used for decoded output. + By default, the gain in the header is applied with no additional offset. + The total gain (including header gain and/or track gain, if applicable, and + this offset), will be clamped to [-32768,32767]/256 dB. + This is more than enough to saturate or underflow 16-bit PCM. + \note The new gain will not be applied to any already buffered, decoded + output. + This means you cannot change it sample-by-sample, as at best it will be + updated packet-by-packet. + It is meant for setting a target volume level, rather than applying smooth + fades, etc. + \param _of The \c OggOpusFile on which to set the gain offset. + \param _gain_type One of #OP_HEADER_GAIN, #OP_ALBUM_GAIN, + #OP_TRACK_GAIN, or #OP_ABSOLUTE_GAIN. + \param _gain_offset_q8 The gain offset to apply, in 1/256ths of a dB. + \return 0 on success or a negative value on error. + \retval #OP_EINVAL The \a _gain_type was unrecognized.*/ +int op_set_gain_offset(OggOpusFile *_of, + int _gain_type,opus_int32 _gain_offset_q8) OP_ARG_NONNULL(1); + +/**Sets whether or not dithering is enabled for 16-bit decoding. + By default, when libopusfile is compiled to use floating-point + internally, calling op_read() or op_read_stereo() will first decode to + float, and then convert to fixed-point using noise-shaping dithering. + This flag can be used to disable that dithering. + When the application uses op_read_float() or op_read_float_stereo(), or when + the library has been compiled to decode directly to fixed point, this flag + has no effect. + \param _of The \c OggOpusFile on which to enable or disable dithering. + \param _enabled A non-zero value to enable dithering, or 0 to disable it.*/ +void op_set_dither_enabled(OggOpusFile *_of,int _enabled) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + per channel. + This is done because +
    +
  1. The channel count cannot be known a priori (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,
    2. +
  2. Returning the samples per channel matches the libopus API + as closely as we're able,
    3. +
  3. Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and
    4. +
  4. This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).
    5. +
+ \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values at 48 kHz + with a nominal range of [-32768,32767). + Multiple channels are interleaved using the + Vorbis + channel ordering. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + values per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + libopusfile may return less data than + requested. + If so, there is no guarantee that the remaining data + in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass NULL if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + op_head(_of,*_li). + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + per channel. +
    +
  1. The channel count cannot be known a priori (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,
    2. +
  2. Returning the samples per channel matches the libopus API + as closely as we're able,
    3. +
  3. Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and
    4. +
  4. This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).
    5. +
+ \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples as + signed floats at 48 kHz with a nominal range of + [-1.0,1.0]. + Multiple channels are interleaved using the + Vorbis + channel ordering. + This must have room for at least \a _buf_size floats. + \param _buf_size The number of floats that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + samples per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass NULL if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + op_head(_of,*_li). + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of, + float *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples per + channel, even though the channel count is known, for consistency with + op_read(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values at 48 kHz + with a nominal range of [-32768,32767). + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples per + channel, even though the channel count is known, for consistency with + op_read_float(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed floats at 48 kHz with a nominal range of + [-1.0,1.0]. + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + that did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float_stereo(OggOpusFile *_of, + float *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility pop +# endif + +# if defined(__cplusplus) +} +# endif + +#endif diff --git a/engine/voice_codecs/celt/voiceencoder_celt.cpp b/engine/voice_codecs/celt/voiceencoder_celt.cpp index 5c3ee628..6aab093f 100644 --- a/engine/voice_codecs/celt/voiceencoder_celt.cpp +++ b/engine/voice_codecs/celt/voiceencoder_celt.cpp @@ -41,13 +41,13 @@ damage. */ #include "ivoicecodec.h" #include "iframeencoder.h" -#ifdef POSIX +/*#ifdef POSIX #include "source/osx/config.h" #else #include "source/msvc/config.h" -#endif +#endif*/ #include -#include "celt.h" +#include // NOTE: This has to be the last file included! @@ -67,21 +67,10 @@ struct celt_versions celt_versions g_CeltVersion[CELT_VERSION] = { - { - 44100, 256, 120 - }, - - { - 22050, 120, 60 - }, - - { - 22050, 256, 60 - }, - - { - 22050, 512, 64 - }, + {44100, 256, 120}, + {22050, 120, 60}, + {22050, 256, 60}, + {22050, 512, 64}, }; class VoiceEncoder_Celt : public IFrameEncoder diff --git a/engine/voice_codecs/celt/wscript b/engine/voice_codecs/celt/wscript new file mode 100755 index 00000000..f67c4afe --- /dev/null +++ b/engine/voice_codecs/celt/wscript @@ -0,0 +1,54 @@ +#! /usr/bin/env python +# encoding: utf-8 + +from waflib import Utils +import os + +top = '.' +PROJECT_NAME = 'vaudio_celt' + +def options(opt): + # stub + return + +def configure(conf): + conf.define('SPEEX_EXPORTS',1) +# conf.define('NO_HOOK_MALLOC',1) + +def build(bld): + source = [ + 'voiceencoder_celt.cpp', + '../frame_encoder/voice_codec_frame.cpp', + '../../../tier1/interface.cpp' + ] + + includes = [ + '.', + '../frame_encoder', + '../../../public', + '../../../public/tier1', + '../../../public/tier0', + '../../', + '../../../common', + '../../audio/public' + ] + + defines = [] + + libs = ['tier0','tier1','vstdlib','CELT'] + + install_path = bld.env.LIBDIR + + bld.shlib( + source = source, + target = PROJECT_NAME, + name = PROJECT_NAME, + features = 'c cxx', + includes = includes, + defines = defines, + use = libs, + install_path = install_path, + subsystem = bld.env.MSVC_SUBSYSTEM, + idx = bld.get_taskgen_count() + ) + diff --git a/engine/voice_codecs/opus/opus/opus.h b/engine/voice_codecs/opus/opus/opus.h new file mode 100644 index 00000000..d282f21d --- /dev/null +++ b/engine/voice_codecs/opus/opus/opus.h @@ -0,0 +1,981 @@ +/* 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 + *
    + *
  • audio_frame is the audio data in opus_int16 (or float for opus_encode_float())
    • + *
  • frame_size is the duration of the frame in samples (per channel)
    • + *
  • packet is the byte array to which the compressed data is written
    • + *
  • 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.
    • + *
+ * + * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet. + * The return value can be negative, 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 OpusEncoder structure. + * @param[in] channels int: 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 opus_int32: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) in input signal + * @param [in] application int: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY) + * @param [out] error int*: @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 OpusEncoder*: Encoder state + * @param [in] Fs opus_int32: Sampling rate of input signal (Hz) + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) in input signal + * @param [in] application int: 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 OpusEncoder*: Encoder state + * @param [in] pcm opus_int16*: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusEncoder*: Encoder state + * @param [in] pcm float*: 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 int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusEncoder allocated by opus_encoder_create(). + * @param[in] st OpusEncoder*: 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 OpusEncoder*: 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 OpusDecoder structure. + * @param [in] channels int: 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 opus_int32: Sample rate to decode at (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: Number of channels (1 or 2) to decode + * @param [out] error int*: #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 OpusDecoder*: Decoder state. + * @param [in] Fs opus_int32: Sampling rate to decode to (Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param [in] channels int: 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 OpusDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len opus_int32: Number of bytes in payload* + * @param [out] pcm opus_int16*: 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 must be a multiple of 2.5 ms. + * @param [in] decode_fec int: 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 OpusDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len opus_int32: Number of bytes in payload + * @param [out] pcm float*: 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 must be a multiple of 2.5 ms. + * @param [in] decode_fec int: 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 OpusDecoder*: 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 OpusDecoder allocated by opus_decoder_create(). + * @param[in] st OpusDecoder*: 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 char*: Opus packet to be parsed + * @param [in] len opus_int32: size of data + * @param [out] out_toc char*: TOC pointer + * @param [out] frames char*[48] encapsulated frames + * @param [out] size opus_int16[48] sizes of the encapsulated frames + * @param [out] payload_offset int*: 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 char*: 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 char*: Opus packet. + * This must contain at least one byte of + * data. + * @param [in] Fs opus_int32: 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 char*: 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 char*: Opus packet + * @param [in] len opus_int32: 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 char*: Opus packet + * @param [in] len opus_int32: Length of packet + * @param [in] Fs opus_int32: 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 OpusDecoder*: Decoder state + * @param [in] packet char*: Opus packet + * @param [in] len opus_int32: 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 float*: Input PCM and modified PCM + * @param [in] frame_size int Number of samples per channel to process + * @param [in] channels int: Number of channels + * @param [in,out] softclip_mem float*: 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 TARGET_DURATION_MS 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 OpusRepacketizer 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 OpusRepacketizer*: 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 OpusRepacketizer allocated by + * opus_repacketizer_create(). + * @param[in] rp OpusRepacketizer*: 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 OpusRepacketizer*: The repacketizer state to which to + * add the packet. + * @param[in] data const unsigned char*: 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 opus_int32: 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 OpusRepacketizer*: The repacketizer state from which to + * construct the new packet. + * @param begin int: The index of the first frame in the current + * repacketizer state to include in the output. + * @param end int: One past the index of the last frame in the + * current repacketizer state to include in the + * output. + * @param[out] data const unsigned char*: The buffer in which to + * store the output packet. + * @param maxlen opus_int32: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * 1276 for a single frame, + * or for multiple frames, + * 1277*(end-begin). + * However, 1*(end-begin) 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 [begin,end) 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 OpusRepacketizer*: 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 OpusRepacketizer*: The repacketizer state from which to + * construct the new packet. + * @param[out] data const unsigned char*: The buffer in which to + * store the output packet. + * @param maxlen opus_int32: The maximum number of bytes to store in + * the output buffer. In order to guarantee + * success, this should be at least + * 1277*opus_repacketizer_get_nb_frames(rp). + * However, + * 1*opus_repacketizer_get_nb_frames(rp) + * 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 const unsigned char*: The buffer containing the + * packet to pad. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param new_len opus_int32: 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 const unsigned char*: The buffer containing the + * packet to strip. + * @param len opus_int32: 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 const unsigned char*: The buffer containing the + * packet to pad. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param new_len opus_int32: The desired size of the packet after padding. + * This must be at least 1. + * @param nb_streams opus_int32: 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 const unsigned char*: The buffer containing the + * packet to strip. + * @param len opus_int32: The size of the packet. + * This must be at least 1. + * @param nb_streams opus_int32: 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 */ diff --git a/engine/voice_codecs/opus/opus/opus_custom.h b/engine/voice_codecs/opus/opus/opus_custom.h new file mode 100644 index 00000000..41f36bf2 --- /dev/null +++ b/engine/voice_codecs/opus/opus/opus_custom.h @@ -0,0 +1,342 @@ +/* Copyright (c) 2007-2008 CSIRO + Copyright (c) 2007-2009 Xiph.Org Foundation + Copyright (c) 2008-2012 Gregory Maxwell + Written by Jean-Marc Valin and Gregory Maxwell */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + @file opus_custom.h + @brief Opus-Custom reference implementation API + */ + +#ifndef OPUS_CUSTOM_H +#define OPUS_CUSTOM_H + +#include "opus_defines.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#ifdef CUSTOM_MODES +# define OPUS_CUSTOM_EXPORT OPUS_EXPORT +# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT +#else +# define OPUS_CUSTOM_EXPORT +# ifdef OPUS_BUILD +# define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE +# else +# define OPUS_CUSTOM_EXPORT_STATIC +# endif +#endif + +/** @defgroup opus_custom Opus Custom + * @{ + * Opus Custom is an optional part of the Opus specification and + * reference implementation which uses a distinct API from the regular + * API and supports frame sizes that are not normally supported.\ Use + * of Opus Custom is discouraged for all but very special applications + * for which a frame size different from 2.5, 5, 10, or 20 ms is needed + * (for either complexity or latency reasons) and where interoperability + * is less important. + * + * In addition to the interoperability limitations the use of Opus custom + * disables a substantial chunk of the codec and generally lowers the + * quality available at a given bitrate. Normally when an application needs + * a different frame size from the codec it should buffer to match the + * sizes but this adds a small amount of delay which may be important + * in some very low latency applications. Some transports (especially + * constant rate RF transports) may also work best with frames of + * particular durations. + * + * Libopus only supports custom modes if they are enabled at compile time. + * + * The Opus Custom API is similar to the regular API but the + * @ref opus_encoder_create and @ref opus_decoder_create calls take + * an additional mode parameter which is a structure produced by + * a call to @ref opus_custom_mode_create. Both the encoder and decoder + * must create a mode using the same sample rate (fs) and frame size + * (frame size) so these parameters must either be signaled out of band + * or fixed in a particular implementation. + * + * Similar to regular Opus the custom modes support on the fly frame size + * switching, but the sizes available depend on the particular frame size in + * use. For some initial frame sizes on a single on the fly size is available. + */ + +/** Contains the state of an encoder. One encoder state is needed + for each stream. It is initialized once at the beginning of the + stream. Do *not* re-initialize the state for every frame. + @brief Encoder state + */ +typedef struct OpusCustomEncoder OpusCustomEncoder; + +/** State of the decoder. One decoder state is needed for each stream. + It is initialized once at the beginning of the stream. Do *not* + re-initialize the state for every frame. + @brief Decoder state + */ +typedef struct OpusCustomDecoder OpusCustomDecoder; + +/** The mode contains all the information necessary to create an + encoder. Both the encoder and decoder need to be initialized + with exactly the same mode, otherwise the output will be + corrupted. + @brief Mode configuration + */ +typedef struct OpusCustomMode OpusCustomMode; + +/** Creates a new mode struct. This will be passed to an encoder or + * decoder. The mode MUST NOT BE DESTROYED until the encoders and + * decoders that use it are destroyed as well. + * @param [in] Fs int: Sampling rate (8000 to 96000 Hz) + * @param [in] frame_size int: Number of samples (per channel) to encode in each + * packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes) + * @param [out] error int*: Returned error code (if NULL, no error will be returned) + * @return A newly created mode + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error); + +/** Destroys a mode struct. Only call this after all encoders and + * decoders using this mode are destroyed as well. + * @param [in] mode OpusCustomMode*: Mode to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode); + + +#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C) + +/* Encoder */ +/** Gets the size of an OpusCustomEncoder structure. + * @param [in] mode OpusCustomMode *: Mode configuration + * @param [in] channels int: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +# ifdef CUSTOM_MODES +/** Initializes a previously allocated encoder state + * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_encoder_create(),opus_custom_encoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] mode OpusCustomMode *: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels int: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_init( + OpusCustomEncoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); +# endif +#endif + + +/** Creates a new encoder state. Each stream needs its own encoder + * state (can't be shared across simultaneous streams). + * @param [in] mode OpusCustomMode*: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * decoder) + * @param [in] channels int: Number of channels + * @param [out] error int*: Returns an error code + * @return Newly created encoder state. +*/ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + + +/** Destroys a an encoder state. + * @param[in] st OpusCustomEncoder*: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st); + +/** Encodes a frame of audio. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] pcm float*: PCM audio in float format, with a normal range of +/-1.0. + * Samples with a range beyond +/-1.0 are supported but will + * be clipped by decoders using the integer API and should + * only be used if it is known that the far end supports + * extended dynamic range. There must be exactly + * frame_size samples per channel. + * @param [in] frame_size int: Number of samples per frame of input signal + * @param [out] compressed char *: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes int: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float( + OpusCustomEncoder *st, + const float *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Encodes a frame of audio. + * @param [in] st OpusCustomEncoder*: Encoder state + * @param [in] pcm opus_int16*: PCM audio in signed 16-bit format (native endian). + * There must be exactly frame_size samples per channel. + * @param [in] frame_size int: Number of samples per frame of input signal + * @param [out] compressed char *: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long. + * @param [in] maxCompressedBytes int: Maximum number of bytes to use for compressing the frame + * (can change from one frame to another) + * @return Number of bytes written to "compressed". + * If negative, an error has occurred (see error codes). It is IMPORTANT that + * the length returned be somehow transmitted to the decoder. Otherwise, no + * decoding is possible. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode( + OpusCustomEncoder *st, + const opus_int16 *pcm, + int frame_size, + unsigned char *compressed, + int maxCompressedBytes +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom encoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_encoderctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + + +#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C) +/* Decoder */ + +/** Gets the size of an OpusCustomDecoder structure. + * @param [in] mode OpusCustomMode *: Mode configuration + * @param [in] channels int: Number of channels + * @returns size + */ +OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size( + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1); + +/** Initializes a previously allocated decoder state + * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size. + * This is intended for applications which use their own allocator instead of malloc. + * @see opus_custom_decoder_create(),opus_custom_decoder_get_size() + * To reset a previously initialized state use the OPUS_RESET_STATE CTL. + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] mode OpusCustomMode *: Contains all the information about the characteristics of + * the stream (must be the same characteristics as used for the + * encoder) + * @param [in] channels int: Number of channels + * @return OPUS_OK Success or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init( + OpusCustomDecoder *st, + const OpusCustomMode *mode, + int channels +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2); + +#endif + + +/** Creates a new decoder state. Each stream needs its own decoder state (can't + * be shared across simultaneous streams). + * @param [in] mode OpusCustomMode: Contains all the information about the characteristics of the + * stream (must be the same characteristics as used for the encoder) + * @param [in] channels int: Number of channels + * @param [out] error int*: Returns an error code + * @return Newly created decoder state. + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create( + const OpusCustomMode *mode, + int channels, + int *error +) OPUS_ARG_NONNULL(1); + +/** Destroys a an decoder state. + * @param[in] st OpusCustomDecoder*: State to be freed. + */ +OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st); + +/** Decode an opus custom frame with floating point output + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len int: Number of bytes in payload + * @param [out] pcm float*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(float) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + float *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Decode an opus custom frame + * @param [in] st OpusCustomDecoder*: Decoder state + * @param [in] data char*: Input payload. Use a NULL pointer to indicate packet loss + * @param [in] len int: Number of bytes in payload + * @param [out] pcm opus_int16*: Output signal (interleaved if 2 channels). length + * is frame_size*channels*sizeof(opus_int16) + * @param [in] frame_size Number of samples per channel of available space in *pcm. + * @returns Number of decoded samples or @ref opus_errorcodes + */ +OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode( + OpusCustomDecoder *st, + const unsigned char *data, + int len, + opus_int16 *pcm, + int frame_size +) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4); + +/** Perform a CTL function on an Opus custom decoder. + * + * Generally the request and subsequent arguments are generated + * by a convenience macro. + * @see opus_genericctls + */ +OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1); + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_CUSTOM_H */ diff --git a/engine/voice_codecs/opus/opus/opus_defines.h b/engine/voice_codecs/opus/opus/opus_defines.h new file mode 100644 index 00000000..d141418b --- /dev/null +++ b/engine/voice_codecs/opus/opus/opus_defines.h @@ -0,0 +1,799 @@ +/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited + Written by Jean-Marc Valin and Koen Vos */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_defines.h + * @brief Opus reference implementation constants + */ + +#ifndef OPUS_DEFINES_H +#define OPUS_DEFINES_H + +#include "opus_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @defgroup opus_errorcodes Error codes + * @{ + */ +/** No error @hideinitializer*/ +#define OPUS_OK 0 +/** One or more invalid/out of range arguments @hideinitializer*/ +#define OPUS_BAD_ARG -1 +/** Not enough bytes allocated in the buffer @hideinitializer*/ +#define OPUS_BUFFER_TOO_SMALL -2 +/** An internal error was detected @hideinitializer*/ +#define OPUS_INTERNAL_ERROR -3 +/** The compressed data passed is corrupted @hideinitializer*/ +#define OPUS_INVALID_PACKET -4 +/** Invalid/unsupported request number @hideinitializer*/ +#define OPUS_UNIMPLEMENTED -5 +/** An encoder or decoder structure is invalid or already freed @hideinitializer*/ +#define OPUS_INVALID_STATE -6 +/** Memory allocation has failed @hideinitializer*/ +#define OPUS_ALLOC_FAIL -7 +/**@}*/ + +/** @cond OPUS_INTERNAL_DOC */ +/**Export control for opus functions */ + +#ifndef OPUS_EXPORT +# if defined(WIN32) +# if defined(OPUS_BUILD) && defined(DLL_EXPORT) +# define OPUS_EXPORT __declspec(dllexport) +# else +# define OPUS_EXPORT +# endif +# elif defined(__GNUC__) && defined(OPUS_BUILD) +# define OPUS_EXPORT __attribute__ ((visibility ("default"))) +# else +# define OPUS_EXPORT +# endif +#endif + +# if !defined(OPUS_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OPUS_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OPUS_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(3,0) +# define OPUS_RESTRICT __restrict__ +# elif (defined(_MSC_VER) && _MSC_VER >= 1400) +# define OPUS_RESTRICT __restrict +# else +# define OPUS_RESTRICT +# endif +#else +# define OPUS_RESTRICT restrict +#endif + +#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) ) +# if OPUS_GNUC_PREREQ(2,7) +# define OPUS_INLINE __inline__ +# elif (defined(_MSC_VER)) +# define OPUS_INLINE __inline +# else +# define OPUS_INLINE +# endif +#else +# define OPUS_INLINE inline +#endif + +/**Warning attributes for opus functions + * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out + * some paranoid null checks. */ +#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__)) +#else +# define OPUS_WARN_UNUSED_RESULT +#endif +#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4) +# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x))) +#else +# define OPUS_ARG_NONNULL(_x) +#endif + +/** These are the actual Encoder CTL ID numbers. + * They should not be used directly by applications. + * In general, SETs should be even and GETs should be odd.*/ +#define OPUS_SET_APPLICATION_REQUEST 4000 +#define OPUS_GET_APPLICATION_REQUEST 4001 +#define OPUS_SET_BITRATE_REQUEST 4002 +#define OPUS_GET_BITRATE_REQUEST 4003 +#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004 +#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005 +#define OPUS_SET_VBR_REQUEST 4006 +#define OPUS_GET_VBR_REQUEST 4007 +#define OPUS_SET_BANDWIDTH_REQUEST 4008 +#define OPUS_GET_BANDWIDTH_REQUEST 4009 +#define OPUS_SET_COMPLEXITY_REQUEST 4010 +#define OPUS_GET_COMPLEXITY_REQUEST 4011 +#define OPUS_SET_INBAND_FEC_REQUEST 4012 +#define OPUS_GET_INBAND_FEC_REQUEST 4013 +#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014 +#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015 +#define OPUS_SET_DTX_REQUEST 4016 +#define OPUS_GET_DTX_REQUEST 4017 +#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020 +#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021 +#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022 +#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023 +#define OPUS_SET_SIGNAL_REQUEST 4024 +#define OPUS_GET_SIGNAL_REQUEST 4025 +#define OPUS_GET_LOOKAHEAD_REQUEST 4027 +/* #define OPUS_RESET_STATE 4028 */ +#define OPUS_GET_SAMPLE_RATE_REQUEST 4029 +#define OPUS_GET_FINAL_RANGE_REQUEST 4031 +#define OPUS_GET_PITCH_REQUEST 4033 +#define OPUS_SET_GAIN_REQUEST 4034 +#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */ +#define OPUS_SET_LSB_DEPTH_REQUEST 4036 +#define OPUS_GET_LSB_DEPTH_REQUEST 4037 +#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039 +#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040 +#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041 +#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042 +#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043 +/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */ +#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046 +#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047 +#define OPUS_GET_IN_DTX_REQUEST 4049 + +/** Defines for the presence of extended APIs. */ +#define OPUS_HAVE_OPUS_PROJECTION_H + +/* Macros to trigger compilation errors when the wrong types are provided to a CTL */ +#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x)) +#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr))) +#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr))) +#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr))) +/** @endcond */ + +/** @defgroup opus_ctlvalues Pre-defined values for CTL interface + * @see opus_genericctls, opus_encoderctls + * @{ + */ +/* Values for the various encoder CTLs */ +#define OPUS_AUTO -1000 /**opus_int32: 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 opus_int32 *: 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 opus_int32: 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 opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Hard CBR. For LPC/hybrid modes at very low bit-rate, this can + * cause noticeable quality degradation.
+ *
1
VBR (default). The exact type of VBR is controlled by + * #OPUS_SET_VBR_CONSTRAINT.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Hard CBR.
+ *
1
VBR (default). The exact type of VBR may be retrieved via + * #OPUS_GET_VBR_CONSTRAINT.
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
0
Unconstrained VBR.
+ *
1
Constrained VBR (default). This creates a maximum of one + * frame of buffering delay assuming a transport with a + * serialization speed of the nominal bitrate.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Unconstrained VBR.
+ *
1
Constrained VBR (default).
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
#OPUS_AUTO
Not forced (default)
+ *
1
Forced mono
+ *
2
Forced stereo
+ *
+ * @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 opus_int32 *: + *
+ *
#OPUS_AUTO
Not forced (default)
+ *
1
Forced mono
+ *
2
Forced stereo
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
OPUS_BANDWIDTH_FULLBAND
20 kHz passband (default)
+ *
+ * @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 opus_int32 *: Allowed values: + *
+ *
#OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
#OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
#OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
#OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
#OPUS_BANDWIDTH_FULLBAND
20 kHz passband (default)
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
#OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
#OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
#OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
#OPUS_BANDWIDTH_FULLBAND
20 kHz passband
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_SIGNAL_VOICE
Bias thresholds towards choosing LPC or Hybrid modes.
+ *
#OPUS_SIGNAL_MUSIC
Bias thresholds towards choosing MDCT modes.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_SIGNAL_VOICE
Bias thresholds towards choosing LPC or Hybrid modes.
+ *
#OPUS_SIGNAL_MUSIC
Bias thresholds towards choosing MDCT modes.
+ *
+ * @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 opus_int32: Returns one of the following values: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Disable inband FEC (default).
+ *
1
Enable inband FEC.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Inband FEC disabled (default).
+ *
1
Inband FEC enabled.
+ *
+ * @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 opus_int32: 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 opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Disable DTX (default).
+ *
1
Enabled DTX.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
DTX disabled (default).
+ *
1
DTX enabled.
+ *
+ * @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 opus_int32: 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 opus_int32 *: 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 really + * know what you are doing. + * @see OPUS_GET_EXPERT_FRAME_DURATION + * @param[in] x opus_int32: Allowed values: + *
+ *
OPUS_FRAMESIZE_ARG
Select frame size from the argument (default).
+ *
OPUS_FRAMESIZE_2_5_MS
Use 2.5 ms frames.
+ *
OPUS_FRAMESIZE_5_MS
Use 5 ms frames.
+ *
OPUS_FRAMESIZE_10_MS
Use 10 ms frames.
+ *
OPUS_FRAMESIZE_20_MS
Use 20 ms frames.
+ *
OPUS_FRAMESIZE_40_MS
Use 40 ms frames.
+ *
OPUS_FRAMESIZE_60_MS
Use 60 ms frames.
+ *
OPUS_FRAMESIZE_80_MS
Use 80 ms frames.
+ *
OPUS_FRAMESIZE_100_MS
Use 100 ms frames.
+ *
OPUS_FRAMESIZE_120_MS
Use 120 ms frames.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
OPUS_FRAMESIZE_ARG
Select frame size from the argument (default).
+ *
OPUS_FRAMESIZE_2_5_MS
Use 2.5 ms frames.
+ *
OPUS_FRAMESIZE_5_MS
Use 5 ms frames.
+ *
OPUS_FRAMESIZE_10_MS
Use 10 ms frames.
+ *
OPUS_FRAMESIZE_20_MS
Use 20 ms frames.
+ *
OPUS_FRAMESIZE_40_MS
Use 40 ms frames.
+ *
OPUS_FRAMESIZE_60_MS
Use 60 ms frames.
+ *
OPUS_FRAMESIZE_80_MS
Use 80 ms frames.
+ *
OPUS_FRAMESIZE_100_MS
Use 100 ms frames.
+ *
OPUS_FRAMESIZE_120_MS
Use 120 ms frames.
+ *
+ * @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 opus_int32: Allowed values: + *
+ *
0
Enable prediction (default).
+ *
1
Disable prediction.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Prediction enabled (default).
+ *
1
Prediction disabled.
+ *
+ * @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 opus_uint32 *: 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 opus_int32 *: Returns one of the following values: + *
+ *
#OPUS_AUTO
(default)
+ *
#OPUS_BANDWIDTH_NARROWBAND
4 kHz passband
+ *
#OPUS_BANDWIDTH_MEDIUMBAND
6 kHz passband
+ *
#OPUS_BANDWIDTH_WIDEBAND
8 kHz passband
+ *
#OPUS_BANDWIDTH_SUPERWIDEBAND
12 kHz passband
+ *
#OPUS_BANDWIDTH_FULLBAND
20 kHz passband
+ *
+ * @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 Fs value passed to opus_encoder_init() + * or opus_decoder_init(). + * @param[out] x opus_int32 *: 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 opus_int32: Allowed values: + *
+ *
0
Enable phase inversion (default).
+ *
1
Disable phase inversion.
+ *
+ * @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 opus_int32 *: Returns one of the following values: + *
+ *
0
Stereo phase inversion enabled (default).
+ *
1
Stereo phase inversion disabled.
+ *
+ * @hideinitializer */ +#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x) +/** Gets the DTX state of the encoder. + * Returns whether the last encoded frame was either a comfort noise update + * during DTX or not encoded because of DTX. + * @param[out] x opus_int32 *: Returns one of the following values: + *
+ *
0
The encoder is not in DTX.
+ *
1
The encoder is in DTX.
+ *
+ * @hideinitializer */ +#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x) + +/**@}*/ + +/** @defgroup opus_decoderctls Decoder related CTLs + * @see opus_genericctls, opus_encoderctls, opus_decoder + * @{ + */ + +/** Configures decoder gain adjustment. + * Scales the decoded output by a factor specified in Q8 dB units. + * This has a maximum range of -32768 to 32767 inclusive, and returns + * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment. + * This setting survives decoder reset. + * + * gain = pow(10, x/(20.0*256)) + * + * @param[in] x opus_int32: 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 opus_int32 *: 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 opus_int32 *: 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 opus_int32 *: 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 int: 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 */ diff --git a/engine/voice_codecs/opus/opus/opus_multistream.h b/engine/voice_codecs/opus/opus/opus_multistream.h new file mode 100644 index 00000000..babcee69 --- /dev/null +++ b/engine/voice_codecs/opus/opus/opus_multistream.h @@ -0,0 +1,660 @@ +/* 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 opus_int32: The index of the stream whose encoder you + * wish to retrieve. + * This must be non-negative and less than + * the streams parameter used + * to initialize the encoder. + * @param[out] y OpusEncoder**: 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 opus_int32: The index of the stream whose decoder you + * wish to retrieve. + * This must be non-negative and less than + * the streams parameter used + * to initialize the decoder. + * @param[out] y OpusDecoder**: 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 + * RFC 7845 + * and is based on the self-delimited Opus framing described in Appendix B of + * RFC 6716. + * Normal Opus packets are just a degenerate case of multistream Opus packets, + * and can be encoded or decoded with the multistream API by setting + * streams to 1 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 mapping table defines which decoded channel i + * should be used for each input/output (I/O) channel j. This table is + * typically provided as an unsigned char array. + * Let i = mapping[j] be the index for I/O channel j. + * If i < 2*coupled_streams, then I/O channel j is + * encoded as the left channel of stream (i/2) if i + * is even, or as the right channel of stream (i/2) if + * i is odd. Otherwise, I/O channel j is encoded as + * mono in stream (i - coupled_streams), 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 i must either + * be the special value 255 or be less than streams + coupled_streams. + * + * The output channels specified by the encoder + * should use the + * Vorbis + * channel ordering. 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 int: The total number of streams to encode from the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) 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 opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param[in] mapping const unsigned char[channels]: 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 int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @param[out] error int *: 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 OpusMSEncoder*: Multistream encoder state to initialize. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param[in] mapping const unsigned char[channels]: 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 int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 OpusMSEncoder*: Multistream encoder state. + * @param[in] pcm const opus_int16*: The input signal as interleaved + * samples. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusMSEncoder*: Multistream encoder state. + * @param[in] pcm const float*: 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 + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusMSEncoder allocated by + * opus_multistream_encoder_create(). + * @param st OpusMSEncoder*: 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 OpusMSEncoder*: 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 OpusMSDecoder structure. + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) 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 opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] mapping const unsigned char[channels]: Mapping from + * coded channels to output channels, as described in + * @ref opus_multistream. + * @param[out] error int *: 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 OpusMSEncoder*: Multistream encoder state to initialize. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] mapping const unsigned char[channels]: 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 OpusMSDecoder*: Multistream decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusMSDecoder*: Multistream decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusMSDecoder*: 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 OpusMSDecoder allocated by + * opus_multistream_decoder_create(). + * @param st OpusMSDecoder: Multistream decoder state to be freed. + */ +OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st); + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_MULTISTREAM_H */ diff --git a/engine/voice_codecs/opus/opus/opus_projection.h b/engine/voice_codecs/opus/opus/opus_projection.h new file mode 100644 index 00000000..9dabf4e8 --- /dev/null +++ b/engine/voice_codecs/opus/opus/opus_projection.h @@ -0,0 +1,568 @@ +/* Copyright (c) 2017 Google Inc. + Written by Andrew Allen */ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER + OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +/** + * @file opus_projection.h + * @brief Opus projection reference API + */ + +#ifndef OPUS_PROJECTION_H +#define OPUS_PROJECTION_H + +#include "opus_multistream.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** @cond OPUS_INTERNAL_DOC */ + +/** These are the actual encoder and decoder CTL ID numbers. + * They should not be used directly by applications.c + * In general, SETs should be even and GETs should be odd.*/ +/**@{*/ +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST 6001 +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST 6003 +#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST 6005 +/**@}*/ + + +/** @endcond */ + +/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs + * + * These are convenience macros that are specific to the + * opus_projection_encoder_ctl() and opus_projection_decoder_ctl() + * interface. + * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, + * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a + * projection encoder or decoder as well. + */ +/**@{*/ + +/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder. + * @param[out] x opus_int32 *: 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 opus_int32 *: 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 unsigned char *: Returns the demixing matrix to the + * supplied pointer location. + * @param y opus_int32: 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 int: The total number of input channels to encode. + * This must be no more than 255. + * @param mapping_family int: 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 opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param mapping_family int: The mapping family to use for selecting + * the appropriate projection. + * @param[out] streams int *: The total number of streams that will + * be encoded from the input. + * @param[out] coupled_streams int *: Number of coupled (2 channel) + * streams that will be encoded from the input. + * @param application int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @param[out] error int *: 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 OpusProjectionEncoder*: Projection encoder state to initialize. + * @param Fs opus_int32: Sampling rate of the input signal (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels in the input signal. + * This must be at most 255. + * It may be greater than the number of + * coded channels (streams + + * coupled_streams). + * @param streams int: The total number of streams to encode from the + * input. + * This must be no more than the number of channels. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than the number of input channels. + * @param application int: The target encoder application. + * This must be one of the following: + *
+ *
#OPUS_APPLICATION_VOIP
+ *
Process signal for improved speech intelligibility.
+ *
#OPUS_APPLICATION_AUDIO
+ *
Favor faithfulness to the original input.
+ *
#OPUS_APPLICATION_RESTRICTED_LOWDELAY
+ *
Configure the minimum possible coding delay by disabling certain modes + * of operation.
+ *
+ * @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 OpusProjectionEncoder*: Projection encoder state. + * @param[in] pcm const opus_int16*: The input signal as interleaved + * samples. + * This must contain + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusProjectionEncoder*: Projection encoder state. + * @param[in] pcm const float*: 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 + * frame_size*channels + * samples. + * @param frame_size int: 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 unsigned char*: Output payload. + * This must contain storage for at + * least \a max_data_bytes. + * @param [in] max_data_bytes opus_int32: 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 OpusProjectionEncoder allocated by + * opus_projection_ambisonics_encoder_create(). + * @param st OpusProjectionEncoder*: 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 OpusProjectionEncoder*: 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 OpusProjectionDecoder structure. + * @param channels int: The total number of output channels. + * This must be no more than 255. + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) 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 opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] demixing_matrix const unsigned char[demixing_matrix_size]: 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 opus_int32: The size in bytes of the + * demixing matrix, as + * described in @ref + * opus_projection_ctls. + * @param[out] error int *: 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 OpusProjectionDecoder*: Projection encoder state to initialize. + * @param Fs opus_int32: Sampling rate to decode at (in Hz). + * This must be one of 8000, 12000, 16000, + * 24000, or 48000. + * @param channels int: Number of channels to output. + * This must be at most 255. + * It may be different from the number of coded + * channels (streams + + * coupled_streams). + * @param streams int: The total number of streams coded in the + * input. + * This must be no more than 255. + * @param coupled_streams int: 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 (streams + + * coupled_streams) must be no + * more than 255. + * @param[in] demixing_matrix const unsigned char[demixing_matrix_size]: 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 opus_int32: 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 OpusProjectionDecoder*: Projection decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusProjectionDecoder*: Projection decoder state. + * @param[in] data const unsigned char*: Input payload. + * Use a NULL + * pointer to indicate packet + * loss. + * @param len opus_int32: Number of bytes in payload. + * @param[out] pcm opus_int16*: Output signal, with interleaved + * samples. + * This must contain room for + * frame_size*channels + * samples. + * @param frame_size int: 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 + * must be a multiple of 2.5 ms. + * @param decode_fec int: 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 OpusProjectionDecoder*: 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 OpusProjectionDecoder allocated by + * opus_projection_decoder_create(). + * @param st OpusProjectionDecoder: Projection decoder state to be freed. + */ +OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st); + + +/**@}*/ + +/**@}*/ + +#ifdef __cplusplus +} +#endif + +#endif /* OPUS_PROJECTION_H */ diff --git a/engine/voice_codecs/opus/opus/opus_types.h b/engine/voice_codecs/opus/opus/opus_types.h new file mode 100644 index 00000000..7cf67558 --- /dev/null +++ b/engine/voice_codecs/opus/opus/opus_types.h @@ -0,0 +1,166 @@ +/* (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 +# 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 + 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 + 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 + 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 */ diff --git a/engine/voice_codecs/opus/opus/opusfile.h b/engine/voice_codecs/opus/opus/opusfile.h new file mode 100644 index 00000000..7c1c89e0 --- /dev/null +++ b/engine/voice_codecs/opus/opus/opusfile.h @@ -0,0 +1,2164 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE libopusfile SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE libopusfile SOURCE CODE IS (C) COPYRIGHT 1994-2012 * + * by the Xiph.Org Foundation and contributors https://xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.h 17182 2010-04-29 03:48:32Z xiphmont $ + + ********************************************************************/ +#if !defined(_opusfile_h) +# define _opusfile_h (1) + +/**\mainpage + \section Introduction + + This is the documentation for the libopusfile C API. + + The libopusfile package provides a convenient high-level API for + decoding and basic manipulation of all Ogg Opus audio streams. + libopusfile is implemented as a layer on top of Xiph.Org's + reference + libogg + and + libopus + libraries. + + libopusfile provides several sets of built-in routines for + file/stream access, and may also use custom stream I/O routines provided by + the embedded environment. + There are built-in I/O routines provided for ANSI-compliant + stdio (FILE *), memory buffers, and URLs + (including URLs, plus optionally and URLs). + + \section Organization + + The main API is divided into several sections: + - \ref stream_open_close + - \ref stream_info + - \ref stream_decoding + - \ref stream_seeking + + Several additional sections are not tied to the main API. + - \ref stream_callbacks + - \ref header_info + - \ref error_codes + + \section Overview + + The libopusfile API always decodes files to 48 kHz. + The original sample rate is not preserved by the lossy compression, though + it is stored in the header to allow you to resample to it after decoding + (the libopusfile API does not currently provide a resampler, + but the + the + Speex resampler is a good choice if you need one). + In general, if you are playing back the audio, you should leave it at + 48 kHz, provided your audio hardware supports it. + When decoding to a file, it may be worth resampling back to the original + sample rate, so as not to surprise users who might not expect the sample + rate to change after encoding to Opus and decoding. + + Opus files can contain anywhere from 1 to 255 channels of audio. + The channel mappings for up to 8 channels are the same as the + Vorbis + mappings. + A special stereo API can convert everything to 2 channels, making it simple + to support multichannel files in an application which only has stereo + output. + Although the libopusfile ABI provides support for the theoretical + maximum number of channels, the current implementation does not support + files with more than 8 channels, as they do not have well-defined channel + mappings. + + Like all Ogg files, Opus files may be "chained". + That is, multiple Opus files may be combined into a single, longer file just + by concatenating the original files. + This is commonly done in internet radio streaming, as it allows the title + and artist to be updated each time the song changes, since each link in the + chain includes its own set of metadata. + + libopusfile fully supports chained files. + It will decode the first Opus stream found in each link of a chained file + (ignoring any other streams that might be concurrently multiplexed with it, + such as a video stream). + + The channel count can also change between links. + If your application is not prepared to deal with this, it can use the stereo + API to ensure the audio from all links will always get decoded into a + common format. + Since libopusfile always decodes to 48 kHz, you do not have to + worry about the sample rate changing between links (as was possible with + Vorbis). + This makes application support for chained files with libopusfile + very easy.*/ + +# if defined(__cplusplus) +extern "C" { +# endif + +# include +# include +# include +# include + +/**@cond PRIVATE*/ + +/*Enable special features for gcc and gcc-compatible compilers.*/ +# if !defined(OP_GNUC_PREREQ) +# if defined(__GNUC__)&&defined(__GNUC_MINOR__) +# define OP_GNUC_PREREQ(_maj,_min) \ + ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min)) +# else +# define OP_GNUC_PREREQ(_maj,_min) 0 +# endif +# endif + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility push(default) +# endif + +typedef struct OpusHead OpusHead; +typedef struct OpusTags OpusTags; +typedef struct OpusPictureTag OpusPictureTag; +typedef struct OpusServerInfo OpusServerInfo; +typedef struct OpusFileCallbacks OpusFileCallbacks; +typedef struct OggOpusFile OggOpusFile; + +/*Warning attributes for libopusfile functions.*/ +# if OP_GNUC_PREREQ(3,4) +# define OP_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__)) +# else +# define OP_WARN_UNUSED_RESULT +# endif +# if OP_GNUC_PREREQ(3,4) +# define OP_ARG_NONNULL(_x) __attribute__((__nonnull__(_x))) +# else +# define OP_ARG_NONNULL(_x) +# endif + +/**@endcond*/ + +/**\defgroup error_codes Error Codes*/ +/*@{*/ +/**\name List of possible error codes + Many of the functions in this library return a negative error code when a + function fails. + This list provides a brief explanation of the common errors. + See each individual function for more details on what a specific error code + means in that context.*/ +/*@{*/ + +/**A request did not succeed.*/ +#define OP_FALSE (-1) +/*Currently not used externally.*/ +#define OP_EOF (-2) +/**There was a hole in the page sequence numbers (e.g., a page was corrupt or + missing).*/ +#define OP_HOLE (-3) +/**An underlying read, seek, or tell operation failed when it should have + succeeded.*/ +#define OP_EREAD (-128) +/**A NULL pointer was passed where one was unexpected, or an + internal memory allocation failed, or an internal library error was + encountered.*/ +#define OP_EFAULT (-129) +/**The stream used a feature that is not implemented, such as an unsupported + channel family.*/ +#define OP_EIMPL (-130) +/**One or more parameters to a function were invalid.*/ +#define OP_EINVAL (-131) +/**A purported Ogg Opus stream did not begin with an Ogg page, a purported + header packet did not start with one of the required strings, "OpusHead" or + "OpusTags", or a link in a chained file was encountered that did not + contain any logical Opus streams.*/ +#define OP_ENOTFORMAT (-132) +/**A required header packet was not properly formatted, contained illegal + values, or was missing altogether.*/ +#define OP_EBADHEADER (-133) +/**The ID header contained an unrecognized version number.*/ +#define OP_EVERSION (-134) +/*Currently not used at all.*/ +#define OP_ENOTAUDIO (-135) +/**An audio packet failed to decode properly. + This is usually caused by a multistream Ogg packet where the durations of + the individual Opus packets contained in it are not all the same.*/ +#define OP_EBADPACKET (-136) +/**We failed to find data we had seen before, or the bitstream structure was + sufficiently malformed that seeking to the target destination was + impossible.*/ +#define OP_EBADLINK (-137) +/**An operation that requires seeking was requested on an unseekable stream.*/ +#define OP_ENOSEEK (-138) +/**The first or last granule position of a link failed basic validity checks.*/ +#define OP_EBADTIMESTAMP (-139) + +/*@}*/ +/*@}*/ + +/**\defgroup header_info Header Information*/ +/*@{*/ + +/**The maximum number of channels in an Ogg Opus stream.*/ +#define OPUS_CHANNEL_COUNT_MAX (255) + +/**Ogg Opus bitstream information. + This contains the basic playback parameters for a stream, and corresponds to + the initial ID header packet of an Ogg Opus stream.*/ +struct OpusHead{ + /**The Ogg Opus format version, in the range 0...255. + The top 4 bits represent a "major" version, and the bottom four bits + represent backwards-compatible "minor" revisions. + The current specification describes version 1. + This library will recognize versions up through 15 as backwards compatible + with the current specification. + An earlier draft of the specification described a version 0, but the only + difference between version 1 and version 0 is that version 0 did + not specify the semantics for handling the version field.*/ + int version; + /**The number of channels, in the range 1...255.*/ + int channel_count; + /**The number of samples that should be discarded from the beginning of the + stream.*/ + unsigned pre_skip; + /**The sampling rate of the original input. + All Opus audio is coded at 48 kHz, and should also be decoded at 48 kHz + for playback (unless the target hardware does not support this sampling + rate). + However, this field may be used to resample the audio back to the original + sampling rate, for example, when saving the output to a file.*/ + opus_uint32 input_sample_rate; + /**The gain to apply to the decoded output, in dB, as a Q8 value in the range + -32768...32767. + The libopusfile API will automatically apply this gain to the + decoded output before returning it, scaling it by + pow(10,output_gain/(20.0*256)). + You can adjust this behavior with op_set_gain_offset().*/ + int output_gain; + /**The channel mapping family, in the range 0...255. + Channel mapping family 0 covers mono or stereo in a single stream. + Channel mapping family 1 covers 1 to 8 channels in one or more streams, + using the Vorbis speaker assignments. + Channel mapping family 255 covers 1 to 255 channels in one or more + streams, but without any defined speaker assignment.*/ + int mapping_family; + /**The number of Opus streams in each Ogg packet, in the range 1...255.*/ + int stream_count; + /**The number of coupled Opus streams in each Ogg packet, in the range + 0...127. + This must satisfy 0 <= coupled_count <= stream_count and + coupled_count + stream_count <= 255. + The coupled streams appear first, before all uncoupled streams, in an Ogg + Opus packet.*/ + int coupled_count; + /**The mapping from coded stream channels to output channels. + Let index=mapping[k] be the value for channel k. + If index<2*coupled_count, then it refers to the left channel + from stream (index/2) if even, and the right channel from + stream (index/2) if odd. + Otherwise, it refers to the output of the uncoupled stream + (index-coupled_count).*/ + unsigned char mapping[OPUS_CHANNEL_COUNT_MAX]; +}; + +/**The metadata from an Ogg Opus stream. + + This structure holds the in-stream metadata corresponding to the 'comment' + header packet of an Ogg Opus stream. + The comment header is meant to be used much like someone jotting a quick + note on the label of a CD. + It should be a short, to the point text note that can be more than a couple + words, but not more than a short paragraph. + + The metadata is stored as a series of (tag, value) pairs, in length-encoded + string vectors, using the same format as Vorbis (without the final "framing + bit"), Theora, and Speex, except for the packet header. + The first occurrence of the '=' character delimits the tag and value. + A particular tag may occur more than once, and order is significant. + The character set encoding for the strings is always UTF-8, but the tag + names are limited to ASCII, and treated as case-insensitive. + See the Vorbis + comment header specification for details. + + In filling in this structure, libopusfile will null-terminate the + #user_comments strings for safety. + However, the bitstream format itself treats them as 8-bit clean vectors, + possibly containing NUL characters, so the #comment_lengths array should be + treated as their authoritative length. + + This structure is binary and source-compatible with a + vorbis_comment, and pointers to it may be freely cast to + vorbis_comment pointers, and vice versa. + It is provided as a separate type to avoid introducing a compile-time + dependency on the libvorbis headers.*/ +struct OpusTags{ + /**The array of comment string vectors.*/ + char **user_comments; + /**An array of the corresponding length of each vector, in bytes.*/ + int *comment_lengths; + /**The total number of comment streams.*/ + int comments; + /**The null-terminated vendor string. + This identifies the software used to encode the stream.*/ + char *vendor; +}; + +/**\name Picture tag image formats*/ +/*@{*/ + +/**The MIME type was not recognized, or the image data did not match the + declared MIME type.*/ +#define OP_PIC_FORMAT_UNKNOWN (-1) +/**The MIME type indicates the image data is really a URL.*/ +#define OP_PIC_FORMAT_URL (0) +/**The image is a JPEG.*/ +#define OP_PIC_FORMAT_JPEG (1) +/**The image is a PNG.*/ +#define OP_PIC_FORMAT_PNG (2) +/**The image is a GIF.*/ +#define OP_PIC_FORMAT_GIF (3) + +/*@}*/ + +/**The contents of a METADATA_BLOCK_PICTURE tag.*/ +struct OpusPictureTag{ + /**The picture type according to the ID3v2 APIC frame: +
    +
  1. Other
    2. +
  2. 32x32 pixels 'file icon' (PNG only)
    3. +
  3. Other file icon
    4. +
  4. Cover (front)
    5. +
  5. Cover (back)
    6. +
  6. Leaflet page
    7. +
  7. Media (e.g. label side of CD)
    8. +
  8. Lead artist/lead performer/soloist
    9. +
  9. Artist/performer
    10. +
  10. Conductor
    11. +
  11. Band/Orchestra
    12. +
  12. Composer
    13. +
  13. Lyricist/text writer
    14. +
  14. Recording Location
    15. +
  15. During recording
    16. +
  16. During performance
    17. +
  17. Movie/video screen capture
    18. +
  18. A bright colored fish
    19. +
  19. Illustration
    20. +
  20. Band/artist logotype
    21. +
  21. Publisher/Studio logotype
    22. +
+ Others are reserved and should not be used. + There may only be one each of picture type 1 and 2 in a file.*/ + opus_int32 type; + /**The MIME type of the picture, in printable ASCII characters 0x20-0x7E. + The MIME type may also be "-->" to signify that the data part + is a URL pointing to the picture instead of the picture data itself. + In this case, a terminating NUL is appended to the URL string in #data, + but #data_length is set to the length of the string excluding that + terminating NUL.*/ + char *mime_type; + /**The description of the picture, in UTF-8.*/ + char *description; + /**The width of the picture in pixels.*/ + opus_uint32 width; + /**The height of the picture in pixels.*/ + opus_uint32 height; + /**The color depth of the picture in bits-per-pixel (not + bits-per-channel).*/ + opus_uint32 depth; + /**For indexed-color pictures (e.g., GIF), the number of colors used, or 0 + for non-indexed pictures.*/ + opus_uint32 colors; + /**The length of the picture data in bytes.*/ + opus_uint32 data_length; + /**The binary picture data.*/ + unsigned char *data; + /**The format of the picture data, if known. + One of +
    +
  • #OP_PIC_FORMAT_UNKNOWN,
    • +
  • #OP_PIC_FORMAT_URL,
    • +
  • #OP_PIC_FORMAT_JPEG,
    • +
  • #OP_PIC_FORMAT_PNG, or
    • +
  • #OP_PIC_FORMAT_GIF.
    • +
*/ + int format; +}; + +/**\name Functions for manipulating header data + + These functions manipulate the #OpusHead and #OpusTags structures, + which describe the audio parameters and tag-value metadata, respectively. + These can be used to query the headers returned by libopusfile, or + to parse Opus headers from sources other than an Ogg Opus stream, provided + they use the same format.*/ +/*@{*/ + +/**Parses the contents of the ID header packet of an Ogg Opus stream. + \param[out] _head Returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be NULL to merely test the header + for validity. + \param[in] _data The contents of the ID header packet. + \param _len The number of bytes of data in the ID header packet. + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusHead" + string. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EIMPL If the channel mapping family was 255, which general + purpose players should not attempt to play. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification: +
    +
  • Insufficient data,
    • +
  • Too much data for the known minor versions,
    • +
  • An unrecognized channel mapping family,
    • +
  • Zero channels or too many channels,
    • +
  • Zero coded streams,
    • +
  • Too many coupled streams, or
    • +
  • An invalid channel mapping index.
    • +
*/ +OP_WARN_UNUSED_RESULT int opus_head_parse(OpusHead *_head, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Converts a granule position to a sample offset for a given Ogg Opus stream. + The sample offset is simply _gp-_head->pre_skip. + Granule position values smaller than OpusHead#pre_skip correspond to audio + that should never be played, and thus have no associated sample offset. + This function returns -1 for such values. + This function also correctly handles extremely large granule positions, + which may have wrapped around to a negative number when stored in a signed + ogg_int64_t value. + \param _head The #OpusHead information from the ID header of the stream. + \param _gp The granule position to convert. + \return The sample offset associated with the given granule position + (counting at a 48 kHz sampling rate), or the special value -1 on + error (i.e., the granule position was smaller than the pre-skip + amount).*/ +ogg_int64_t opus_granule_sample(const OpusHead *_head,ogg_int64_t _gp) + OP_ARG_NONNULL(1); + +/**Parses the contents of the 'comment' header packet of an Ogg Opus stream. + \param[out] _tags An uninitialized #OpusTags structure. + This returns the contents of the parsed packet. + The contents of this structure are untouched on error. + This may be NULL to merely test the header + for validity. + \param[in] _data The contents of the 'comment' header packet. + \param _len The number of bytes of data in the 'info' header packet. + \retval 0 Success. + \retval #OP_ENOTFORMAT If the data does not start with the "OpusTags" + string. + \retval #OP_EBADHEADER If the contents of the packet otherwise violate the + Ogg Opus specification. + \retval #OP_EFAULT If there wasn't enough memory to store the tags.*/ +OP_WARN_UNUSED_RESULT int opus_tags_parse(OpusTags *_tags, + const unsigned char *_data,size_t _len) OP_ARG_NONNULL(2); + +/**Performs a deep copy of an #OpusTags structure. + \param _dst The #OpusTags structure to copy into. + If this function fails, the contents of this structure remain + untouched. + \param _src The #OpusTags structure to copy from. + \retval 0 Success. + \retval #OP_EFAULT If there wasn't enough memory to copy the tags.*/ +int opus_tags_copy(OpusTags *_dst,const OpusTags *_src) OP_ARG_NONNULL(1); + +/**Initializes an #OpusTags structure. + This should be called on a freshly allocated #OpusTags structure before + attempting to use it. + \param _tags The #OpusTags structure to initialize.*/ +void opus_tags_init(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Add a (tag, value) pair to an initialized #OpusTags structure. + \note Neither opus_tags_add() nor opus_tags_add_comment() support values + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the (tag, value) pair to. + \param _tag A NUL-terminated, case-insensitive, ASCII string containing + the tag to add (without an '=' character). + \param _value A NUL-terminated UTF-8 containing the corresponding value. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add(OpusTags *_tags,const char *_tag,const char *_value) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) OP_ARG_NONNULL(3); + +/**Add a comment to an initialized #OpusTags structure. + \note Neither opus_tags_add_comment() nor opus_tags_add() support comments + containing embedded NULs, although the bitstream format does support them. + To add such tags, you will need to manipulate the #OpusTags structure + directly. + \param _tags The #OpusTags structure to add the comment to. + \param _comment A NUL-terminated UTF-8 string containing the comment in + "TAG=value" form. + \return 0 on success, or a negative value on failure. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_add_comment(OpusTags *_tags,const char *_comment) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Replace the binary suffix data at the end of the packet (if any). + \param _tags An initialized #OpusTags structure. + \param _data A buffer of binary data to append after the encoded user + comments. + The least significant bit of the first byte of this data must + be set (to ensure the data is preserved by other editors). + \param _len The number of bytes of binary data to append. + This may be zero to remove any existing binary suffix data. + \return 0 on success, or a negative value on error. + \retval #OP_EINVAL \a _len was negative, or \a _len was positive but + \a _data was NULL or the least significant + bit of the first byte was not set. + \retval #OP_EFAULT An internal memory allocation failed.*/ +int opus_tags_set_binary_suffix(OpusTags *_tags, + const unsigned char *_data,int _len) OP_ARG_NONNULL(1); + +/**Look up a comment value by its tag. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \param _count The instance of the tag. + The same tag can appear multiple times, each with a distinct + value, so an index is required to retrieve them all. + The order in which these values appear is significant and + should be preserved. + Use opus_tags_query_count() to get the legal range for the + \a _count parameter. + \return A pointer to the queried tag's value. + This points directly to data in the #OpusTags structure. + It should not be modified or freed by the application, and + modifications to the structure may invalidate the pointer. + \retval NULL If no matching tag is found.*/ +const char *opus_tags_query(const OpusTags *_tags,const char *_tag,int _count) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Look up the number of instances of a tag. + Call this first when querying for a specific tag and then iterate over the + number of instances with separate calls to opus_tags_query() to retrieve + all the values for that tag in order. + \param _tags An initialized #OpusTags structure. + \param _tag The tag to look up. + \return The number of instances of this particular tag.*/ +int opus_tags_query_count(const OpusTags *_tags,const char *_tag) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Retrieve the binary suffix data at the end of the packet (if any). + \param _tags An initialized #OpusTags structure. + \param[out] _len Returns the number of bytes of binary suffix data returned. + \return A pointer to the binary suffix data, or NULL if none + was present.*/ +const unsigned char *opus_tags_get_binary_suffix(const OpusTags *_tags, + int *_len) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Get the album gain from an R128_ALBUM_GAIN tag, if one was specified. + This searches for the first R128_ALBUM_GAIN tag with a valid signed, + 16-bit decimal integer value and returns the value. + This routine is exposed merely for convenience for applications which wish + to do something special with the album gain (i.e., display it). + If you simply wish to apply the album gain instead of the header gain, you + can use op_set_gain_offset() with an #OP_ALBUM_GAIN type and no offset. + \param _tags An initialized #OpusTags structure. + \param[out] _gain_q8 The album gain, in 1/256ths of a dB. + This will lie in the range [-32768,32767], and should + be applied in addition to the header gain. + On error, no value is returned, and the previous + contents remain unchanged. + \return 0 on success, or a negative value on error. + \retval #OP_FALSE There was no album gain available in the given tags.*/ +int opus_tags_get_album_gain(const OpusTags *_tags,int *_gain_q8) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Get the track gain from an R128_TRACK_GAIN tag, if one was specified. + This searches for the first R128_TRACK_GAIN tag with a valid signed, + 16-bit decimal integer value and returns the value. + This routine is exposed merely for convenience for applications which wish + to do something special with the track gain (i.e., display it). + If you simply wish to apply the track gain instead of the header gain, you + can use op_set_gain_offset() with an #OP_TRACK_GAIN type and no offset. + \param _tags An initialized #OpusTags structure. + \param[out] _gain_q8 The track gain, in 1/256ths of a dB. + This will lie in the range [-32768,32767], and should + be applied in addition to the header gain. + On error, no value is returned, and the previous + contents remain unchanged. + \return 0 on success, or a negative value on error. + \retval #OP_FALSE There was no track gain available in the given tags.*/ +int opus_tags_get_track_gain(const OpusTags *_tags,int *_gain_q8) + OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Clears the #OpusTags structure. + This should be called on an #OpusTags structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _tags The #OpusTags structure to clear.*/ +void opus_tags_clear(OpusTags *_tags) OP_ARG_NONNULL(1); + +/**Check if \a _comment is an instance of a \a _tag_name tag. + \see opus_tagncompare + \param _tag_name A NUL-terminated, case-insensitive, ASCII string containing + the name of the tag to check for (without the terminating + '=' character). + \param _comment The comment string to check. + \return An integer less than, equal to, or greater than zero if \a _comment + is found respectively, to be less than, to match, or be greater + than a "tag=value" string whose tag matches \a _tag_name.*/ +int opus_tagcompare(const char *_tag_name,const char *_comment); + +/**Check if \a _comment is an instance of a \a _tag_name tag. + This version is slightly more efficient than opus_tagcompare() if the length + of the tag name is already known (e.g., because it is a constant). + \see opus_tagcompare + \param _tag_name A case-insensitive ASCII string containing the name of the + tag to check for (without the terminating '=' character). + \param _tag_len The number of characters in the tag name. + This must be non-negative. + \param _comment The comment string to check. + \return An integer less than, equal to, or greater than zero if \a _comment + is found respectively, to be less than, to match, or be greater + than a "tag=value" string whose tag matches the first \a _tag_len + characters of \a _tag_name.*/ +int opus_tagncompare(const char *_tag_name,int _tag_len,const char *_comment); + +/**Parse a single METADATA_BLOCK_PICTURE tag. + This decodes the BASE64-encoded content of the tag and returns a structure + with the MIME type, description, image parameters (if known), and the + compressed image data. + If the MIME type indicates the presence of an image format we recognize + (JPEG, PNG, or GIF) and the actual image data contains the magic signature + associated with that format, then the OpusPictureTag::format field will be + set to the corresponding format. + This is provided as a convenience to avoid requiring applications to parse + the MIME type and/or do their own format detection for the commonly used + formats. + In this case, we also attempt to extract the image parameters directly from + the image data (overriding any that were present in the tag, which the + specification says applications are not meant to rely on). + The application must still provide its own support for actually decoding the + image data and, if applicable, retrieving that data from URLs. + \param[out] _pic Returns the parsed picture data. + No sanitation is done on the type, MIME type, or + description fields, so these might return invalid values. + The contents of this structure are left unmodified on + failure. + \param _tag The METADATA_BLOCK_PICTURE tag contents. + The leading "METADATA_BLOCK_PICTURE=" portion is optional, + to allow the function to be used on either directly on the + values in OpusTags::user_comments or on the return value + of opus_tags_query(). + \return 0 on success or a negative value on error. + \retval #OP_ENOTFORMAT The METADATA_BLOCK_PICTURE contents were not valid. + \retval #OP_EFAULT There was not enough memory to store the picture tag + contents.*/ +OP_WARN_UNUSED_RESULT int opus_picture_tag_parse(OpusPictureTag *_pic, + const char *_tag) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Initializes an #OpusPictureTag structure. + This should be called on a freshly allocated #OpusPictureTag structure + before attempting to use it. + \param _pic The #OpusPictureTag structure to initialize.*/ +void opus_picture_tag_init(OpusPictureTag *_pic) OP_ARG_NONNULL(1); + +/**Clears the #OpusPictureTag structure. + This should be called on an #OpusPictureTag structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _pic The #OpusPictureTag structure to clear.*/ +void opus_picture_tag_clear(OpusPictureTag *_pic) OP_ARG_NONNULL(1); + +/*@}*/ + +/*@}*/ + +/**\defgroup url_options URL Reading Options*/ +/*@{*/ +/**\name URL reading options + Options for op_url_stream_create() and associated functions. + These allow you to provide proxy configuration parameters, skip SSL + certificate checks, etc. + Options are processed in order, and if the same option is passed multiple + times, only the value specified by the last occurrence has an effect + (unless otherwise specified). + They may be expanded in the future.*/ +/*@{*/ + +/**@cond PRIVATE*/ + +/*These are the raw numbers used to define the request codes. + They should not be used directly.*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST (6464) +#define OP_HTTP_PROXY_HOST_REQUEST (6528) +#define OP_HTTP_PROXY_PORT_REQUEST (6592) +#define OP_HTTP_PROXY_USER_REQUEST (6656) +#define OP_HTTP_PROXY_PASS_REQUEST (6720) +#define OP_GET_SERVER_INFO_REQUEST (6784) + +#define OP_URL_OPT(_request) ((char *)(_request)) + +/*These macros trigger compilation errors or warnings if the wrong types are + provided to one of the URL options.*/ +#define OP_CHECK_INT(_x) ((void)((_x)==(opus_int32)0),(opus_int32)(_x)) +#define OP_CHECK_CONST_CHAR_PTR(_x) ((_x)+((_x)-(const char *)(_x))) +#define OP_CHECK_SERVER_INFO_PTR(_x) ((_x)+((_x)-(OpusServerInfo *)(_x))) + +/**@endcond*/ + +/**HTTP/Shoutcast/Icecast server information associated with a URL.*/ +struct OpusServerInfo{ + /**The name of the server (icy-name/ice-name). + This is NULL if there was no icy-name or + ice-name header.*/ + char *name; + /**A short description of the server (icy-description/ice-description). + This is NULL if there was no icy-description or + ice-description header.*/ + char *description; + /**The genre the server falls under (icy-genre/ice-genre). + This is NULL if there was no icy-genre or + ice-genre header.*/ + char *genre; + /**The homepage for the server (icy-url/ice-url). + This is NULL if there was no icy-url or + ice-url header.*/ + char *url; + /**The software used by the origin server (Server). + This is NULL if there was no Server header.*/ + char *server; + /**The media type of the entity sent to the recepient (Content-Type). + This is NULL if there was no Content-Type + header.*/ + char *content_type; + /**The nominal stream bitrate in kbps (icy-br/ice-bitrate). + This is -1 if there was no icy-br or + ice-bitrate header.*/ + opus_int32 bitrate_kbps; + /**Flag indicating whether the server is public (1) or not + (0) (icy-pub/ice-public). + This is -1 if there was no icy-pub or + ice-public header.*/ + int is_public; + /**Flag indicating whether the server is using HTTPS instead of HTTP. + This is 0 unless HTTPS is being used. + This may not match the protocol used in the original URL if there were + redirections.*/ + int is_ssl; +}; + +/**Initializes an #OpusServerInfo structure. + All fields are set as if the corresponding header was not available. + \param _info The #OpusServerInfo structure to initialize. + \note If you use this function, you must link against libopusurl.*/ +void opus_server_info_init(OpusServerInfo *_info) OP_ARG_NONNULL(1); + +/**Clears the #OpusServerInfo structure. + This should be called on an #OpusServerInfo structure after it is no longer + needed. + It will free all memory used by the structure members. + \param _info The #OpusServerInfo structure to clear. + \note If you use this function, you must link against libopusurl.*/ +void opus_server_info_clear(OpusServerInfo *_info) OP_ARG_NONNULL(1); + +/**Skip the certificate check when connecting via TLS/SSL (https). + \param _b opus_int32: Whether or not to skip the certificate + check. + The check will be skipped if \a _b is non-zero, and will not be + skipped if \a _b is zero. + \hideinitializer*/ +#define OP_SSL_SKIP_CERTIFICATE_CHECK(_b) \ + OP_URL_OPT(OP_SSL_SKIP_CERTIFICATE_CHECK_REQUEST),OP_CHECK_INT(_b) + +/**Proxy connections through the given host. + If no port is specified via #OP_HTTP_PROXY_PORT, the port number defaults + to 8080 (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _host const char *: The proxy server hostname. + This may be NULL to disable the use of a proxy + server. + \hideinitializer*/ +#define OP_HTTP_PROXY_HOST(_host) \ + OP_URL_OPT(OP_HTTP_PROXY_HOST_REQUEST),OP_CHECK_CONST_CHAR_PTR(_host) + +/**Use the given port when proxying connections. + This option only has an effect if #OP_HTTP_PROXY_HOST is specified with a + non-NULL \a _host. + If this option is not provided, the proxy port number defaults to 8080 + (http-alt). + All proxy parameters are ignored for non-http and non-https URLs. + \param _port opus_int32: The proxy server port. + This must be in the range 0...65535 (inclusive), or the + URL function this is passed to will fail. + \hideinitializer*/ +#define OP_HTTP_PROXY_PORT(_port) \ + OP_URL_OPT(OP_HTTP_PROXY_PORT_REQUEST),OP_CHECK_INT(_port) + +/**Use the given user name for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _user const char *: The proxy server user name. + This may be NULL to disable proxy + authentication. + A non-NULL value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_PASS + are also specified with non-NULL + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_USER(_user) \ + OP_URL_OPT(OP_HTTP_PROXY_USER_REQUEST),OP_CHECK_CONST_CHAR_PTR(_user) + +/**Use the given password for authentication when proxying connections. + All proxy parameters are ignored for non-http and non-https URLs. + \param _pass const char *: The proxy server password. + This may be NULL to disable proxy + authentication. + A non-NULL value only has an effect + if #OP_HTTP_PROXY_HOST and #OP_HTTP_PROXY_USER + are also specified with non-NULL + arguments. + \hideinitializer*/ +#define OP_HTTP_PROXY_PASS(_pass) \ + OP_URL_OPT(OP_HTTP_PROXY_PASS_REQUEST),OP_CHECK_CONST_CHAR_PTR(_pass) + +/**Parse information about the streaming server (if any) and return it. + Very little validation is done. + In particular, OpusServerInfo::url may not be a valid URL, + OpusServerInfo::bitrate_kbps may not really be in kbps, and + OpusServerInfo::content_type may not be a valid MIME type. + The character set of the string fields is not specified anywhere, and should + not be assumed to be valid UTF-8. + \param _info OpusServerInfo *: Returns information about the server. + If there is any error opening the stream, the + contents of this structure remain + unmodified. + On success, fills in the structure with the + server information that was available, if + any. + After a successful return, the contents of + this structure should be freed by calling + opus_server_info_clear(). + \hideinitializer*/ +#define OP_GET_SERVER_INFO(_info) \ + OP_URL_OPT(OP_GET_SERVER_INFO_REQUEST),OP_CHECK_SERVER_INFO_PTR(_info) + +/*@}*/ +/*@}*/ + +/**\defgroup stream_callbacks Abstract Stream Reading Interface*/ +/*@{*/ +/**\name Functions for reading from streams + These functions define the interface used to read from and seek in a stream + of data. + A stream does not need to implement seeking, but the decoder will not be + able to seek if it does not do so. + These functions also include some convenience routines for working with + standard FILE pointers, complete streams stored in a single + block of memory, or URLs.*/ +/*@{*/ + +/**Reads up to \a _nbytes bytes of data from \a _stream. + \param _stream The stream to read from. + \param[out] _ptr The buffer to store the data in. + \param _nbytes The maximum number of bytes to read. + This function may return fewer, though it will not + return zero unless it reaches end-of-file. + \return The number of bytes successfully read, or a negative value on + error.*/ +typedef int (*op_read_func)(void *_stream,unsigned char *_ptr,int _nbytes); + +/**Sets the position indicator for \a _stream. + The new position, measured in bytes, is obtained by adding \a _offset + bytes to the position specified by \a _whence. + If \a _whence is set to SEEK_SET, SEEK_CUR, or + SEEK_END, the offset is relative to the start of the stream, + the current position indicator, or end-of-file, respectively. + \retval 0 Success. + \retval -1 Seeking is not supported or an error occurred. + errno need not be set.*/ +typedef int (*op_seek_func)(void *_stream,opus_int64 _offset,int _whence); + +/**Obtains the current value of the position indicator for \a _stream. + \return The current position indicator.*/ +typedef opus_int64 (*op_tell_func)(void *_stream); + +/**Closes the underlying stream. + \retval 0 Success. + \retval EOF An error occurred. + errno need not be set.*/ +typedef int (*op_close_func)(void *_stream); + +/**The callbacks used to access non-FILE stream resources. + The function prototypes are basically the same as for the stdio functions + fread(), fseek(), ftell(), and + fclose(). + The differences are that the FILE * arguments have been + replaced with a void *, which is to be used as a pointer to + whatever internal data these functions might need, that #seek and #tell + take and return 64-bit offsets, and that #seek must return -1 if + the stream is unseekable.*/ +struct OpusFileCallbacks{ + /**Used to read data from the stream. + This must not be NULL.*/ + op_read_func read; + /**Used to seek in the stream. + This may be NULL if seeking is not implemented.*/ + op_seek_func seek; + /**Used to return the current read position in the stream. + This may be NULL if seeking is not implemented.*/ + op_tell_func tell; + /**Used to close the stream when the decoder is freed. + This may be NULL to leave the stream open.*/ + op_close_func close; +}; + +/**Opens a stream with fopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + On Windows, this string must be UTF-8 (to allow access to + files whose names cannot be represented in the current + MBCS code page). + All other systems use the native character encoding. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2) + OP_ARG_NONNULL(3); + +/**Opens a stream with fdopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _fd The file descriptor to open. + \param _mode The mode to open the file in. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_fdopen(OpusFileCallbacks *_cb, + int _fd,const char *_mode) OP_ARG_NONNULL(1) OP_ARG_NONNULL(3); + +/**Opens a stream with freopen() and fills in a set of callbacks + that can be used to access it. + This is useful to avoid writing your own portable 64-bit seeking wrappers, + and also avoids cross-module linking issues on Windows, where a + FILE * must be accessed by routines defined in the same module + that opened it. + \param[out] _cb The callbacks to use for this file. + If there is an error opening the file, nothing will be + filled in here. + \param _path The path to the file to open. + On Windows, this string must be UTF-8 (to allow access + to files whose names cannot be represented in the + current MBCS code page). + All other systems use the native character encoding. + \param _mode The mode to open the file in. + \param _stream A stream previously returned by op_fopen(), op_fdopen(), + or op_freopen(). + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_freopen(OpusFileCallbacks *_cb, + const char *_path,const char *_mode,void *_stream) OP_ARG_NONNULL(1) + OP_ARG_NONNULL(2) OP_ARG_NONNULL(3) OP_ARG_NONNULL(4); + +/**Creates a stream that reads from the given block of memory. + This block of memory must contain the complete stream to decode. + This is useful for caching small streams (e.g., sound effects) in RAM. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _data The block of memory to read from. + \param _size The size of the block of memory. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_mem_stream_create(OpusFileCallbacks *_cb, + const unsigned char *_data,size_t _size) OP_ARG_NONNULL(1); + +/**Creates a stream that reads from the given URL. + This function behaves identically to op_url_stream_create(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will + be filled in here. + \param _url The URL to read from. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[in,out] _ap A list of the \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with NULL. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_vcreate(OpusFileCallbacks *_cb, + const char *_url,va_list _ap) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/**Creates a stream that reads from the given URL. + \note If you use this function, you must link against libopusurl. + \param[out] _cb The callbacks to use for this stream. + If there is an error creating the stream, nothing will be + filled in here. + \param _url The URL to read from. + Currently only the , , and schemes + are supported. + Both and may be disabled at compile time, + in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, before + passing them to this function. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + NULL. + \return A stream handle to use with the callbacks, or NULL on + error.*/ +OP_WARN_UNUSED_RESULT void *op_url_stream_create(OpusFileCallbacks *_cb, + const char *_url,...) OP_ARG_NONNULL(1) OP_ARG_NONNULL(2); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_open_close Opening and Closing*/ +/*@{*/ +/**\name Functions for opening and closing streams + + These functions allow you to test a stream to see if it is Opus, open it, + and close it. + Several flavors are provided for each of the built-in stream types, plus a + more general version which takes a set of application-provided callbacks.*/ +/*@{*/ + +/**Test to see if this is an Opus stream. + For good results, you will need at least 57 bytes (for a pure Opus-only + stream). + Something like 512 bytes will give more reliable results for multiplexed + streams. + This function is meant to be a quick-rejection filter. + Its purpose is not to guarantee that a stream is a valid Opus stream, but to + ensure that it looks enough like Opus that it isn't going to be recognized + as some other format (except possibly an Opus stream that is also + multiplexed with other codecs, such as video). + \param[out] _head The parsed ID header contents. + You may pass NULL if you do not need + this information. + If the function fails, the contents of this structure + remain untouched. + \param _initial_data An initial buffer of data from the start of the + stream. + \param _initial_bytes The number of bytes in \a _initial_data. + \return 0 if the data appears to be Opus, or a negative value on error. + \retval #OP_FALSE There was not enough data to tell if this was an Opus + stream or not. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_ENOTFORMAT If the data did not contain a recognizable ID + header for an Opus stream. + \retval #OP_EVERSION If the version field signaled a version this library + does not know how to parse. + \retval #OP_EBADHEADER The ID header was not properly formatted or contained + illegal values.*/ +int op_test(OpusHead *_head, + const unsigned char *_initial_data,size_t _initial_bytes); + +/**Open a stream from the given file path. + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Open a stream from a memory buffer. + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Open a stream from a URL. + This function behaves identically to op_open_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always + fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, + with internationalized domain names encoded in + punycode, before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with NULL. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vopen_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Open a stream from a URL. + \note If you use this function, you must link against libopusurl. + \param _url The URL to open. + Currently only the , , and schemes + are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated with + NULL. + \return A freshly opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Open a stream using the given set of callbacks to access it. + \param _stream The stream to read from (e.g., a FILE *). + This value will be passed verbatim as the first + argument to all of the callbacks. + \param _cb The callbacks with which to access the stream. + read() must + be implemented. + seek() and + tell() may + be NULL, or may always return -1 to + indicate a stream is unseekable, but if + seek() is + implemented and succeeds on a particular stream, then + tell() must + also. + close() may + be NULL, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be opened, even if it is unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + tell() + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + The failure code will be one of +
+
#OP_EREAD
+
An underlying read, seek, or tell operation + failed when it should have succeeded, or we failed + to find data in the stream we had seen before.
+
#OP_EFAULT
+
There was a memory allocation failure, or an + internal library error.
+
#OP_EIMPL
+
The stream used a feature that is not + implemented, such as an unsupported channel + family.
+
#OP_EINVAL
+
seek() + was implemented and succeeded on this source, but + tell() + did not, or the starting position indicator was + not equal to \a _initial_bytes.
+
#OP_ENOTFORMAT
+
The stream contained a link that did not have + any logical Opus streams in it.
+
#OP_EBADHEADER
+
A required header packet was not properly + formatted, contained illegal values, or was missing + altogether.
+
#OP_EVERSION
+
An ID header contained an unrecognized version + number.
+
#OP_EBADLINK
+
We failed to find data we had seen before after + seeking.
+
#OP_EBADTIMESTAMP
+
The first or last timestamp in a link failed + basic validity checks.
+
+ \return A freshly opened \c OggOpusFile, or NULL on error. + libopusfile does not take ownership of the stream + if the call fails. + The calling application is responsible for closing the stream if + this call returns an error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_open_callbacks(void *_stream, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Partially open a stream from the given file path. + \see op_test_callbacks + \param _path The path to the file to open. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + The failure code will be #OP_EFAULT if the file could not + be opened, or one of the other failure codes from + op_open_callbacks() otherwise. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_file(const char *_path,int *_error) + OP_ARG_NONNULL(1); + +/**Partially open a stream from a memory buffer. + \see op_test_callbacks + \param _data The memory buffer to open. + \param _size The number of bytes in the buffer. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure codes. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_memory(const unsigned char *_data, + size_t _size,int *_error); + +/**Partially open a stream from a URL. + This function behaves identically to op_test_url(), except that it + takes a va_list instead of a variable number of arguments. + It does not call the va_end macro, and because it invokes the + va_arg macro, the value of \a _ap is undefined after the call. + \note If you use this function, you must link against libopusurl. + \see op_test_url + \see op_test_callbacks + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always + fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, + with internationalized domain names encoded in + punycode, before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \param[in,out] _ap A list of the \ref url_options "optional flags" to + use. + This is a variable-length list of options terminated + with NULL. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_vtest_url(const char *_url, + int *_error,va_list _ap) OP_ARG_NONNULL(1); + +/**Partially open a stream from a URL. + \note If you use this function, you must link against libopusurl. + \see op_test_callbacks + \param _url The URL to open. + Currently only the , , and + schemes are supported. + Both and may be disabled at compile + time, in which case opening such URLs will always fail. + Currently this only supports URIs. + IRIs should be converted to UTF-8 and URL-escaped, with + internationalized domain names encoded in punycode, + before passing them to this function. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want the + failure code. + See op_open_callbacks() for a full list of failure + codes. + \param ... The \ref url_options "optional flags" to use. + This is a variable-length list of options terminated + with NULL. + \return A partially opened \c OggOpusFile, or NULL on error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_url(const char *_url, + int *_error,...) OP_ARG_NONNULL(1); + +/**Partially open a stream using the given set of callbacks to access it. + This tests for Opusness and loads the headers for the first link. + It does not seek (although it tests for seekability). + You can query a partially open stream for the few pieces of basic + information returned by op_serialno(), op_channel_count(), op_head(), and + op_tags() (but only for the first link). + You may also determine if it is seekable via a call to op_seekable(). + You cannot read audio from the stream, seek, get the size or duration, + get information from links other than the first one, or even get the total + number of links until you finish opening the stream with op_test_open(). + If you do not need to do any of these things, you can dispose of it with + op_free() instead. + + This function is provided mostly to simplify porting existing code that used + libvorbisfile. + For new code, you are likely better off using op_test() instead, which + is less resource-intensive, requires less data to succeed, and imposes a + hard limit on the amount of data it examines (important for unseekable + streams, where all such data must be buffered until you are sure of the + stream type). + \param _stream The stream to read from (e.g., a FILE *). + This value will be passed verbatim as the first + argument to all of the callbacks. + \param _cb The callbacks with which to access the stream. + read() must + be implemented. + seek() and + tell() may + be NULL, or may always return -1 to + indicate a stream is unseekable, but if + seek() is + implemented and succeeds on a particular stream, then + tell() must + also. + close() may + be NULL, but if it is not, it will be + called when the \c OggOpusFile is destroyed by + op_free(). + It will not be called if op_open_callbacks() fails + with an error. + \param _initial_data An initial buffer of data from the start of the + stream. + Applications can read some number of bytes from the + start of the stream to help identify this as an Opus + stream, and then provide them here to allow the + stream to be tested more thoroughly, even if it is + unseekable. + \param _initial_bytes The number of bytes in \a _initial_data. + If the stream is seekable, its current position (as + reported by + tell() + at the start of this function) must be equal to + \a _initial_bytes. + Otherwise, seeking to absolute positions will + generate inconsistent results. + \param[out] _error Returns 0 on success, or a failure code on error. + You may pass in NULL if you don't want + the failure code. + See op_open_callbacks() for a full list of failure + codes. + \return A partially opened \c OggOpusFile, or NULL on error. + libopusfile does not take ownership of the stream + if the call fails. + The calling application is responsible for closing the stream if + this call returns an error.*/ +OP_WARN_UNUSED_RESULT OggOpusFile *op_test_callbacks(void *_stream, + const OpusFileCallbacks *_cb,const unsigned char *_initial_data, + size_t _initial_bytes,int *_error) OP_ARG_NONNULL(2); + +/**Finish opening a stream partially opened with op_test_callbacks() or one of + the associated convenience functions. + If this function fails, you are still responsible for freeing the + \c OggOpusFile with op_free(). + \param _of The \c OggOpusFile to finish opening. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read, seek, or tell operation failed + when it should have succeeded. + \retval #OP_EFAULT There was a memory allocation failure, or an + internal library error. + \retval #OP_EIMPL The stream used a feature that is not implemented, + such as an unsupported channel family. + \retval #OP_EINVAL The stream was not partially opened with + op_test_callbacks() or one of the associated + convenience functions. + \retval #OP_ENOTFORMAT The stream contained a link that did not have any + logical Opus streams in it. + \retval #OP_EBADHEADER A required header packet was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An ID header contained an unrecognized version + number. + \retval #OP_EBADLINK We failed to find data we had seen before after + seeking. + \retval #OP_EBADTIMESTAMP The first or last timestamp in a link failed basic + validity checks.*/ +int op_test_open(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Release all memory used by an \c OggOpusFile. + \param _of The \c OggOpusFile to free.*/ +void op_free(OggOpusFile *_of); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_info Stream Information*/ +/*@{*/ +/**\name Functions for obtaining information about streams + + These functions allow you to get basic information about a stream, including + seekability, the number of links (for chained streams), plus the size, + duration, bitrate, header parameters, and meta information for each link + (or, where available, the stream as a whole). + Some of these (size, duration) are only available for seekable streams. + You can also query the current stream position, link, and playback time, + and instantaneous bitrate during playback. + + Some of these functions may be used successfully on the partially open + streams returned by op_test_callbacks() or one of the associated + convenience functions. + Their documention will indicate so explicitly.*/ +/*@{*/ + +/**Returns whether or not the stream being read is seekable. + This is true if +
    +
  1. The seek() and + tell() callbacks are both + non-NULL,
    2. +
  2. The seek() callback was + successfully executed at least once, and
    3. +
  3. The tell() callback was + successfully able to report the position indicator afterwards.
    4. +
+ This function may be called on partially-opened streams. + \param _of The \c OggOpusFile whose seekable status is to be returned. + \return A non-zero value if seekable, and 0 if unseekable.*/ +int op_seekable(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Returns the number of links in this chained stream. + This function may be called on partially-opened streams, but it will always + return 1. + The actual number of links is not known until the stream is fully opened. + \param _of The \c OggOpusFile from which to retrieve the link count. + \return For fully-open seekable streams, this returns the total number of + links in the whole stream, which will be at least 1. + For partially-open or unseekable streams, this always returns 1.*/ +int op_link_count(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Get the serial number of the given link in a (possibly-chained) Ogg Opus + stream. + This function may be called on partially-opened streams, but it will always + return the serial number of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the serial number. + \param _li The index of the link whose serial number should be retrieved. + Use a negative number to get the serial number of the current + link. + \return The serial number of the given link. + If \a _li is greater than the total number of links, this returns + the serial number of the last link. + If the stream is not seekable, this always returns the serial number + of the current link.*/ +opus_uint32 op_serialno(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the channel count of the given link in a (possibly-chained) Ogg Opus + stream. + This is equivalent to op_head(_of,_li)->channel_count, but + is provided for convenience. + This function may be called on partially-opened streams, but it will always + return the channel count of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the channel count. + \param _li The index of the link whose channel count should be retrieved. + Use a negative number to get the channel count of the current + link. + \return The channel count of the given link. + If \a _li is greater than the total number of links, this returns + the channel count of the last link. + If the stream is not seekable, this always returns the channel count + of the current link.*/ +int op_channel_count(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total (compressed) size of the stream, or of an individual link in + a (possibly-chained) Ogg Opus stream, including all headers and Ogg muxing + overhead. + \warning If the Opus stream (or link) is concurrently multiplexed with other + logical streams (e.g., video), this returns the size of the entire stream + (or link), not just the number of bytes in the first logical Opus stream. + Returning the latter would require scanning the entire file. + \param _of The \c OggOpusFile from which to retrieve the compressed size. + \param _li The index of the link whose compressed size should be computed. + Use a negative number to get the compressed size of the entire + stream. + \return The compressed size of the entire stream if \a _li is negative, the + compressed size of link \a _li if it is non-negative, or a negative + value on error. + The compressed size of the entire stream may be smaller than that + of the underlying stream if trailing garbage was detected in the + file. + \retval #OP_EINVAL The stream is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +opus_int64 op_raw_total(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the total PCM length (number of samples at 48 kHz) of the stream, or of + an individual link in a (possibly-chained) Ogg Opus stream. + Users looking for op_time_total() should use op_pcm_total() + instead. + Because timestamps in Opus are fixed at 48 kHz, there is no need for a + separate function to convert this to seconds (and leaving it out avoids + introducing floating point to the API, for those that wish to avoid it). + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \param _li The index of the link whose PCM length should be computed. + Use a negative number to get the PCM length of the entire stream. + \return The PCM length of the entire stream if \a _li is negative, the PCM + length of link \a _li if it is non-negative, or a negative value on + error. + \retval #OP_EINVAL The stream is not seekable (so we can't know the length), + \a _li wasn't less than the total number of links in + the stream, or the stream was only partially open.*/ +ogg_int64_t op_pcm_total(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the ID header information for the given link in a (possibly chained) Ogg + Opus stream. + This function may be called on partially-opened streams, but it will always + return the ID header information of the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the ID header + information. + \param _li The index of the link whose ID header information should be + retrieved. + Use a negative number to get the ID header information of the + current link. + For an unseekable stream, \a _li is ignored, and the ID header + information for the current link is always returned, if + available. + \return The contents of the ID header for the given link.*/ +const OpusHead *op_head(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Get the comment header information for the given link in a (possibly + chained) Ogg Opus stream. + This function may be called on partially-opened streams, but it will always + return the tags from the Opus stream in the first link. + \param _of The \c OggOpusFile from which to retrieve the comment header + information. + \param _li The index of the link whose comment header information should be + retrieved. + Use a negative number to get the comment header information of + the current link. + For an unseekable stream, \a _li is ignored, and the comment + header information for the current link is always returned, if + available. + \return The contents of the comment header for the given link, or + NULL if this is an unseekable stream that encountered + an invalid link.*/ +const OpusTags *op_tags(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Retrieve the index of the current link. + This is the link that produced the data most recently read by + op_read_float() or its associated functions, or, after a seek, the link + that the seek target landed in. + Reading more data may advance the link index (even on the first read after a + seek). + \param _of The \c OggOpusFile from which to retrieve the current link index. + \return The index of the current link on success, or a negative value on + failure. + For seekable streams, this is a number between 0 (inclusive) and the + value returned by op_link_count() (exclusive). + For unseekable streams, this value starts at 0 and increments by one + each time a new link is encountered (even though op_link_count() + always returns 1). + \retval #OP_EINVAL The stream was only partially open.*/ +int op_current_link(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Computes the bitrate of the stream, or of an individual link in a + (possibly-chained) Ogg Opus stream. + The stream must be seekable to compute the bitrate. + For unseekable streams, use op_bitrate_instant() to get periodic estimates. + \warning If the Opus stream (or link) is concurrently multiplexed with other + logical streams (e.g., video), this uses the size of the entire stream (or + link) to compute the bitrate, not just the number of bytes in the first + logical Opus stream. + Returning the latter requires scanning the entire file, but this may be done + by decoding the whole file and calling op_bitrate_instant() once at the + end. + Install a trivial decoding callback with op_set_decode_callback() if you + wish to skip actual decoding during this process. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \param _li The index of the link whose bitrate should be computed. + Use a negative number to get the bitrate of the whole stream. + \return The bitrate on success, or a negative value on error. + \retval #OP_EINVAL The stream was only partially open, the stream was not + seekable, or \a _li was larger than the number of + links.*/ +opus_int32 op_bitrate(const OggOpusFile *_of,int _li) OP_ARG_NONNULL(1); + +/**Compute the instantaneous bitrate, measured as the ratio of bits to playable + samples decoded since a) the last call to op_bitrate_instant(), b) the last + seek, or c) the start of playback, whichever was most recent. + This will spike somewhat after a seek or at the start/end of a chain + boundary, as pre-skip, pre-roll, and end-trimming causes samples to be + decoded but not played. + \param _of The \c OggOpusFile from which to retrieve the bitrate. + \return The bitrate, in bits per second, or a negative value on error. + \retval #OP_FALSE No data has been decoded since any of the events + described above. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int32 op_bitrate_instant(OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the current value of the position indicator for \a _of. + \param _of The \c OggOpusFile from which to retrieve the position indicator. + \return The byte position that is currently being read from. + \retval #OP_EINVAL The stream was only partially open.*/ +opus_int64 op_raw_tell(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/**Obtain the PCM offset of the next sample to be read. + If the stream is not properly timestamped, this might not increment by the + proper amount between reads, or even return monotonically increasing + values. + \param _of The \c OggOpusFile from which to retrieve the PCM offset. + \return The PCM offset of the next sample to be read. + \retval #OP_EINVAL The stream was only partially open.*/ +ogg_int64_t op_pcm_tell(const OggOpusFile *_of) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_seeking Seeking*/ +/*@{*/ +/**\name Functions for seeking in Opus streams + + These functions let you seek in Opus streams, if the underlying stream + support it. + Seeking is implemented for all built-in stream I/O routines, though some + individual streams may not be seekable (pipes, live HTTP streams, or HTTP + streams from a server that does not support Range requests). + + op_raw_seek() is the fastest: it is guaranteed to perform at most one + physical seek, but, since the target is a byte position, makes no guarantee + how close to a given time it will come. + op_pcm_seek() provides sample-accurate seeking. + The number of physical seeks it requires is still quite small (often 1 or + 2, even in highly variable bitrate streams). + + Seeking in Opus requires decoding some pre-roll amount before playback to + allow the internal state to converge (as if recovering from packet loss). + This is handled internally by libopusfile, but means there is + little extra overhead for decoding up to the exact position requested + (since it must decode some amount of audio anyway). + It also means that decoding after seeking may not return exactly the same + values as would be obtained by decoding the stream straight through. + However, such differences are expected to be smaller than the loss + introduced by Opus's lossy compression.*/ +/*@{*/ + +/**Seek to a byte offset relative to the compressed data. + This also scans packets to update the PCM cursor. + It will cross a logical bitstream boundary, but only if it can't get any + packets out of the tail of the link to which it seeks. + \param _of The \c OggOpusFile in which to seek. + \param _byte_offset The byte position to seek to. + This must be between 0 and #op_raw_total(\a _of,\c -1) + (inclusive). + \return 0 on success, or a negative error code on failure. + \retval #OP_EREAD The underlying seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK Failed to initialize a decoder for a stream for an + unknown reason.*/ +int op_raw_seek(OggOpusFile *_of,opus_int64 _byte_offset) OP_ARG_NONNULL(1); + +/**Seek to the specified PCM offset, such that decoding will begin at exactly + the requested position. + \param _of The \c OggOpusFile in which to seek. + \param _pcm_offset The PCM offset to seek to. + This is in samples at 48 kHz relative to the start of the + stream. + \return 0 on success, or a negative value on error. + \retval #OP_EREAD An underlying read or seek operation failed. + \retval #OP_EINVAL The stream was only partially open, or the target was + outside the valid range for the stream. + \retval #OP_ENOSEEK This stream is not seekable. + \retval #OP_EBADLINK We failed to find data we had seen before, or the + bitstream structure was sufficiently malformed that + seeking to the target destination was impossible.*/ +int op_pcm_seek(OggOpusFile *_of,ogg_int64_t _pcm_offset) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +/**\defgroup stream_decoding Decoding*/ +/*@{*/ +/**\name Functions for decoding audio data + + These functions retrieve actual decoded audio data from the stream. + The general functions, op_read() and op_read_float() return 16-bit or + floating-point output, both using native endian ordering. + The number of channels returned can change from link to link in a chained + stream. + There are special functions, op_read_stereo() and op_read_float_stereo(), + which always output two channels, to simplify applications which do not + wish to handle multichannel audio. + These downmix multichannel files to two channels, so they can always return + samples in the same format for every link in a chained file. + + If the rest of your audio processing chain can handle floating point, the + floating-point routines should be preferred, as they prevent clipping and + other issues which might be avoided entirely if, e.g., you scale down the + volume at some other stage. + However, if you intend to consume 16-bit samples directly, the conversion in + libopusfile provides noise-shaping dithering and, if compiled + against libopus 1.1 or later, soft-clipping prevention. + + libopusfile can also be configured at compile time to use the + fixed-point libopus API. + If so, libopusfile's floating-point API may also be disabled. + In that configuration, nothing in libopusfile will use any + floating-point operations, to simplify support on devices without an + adequate FPU. + + \warning HTTPS streams may be be vulnerable to truncation attacks if you do + not check the error return code from op_read_float() or its associated + functions. + If the remote peer does not close the connection gracefully (with a TLS + "close notify" message), these functions will return #OP_EREAD instead of 0 + when they reach the end of the file. + If you are reading from an URL (particularly if seeking is not + supported), you should make sure to check for this error and warn the user + appropriately.*/ +/*@{*/ + +/**Indicates that the decoding callback should produce signed 16-bit + native-endian output samples.*/ +#define OP_DEC_FORMAT_SHORT (7008) +/**Indicates that the decoding callback should produce 32-bit native-endian + float samples.*/ +#define OP_DEC_FORMAT_FLOAT (7040) + +/**Indicates that the decoding callback did not decode anything, and that + libopusfile should decode normally instead.*/ +#define OP_DEC_USE_DEFAULT (6720) + +/**Called to decode an Opus packet. + This should invoke the functional equivalent of opus_multistream_decode() or + opus_multistream_decode_float(), except that it returns 0 on success + instead of the number of decoded samples (which is known a priori). + \param _ctx The application-provided callback context. + \param _decoder The decoder to use to decode the packet. + \param[out] _pcm The buffer to decode into. + This will always have enough room for \a _nchannels of + \a _nsamples samples, which should be placed into this + buffer interleaved. + \param _op The packet to decode. + This will always have its granule position set to a valid + value. + \param _nsamples The number of samples expected from the packet. + \param _nchannels The number of channels expected from the packet. + \param _format The desired sample output format. + This is either #OP_DEC_FORMAT_SHORT or + #OP_DEC_FORMAT_FLOAT. + \param _li The index of the link from which this packet was decoded. + \return A non-negative value on success, or a negative value on error. + Any error codes should be the same as those returned by + opus_multistream_decode() or opus_multistream_decode_float(). + Success codes are as follows: + \retval 0 Decoding was successful. + The application has filled the buffer with + exactly \a _nsamples*\a + _nchannels samples in the requested + format. + \retval #OP_DEC_USE_DEFAULT No decoding was done. + libopusfile should do the decoding + by itself instead.*/ +typedef int (*op_decode_cb_func)(void *_ctx,OpusMSDecoder *_decoder,void *_pcm, + const ogg_packet *_op,int _nsamples,int _nchannels,int _format,int _li); + +/**Sets the packet decode callback function. + If set, this is called once for each packet that needs to be decoded. + This can be used by advanced applications to do additional processing on the + compressed or uncompressed data. + For example, an application might save the final entropy coder state for + debugging and testing purposes, or it might apply additional filters + before the downmixing, dithering, or soft-clipping performed by + libopusfile, so long as these filters do not introduce any + latency. + + A call to this function is no guarantee that the audio will eventually be + delivered to the application. + libopusfile may discard some or all of the decoded audio data + (i.e., at the beginning or end of a link, or after a seek), however the + callback is still required to provide all of it. + \param _of The \c OggOpusFile on which to set the decode callback. + \param _decode_cb The callback function to call. + This may be NULL to disable calling the + callback. + \param _ctx The application-provided context pointer to pass to the + callback on each call.*/ +void op_set_decode_callback(OggOpusFile *_of, + op_decode_cb_func _decode_cb,void *_ctx) OP_ARG_NONNULL(1); + +/**Gain offset type that indicates that the provided offset is relative to the + header gain. + This is the default.*/ +#define OP_HEADER_GAIN (0) + +/**Gain offset type that indicates that the provided offset is relative to the + R128_ALBUM_GAIN value (if any), in addition to the header gain.*/ +#define OP_ALBUM_GAIN (3007) + +/**Gain offset type that indicates that the provided offset is relative to the + R128_TRACK_GAIN value (if any), in addition to the header gain.*/ +#define OP_TRACK_GAIN (3008) + +/**Gain offset type that indicates that the provided offset should be used as + the gain directly, without applying any the header or track gains.*/ +#define OP_ABSOLUTE_GAIN (3009) + +/**Sets the gain to be used for decoded output. + By default, the gain in the header is applied with no additional offset. + The total gain (including header gain and/or track gain, if applicable, and + this offset), will be clamped to [-32768,32767]/256 dB. + This is more than enough to saturate or underflow 16-bit PCM. + \note The new gain will not be applied to any already buffered, decoded + output. + This means you cannot change it sample-by-sample, as at best it will be + updated packet-by-packet. + It is meant for setting a target volume level, rather than applying smooth + fades, etc. + \param _of The \c OggOpusFile on which to set the gain offset. + \param _gain_type One of #OP_HEADER_GAIN, #OP_ALBUM_GAIN, + #OP_TRACK_GAIN, or #OP_ABSOLUTE_GAIN. + \param _gain_offset_q8 The gain offset to apply, in 1/256ths of a dB. + \return 0 on success or a negative value on error. + \retval #OP_EINVAL The \a _gain_type was unrecognized.*/ +int op_set_gain_offset(OggOpusFile *_of, + int _gain_type,opus_int32 _gain_offset_q8) OP_ARG_NONNULL(1); + +/**Sets whether or not dithering is enabled for 16-bit decoding. + By default, when libopusfile is compiled to use floating-point + internally, calling op_read() or op_read_stereo() will first decode to + float, and then convert to fixed-point using noise-shaping dithering. + This flag can be used to disable that dithering. + When the application uses op_read_float() or op_read_float_stereo(), or when + the library has been compiled to decode directly to fixed point, this flag + has no effect. + \param _of The \c OggOpusFile on which to enable or disable dithering. + \param _enabled A non-zero value to enable dithering, or 0 to disable it.*/ +void op_set_dither_enabled(OggOpusFile *_of,int _enabled) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + per channel. + This is done because +
    +
  1. The channel count cannot be known a priori (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,
    2. +
  2. Returning the samples per channel matches the libopus API + as closely as we're able,
    3. +
  3. Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and
    4. +
  4. This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).
    5. +
+ \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values at 48 kHz + with a nominal range of [-32768,32767). + Multiple channels are interleaved using the + Vorbis + channel ordering. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + values per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + libopusfile may return less data than + requested. + If so, there is no guarantee that the remaining data + in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass NULL if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + op_head(_of,*_li). + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream. + \note Although \a _buf_size must indicate the total number of values that + can be stored in \a _pcm, the return value is the number of samples + per channel. +
    +
  1. The channel count cannot be known a priori (reading more samples might + advance us into the next link, with a different channel count), so + \a _buf_size cannot also be in units of samples per channel,
    2. +
  2. Returning the samples per channel matches the libopus API + as closely as we're able,
    3. +
  3. Returning the total number of values instead of samples per channel + would mean the caller would need a division to compute the samples per + channel, and might worry about the possibility of getting back samples + for some channels and not others, and
    4. +
  4. This approach is relatively fool-proof: if an application passes too + small a value to \a _buf_size, they will simply get fewer samples back, + and if they assume the return value is the total number of values, then + they will simply read too few (rather than reading too many and going + off the end of the buffer).
    5. +
+ \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples as + signed floats at 48 kHz with a nominal range of + [-1.0,1.0]. + Multiple channels are interleaved using the + Vorbis + channel ordering. + This must have room for at least \a _buf_size floats. + \param _buf_size The number of floats that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (5760 + samples per channel). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \param[out] _li The index of the link this data was decoded from. + You may pass NULL if you do not need this + information. + If this function fails (returning a negative value), + this parameter is left unset. + \return The number of samples read per channel on success, or a negative + value on failure. + The channel count can be retrieved on success by calling + op_head(_of,*_li). + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for all channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float(OggOpusFile *_of, + float *_pcm,int _buf_size,int *_li) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples per + channel, even though the channel count is known, for consistency with + op_read(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed native-endian 16-bit values at 48 kHz + with a nominal range of [-32768,32767). + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_stereo(OggOpusFile *_of, + opus_int16 *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/**Reads more samples from the stream and downmixes to stereo, if necessary. + This function is intended for simple players that want a uniform output + format, even if the channel count changes between links in a chained + stream. + \note \a _buf_size indicates the total number of values that can be stored + in \a _pcm, while the return value is the number of samples per + channel, even though the channel count is known, for consistency with + op_read_float(). + \param _of The \c OggOpusFile from which to read. + \param[out] _pcm A buffer in which to store the output PCM samples, as + signed floats at 48 kHz with a nominal range of + [-1.0,1.0]. + The left and right channels are interleaved in the + buffer. + This must have room for at least \a _buf_size values. + \param _buf_size The number of values that can be stored in \a _pcm. + It is recommended that this be large enough for at + least 120 ms of data at 48 kHz per channel (11520 + values total). + Smaller buffers will simply return less data, possibly + consuming more memory to buffer the data internally. + If less than \a _buf_size values are returned, + libopusfile makes no guarantee that the + remaining data in \a _pcm will be unmodified. + \return The number of samples read per channel on success, or a negative + value on failure. + The number of samples returned may be 0 if the buffer was too small + to store even a single sample for both channels, or if end-of-file + was reached. + The list of possible failure codes follows. + Most of them can only be returned by unseekable, chained streams + that encounter a new link. + \retval #OP_HOLE There was a hole in the data, and some samples + may have been skipped. + Call this function again to continue decoding + past the hole. + \retval #OP_EREAD An underlying read operation failed. + This may signal a truncation attack from an + source. + \retval #OP_EFAULT An internal memory allocation failed. + \retval #OP_EIMPL An unseekable stream encountered a new link that + used a feature that is not implemented, such as + an unsupported channel family. + \retval #OP_EINVAL The stream was only partially open. + \retval #OP_ENOTFORMAT An unseekable stream encountered a new link that + that did not have any logical Opus streams in it. + \retval #OP_EBADHEADER An unseekable stream encountered a new link with a + required header packet that was not properly + formatted, contained illegal values, or was + missing altogether. + \retval #OP_EVERSION An unseekable stream encountered a new link with + an ID header that contained an unrecognized + version number. + \retval #OP_EBADPACKET Failed to properly decode the next packet. + \retval #OP_EBADLINK We failed to find data we had seen before. + \retval #OP_EBADTIMESTAMP An unseekable stream encountered a new link with + a starting timestamp that failed basic validity + checks.*/ +OP_WARN_UNUSED_RESULT int op_read_float_stereo(OggOpusFile *_of, + float *_pcm,int _buf_size) OP_ARG_NONNULL(1); + +/*@}*/ +/*@}*/ + +# if OP_GNUC_PREREQ(4,0) +# pragma GCC visibility pop +# endif + +# if defined(__cplusplus) +} +# endif + +#endif diff --git a/engine/voice_codecs/opus/voiceencoder_opus.cpp b/engine/voice_codecs/opus/voiceencoder_opus.cpp new file mode 100644 index 00000000..203539bc --- /dev/null +++ b/engine/voice_codecs/opus/voiceencoder_opus.cpp @@ -0,0 +1,184 @@ +//========= Copyright Valve Corporation, All rights reserved. ============// +// +// Purpose: +// +// $NoKeywords: $ +// +//=============================================================================// +#include "ivoicecodec.h" +#include "iframeencoder.h" + +#include +#include +#include + + +// NOTE: This has to be the last file included! +#include "tier0/memdbgon.h" +#include "tier0/dbg.h" + +#define CHANNELS 1 + +struct opus_options +{ + int iSampleRate; + int iRawFrameSize; + int iPacketSize; +}; + +opus_options g_OpusOpts[] = +{ + {44100, 256, 120}, + {22050, 120, 60}, + {22050, 256, 60}, + {22050, 512, 64}, +}; + +class VoiceEncoder_Opus : public IFrameEncoder +{ +public: + VoiceEncoder_Opus(); + virtual ~VoiceEncoder_Opus(); + + // Interfaces IFrameDecoder + + bool Init(int quality, int &rawFrameSize, int &encodedFrameSize); + void Release(); + void DecodeFrame(const char *pCompressed, char *pDecompressedBytes); + void EncodeFrame(const char *pUncompressedBytes, char *pCompressed); + bool ResetState(); + +private: + + bool InitStates(); + void TermStates(); + + OpusCustomEncoder *m_EncoderState; // Celt internal encoder state + OpusCustomDecoder *m_DecoderState; // Celt internal decoder state + OpusCustomMode *m_Mode; + + int m_iVersion; +}; + +extern IVoiceCodec* CreateVoiceCodec_Frame(IFrameEncoder *pEncoder); + +void* CreateCeltVoiceCodec() +{ + IFrameEncoder *pEncoder = new VoiceEncoder_Opus; + return CreateVoiceCodec_Frame( pEncoder ); +} + +EXPOSE_INTERFACE_FN(CreateCeltVoiceCodec, IVoiceCodec, "vaudio_opus") + +////////////////////////////////////////////////////////////////////// +// Construction/Destruction +////////////////////////////////////////////////////////////////////// + +VoiceEncoder_Opus::VoiceEncoder_Opus() +{ + m_EncoderState = NULL; + m_DecoderState = NULL; + m_Mode = NULL; + m_iVersion = 0; +} + +VoiceEncoder_Opus::~VoiceEncoder_Opus() +{ + TermStates(); +} + +bool VoiceEncoder_Opus::Init( int quality, int &rawFrameSize, int &encodedFrameSize) +{ + m_iVersion = quality; + + rawFrameSize = g_OpusOpts[m_iVersion].iRawFrameSize * BYTES_PER_SAMPLE; + + int iError = 0; + + m_Mode = opus_custom_mode_create( g_OpusOpts[m_iVersion].iSampleRate, g_OpusOpts[m_iVersion].iRawFrameSize, &iError ); + + m_EncoderState = opus_custom_encoder_create( m_Mode, CHANNELS, NULL); + m_DecoderState = opus_custom_decoder_create( m_Mode, CHANNELS, NULL); + + if ( !InitStates() ) + return false; + + encodedFrameSize = g_OpusOpts[m_iVersion].iPacketSize; + + return true; +} + +void VoiceEncoder_Opus::Release() +{ + delete this; +} + +void VoiceEncoder_Opus::EncodeFrame(const char *pUncompressedBytes, char *pCompressed) +{ + unsigned char output[1024]; + + opus_custom_encode( m_EncoderState, (opus_int16*)pUncompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize, output, g_OpusOpts[m_iVersion].iPacketSize ); + + for ( int i = 0; i < g_OpusOpts[m_iVersion].iPacketSize; i++ ) + { + *pCompressed = (char)output[i]; + pCompressed++; + } +} + +void VoiceEncoder_Opus::DecodeFrame(const char *pCompressed, char *pDecompressedBytes) +{ + unsigned char output[1024]; + char *out = (char *)pCompressed; + + if ( !pCompressed ) + { + int decoded = opus_custom_decode( m_DecoderState, NULL, g_OpusOpts[m_iVersion].iPacketSize, (opus_int16 *)pDecompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize ); + return; + } + + for ( int i = 0; i < g_OpusOpts[m_iVersion].iPacketSize; i++ ) + { + output[i] = ( unsigned char ) ( ( *out < 0 ) ? (*out + 256) : *out ); + out++; + } + + //celt_decoder_ctl( m_DecoderState, CELT_RESET_STATE_REQUEST, NULL ); + int decoded = opus_custom_decode( m_DecoderState, output, g_OpusOpts[m_iVersion].iPacketSize, (opus_int16 *)pDecompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize ); +} + +bool VoiceEncoder_Opus::ResetState() +{ + opus_custom_encoder_ctl(m_EncoderState, OPUS_RESET_STATE ); + opus_custom_decoder_ctl(m_DecoderState, OPUS_RESET_STATE ); + + return true; +} + +bool VoiceEncoder_Opus::InitStates() +{ + if ( !m_EncoderState || !m_DecoderState ) + return false; + + opus_custom_encoder_ctl(m_EncoderState, OPUS_RESET_STATE ); + opus_custom_decoder_ctl(m_DecoderState, OPUS_RESET_STATE ); + + return true; +} + +void VoiceEncoder_Opus::TermStates() +{ + if( m_EncoderState ) + { + opus_custom_encoder_destroy( m_EncoderState ); + m_EncoderState = NULL; + } + + if( m_DecoderState ) + { + opus_custom_decoder_destroy( m_DecoderState ); + m_DecoderState = NULL; + } + + opus_custom_mode_destroy( m_Mode ); +} diff --git a/engine/voice_codecs/opus/wscript b/engine/voice_codecs/opus/wscript new file mode 100755 index 00000000..ce9207d3 --- /dev/null +++ b/engine/voice_codecs/opus/wscript @@ -0,0 +1,53 @@ +#! /usr/bin/env python +# encoding: utf-8 + +from waflib import Utils +import os + +top = '.' +PROJECT_NAME = 'vaudio_opus' + +def options(opt): + return + +def configure(conf): + conf.define('OPUS_EXPORTS',1) + +def build(bld): + source = [ + 'voiceencoder_opus.cpp', + '../frame_encoder/voice_codec_frame.cpp', + '../../../tier1/interface.cpp' + ] + + includes = [ + '.', + '../frame_encoder', + '../../../public', + '../../../public/tier1', + '../../../public/tier0', + '../../', + '../../../common', + '../../audio/public', + 'celt' + ] + + defines = [] + + libs = ['tier0','tier1','vstdlib','OPUS'] + + install_path = bld.env.LIBDIR + + bld.shlib( + source = source, + target = PROJECT_NAME, + name = PROJECT_NAME, + features = 'c cxx', + includes = includes, + defines = defines, + use = libs, + install_path = install_path, + subsystem = bld.env.MSVC_SUBSYSTEM, + idx = bld.get_taskgen_count() + ) + diff --git a/engine/voice_codecs/speex/VoiceEncoder_Speex.h b/engine/voice_codecs/speex/VoiceEncoder_Speex.h index c3672427..1bd83fa7 100644 --- a/engine/voice_codecs/speex/VoiceEncoder_Speex.h +++ b/engine/voice_codecs/speex/VoiceEncoder_Speex.h @@ -45,7 +45,7 @@ damage. */ #include "iframeencoder.h" -#include +#include class VoiceEncoder_Speex : public IFrameEncoder { diff --git a/engine/voice_codecs/speex/source/AUTHORS b/engine/voice_codecs/speex/source/AUTHORS deleted file mode 100644 index b3fd7f53..00000000 --- a/engine/voice_codecs/speex/source/AUTHORS +++ /dev/null @@ -1,15 +0,0 @@ -Jean-Marc Valin - All the code except the following - -David Rowe via VoiceTronix - lsp.c lsp.h - Also ideas and feedback - -John Francis Edwards: - wave_out.[ch], some #ifdefs for windows port and MSVC project files - -Atsuhiko Yamanaka : - Patch to speexenc.c to add Vorbis comment format - -Radim Kolar : - Patch to speexenc.c for supporting more input formats diff --git a/engine/voice_codecs/speex/source/COPYING b/engine/voice_codecs/speex/source/COPYING deleted file mode 100644 index 37beceff..00000000 --- a/engine/voice_codecs/speex/source/COPYING +++ /dev/null @@ -1,26 +0,0 @@ -Redistribution and use in source and binary forms, with or without -modification, are permitted provided that the following conditions -are met: - -- Redistributions of source code must retain the above copyright -notice, this list of conditions and the following disclaimer. - -- Redistributions in binary form must reproduce the above copyright -notice, this list of conditions and the following disclaimer in the -documentation and/or other materials provided with the distribution. - -- Neither the name of the Xiph.org Foundation nor the names of its -contributors may be used to endorse or promote products derived from -this software without specific prior written permission. - -THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS -``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT -LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR -A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR -CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, -EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, -PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR -PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF -LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING -NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS -SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. diff --git a/engine/voice_codecs/speex/source/ChangeLog b/engine/voice_codecs/speex/source/ChangeLog deleted file mode 100644 index 10db2987..00000000 --- a/engine/voice_codecs/speex/source/ChangeLog +++ /dev/null @@ -1,6 +0,0 @@ -2002/03/27 Jean-Marc Valin: -Working encoder and decoder for both narrowband and wideband. - -2002/02/27 Jean-Marc Valin: -Got the basic encoder working as a demo with quantization only on some -parameters. diff --git a/engine/voice_codecs/speex/source/INSTALL b/engine/voice_codecs/speex/source/INSTALL deleted file mode 100644 index 15f2e5f4..00000000 --- a/engine/voice_codecs/speex/source/INSTALL +++ /dev/null @@ -1,8 +0,0 @@ -Installing Speex is as easy as: - -% ./configure [--prefix=] -% make -% make install - -Note that if you are using the code from CVS, you will need to run "autogen.sh" -instead of "configure". diff --git a/engine/voice_codecs/speex/source/Makefile.am b/engine/voice_codecs/speex/source/Makefile.am deleted file mode 100644 index 113f18da..00000000 --- a/engine/voice_codecs/speex/source/Makefile.am +++ /dev/null @@ -1,11 +0,0 @@ -## Process this file with automake to produce Makefile.in. -*-Makefile-*- - -# Disable automatic dependency tracking if using other tools than gcc and gmake -#AUTOMAKE_OPTIONS = no-dependencies - -EXTRA_DIST = Speex.spec Speex.spec.in - -SUBDIRS = libspeex @src@ doc win32 - -rpm: dist - rpmbuild -ta ${PACKAGE}-${VERSION}.tar.gz diff --git a/engine/voice_codecs/speex/source/Makefile.in b/engine/voice_codecs/speex/source/Makefile.in deleted file mode 100644 index 83e20fa6..00000000 --- a/engine/voice_codecs/speex/source/Makefile.in +++ /dev/null @@ -1,347 +0,0 @@ -# Makefile.in generated automatically by automake 1.4-p6 from Makefile.am - -# Copyright (C) 1994, 1995-8, 1999, 2001 Free Software Foundation, Inc. -# This Makefile.in is free software; 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Unlike other codecs like MP3 and Ogg Vorbis, Speex is designed to compress voice at bitrates in the 2-45 kbps range. Possible applications include VoIP, internet audio streaming, archiving of speech data (e.g. voice mail), and audio books. In some sense, it is meant to be complementary to the Ogg Vorbis codec. - -To use the Speex command line tools: - -% speexenc [options] input_file.wav compressed_file.spx - -% speexdec [options] compressed_file.spx output_file.wav diff --git a/engine/voice_codecs/speex/source/Speex.spec b/engine/voice_codecs/speex/source/Speex.spec deleted file mode 100644 index ffc94c1b..00000000 --- a/engine/voice_codecs/speex/source/Speex.spec +++ /dev/null @@ -1,68 +0,0 @@ -%define name speex -%define ver 1.0.1 -%define rel 1 - -Summary: An open-source, patent-free speech codec -Name: %name -Version: %ver -Release: %rel -Copyright: BSD -Group: Application/Devel -Source: http://www.speex.org/download/%{name}-%{ver}.tar.gz -URL: http://www.speex.org/ -Vendor: Speex -Packager: Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) -BuildRoot: /var/tmp/%{name}-build-root -Docdir: /usr/share/doc - -%description -Speex is a patent-free audio codec designed especially for voice (unlike -Vorbis which targets general audio) signals and providing good narrowband -and wideband quality. This project aims to be complementary to the Vorbis -codec. - -%package devel -Summary: Speex development files -Group: Development/Libraries -Requires: %{name} = %{version} - -%description devel -Speex development files. - -%changelog -* Thu Oct 03 2002 Jean-Marc Valin -- Added devel package inspired from PLD spec file - -* Tue Jul 30 2002 Fredrik Rambris 0.5.2 -- Added buildroot and docdir and ldconfig. Makes it builadble by non-roots - and also doesn't write to actual library paths when building. - -%prep -%setup - -%build -export CFLAGS='-O3 -DRELEASE' -./configure --prefix=/usr --enable-shared --enable-static -make - -%install -rm -rf $RPM_BUILD_ROOT -make DESTDIR=$RPM_BUILD_ROOT install - -%post -p /sbin/ldconfig -%postun -p /sbin/ldconfig - -%files -%defattr(644,root,root,755) -%doc COPYING AUTHORS ChangeLog NEWS README -%doc doc/manual.pdf -/usr/share/man/man1/speexenc.1* -/usr/share/man/man1/speexdec.1* -%attr(755,root,root) %{_bindir}/speex* -%attr(755,root,root) %{_libdir}/libspeex*.so* - -%files devel -%defattr(644,root,root,755) -%attr(755,root,root) %{_libdir}/libspeex*.la -%{_includedir}/speex*.h -%{_libdir}/libspeex*.a diff --git a/engine/voice_codecs/speex/source/Speex.spec.in b/engine/voice_codecs/speex/source/Speex.spec.in deleted file mode 100644 index 9c64d8af..00000000 --- a/engine/voice_codecs/speex/source/Speex.spec.in +++ /dev/null @@ -1,68 +0,0 @@ -%define name @PACKAGE@ -%define ver @VERSION@ -%define rel 1 - -Summary: An open-source, patent-free speech codec -Name: %name -Version: %ver -Release: %rel -Copyright: BSD -Group: Application/Devel -Source: http://www.speex.org/download/%{name}-%{ver}.tar.gz -URL: http://www.speex.org/ -Vendor: Speex -Packager: Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) -BuildRoot: /var/tmp/%{name}-build-root -Docdir: /usr/share/doc - -%description -Speex is a patent-free audio codec designed especially for voice (unlike -Vorbis which targets general audio) signals and providing good narrowband -and wideband quality. This project aims to be complementary to the Vorbis -codec. - -%package devel -Summary: Speex development files -Group: Development/Libraries -Requires: %{name} = %{version} - -%description devel -Speex development files. - -%changelog -* Thu Oct 03 2002 Jean-Marc Valin -- Added devel package inspired from PLD spec file - -* Tue Jul 30 2002 Fredrik Rambris 0.5.2 -- Added buildroot and docdir and ldconfig. Makes it builadble by non-roots - and also doesn't write to actual library paths when building. - -%prep -%setup - -%build -export CFLAGS='-O3 -DRELEASE' -./configure --prefix=/usr --enable-shared --enable-static -make - -%install -rm -rf $RPM_BUILD_ROOT -make DESTDIR=$RPM_BUILD_ROOT install - -%post -p /sbin/ldconfig -%postun -p /sbin/ldconfig - -%files -%defattr(644,root,root,755) -%doc COPYING AUTHORS ChangeLog NEWS README -%doc doc/manual.pdf -/usr/share/man/man1/speexenc.1* -/usr/share/man/man1/speexdec.1* -%attr(755,root,root) %{_bindir}/speex* -%attr(755,root,root) %{_libdir}/libspeex*.so* - -%files devel -%defattr(644,root,root,755) -%attr(755,root,root) %{_libdir}/libspeex*.la -%{_includedir}/speex*.h -%{_libdir}/libspeex*.a diff --git a/engine/voice_codecs/speex/source/TODO b/engine/voice_codecs/speex/source/TODO deleted file mode 100644 index 12a1a535..00000000 --- a/engine/voice_codecs/speex/source/TODO +++ /dev/null @@ -1,18 +0,0 @@ -Features --Add maximum/minimum bit-rate control for VBR --Get the encoder to use the rate of packet loss (more conservative pitch gains) - -Long-term quality improvements --Improve perceptual enhancement (including wideband) - -Standards -*Complete Speex RTP profile --MIME type registration --MS ACM wrapper - -*required for 1.0 - -ideas: -Peelable stream (double codebook, higher bands, stereo) -LPC from spectral domain -Better psycho-acoustics? Masking curve from Vorbis diff --git a/engine/voice_codecs/speex/source/acinclude.m4 b/engine/voice_codecs/speex/source/acinclude.m4 deleted file mode 100644 index 8efb6599..00000000 --- a/engine/voice_codecs/speex/source/acinclude.m4 +++ /dev/null @@ -1,84 +0,0 @@ -AC_DEFUN(AC_FIND_FILE, -[ -$3=NONE -for i in $2; -do - for j in $1; - do - if test -r "$i/$j"; then - $3=$i - break 2 - fi - done -done -]) - -AC_DEFUN(AC_PATH_LIBOGG, -[ -OGG_LIBS="-logg" - -AC_MSG_CHECKING([for libogg]) - -ac_ogg_includes=NONE ac_ogg_libraries=NONE ac_ogg_bindir=NONE -ogg_libraries="" -ogg_includes="" -AC_ARG_WITH(ogg-dir, - [ --with-ogg-dir=DIR where the root of OGG is installed ], - [ ac_ogg_includes="$withval"/include - ac_ogg_libraries="$withval"/lib - ]) - -AC_ARG_WITH(ogg-includes, - [ --with-ogg-includes=DIR where the OGG includes are. ], - [ - ac_ogg_includes="$withval" - ]) - -ogg_libs_given=no - -AC_ARG_WITH(ogg-libraries, - [ --with-ogg-libraries=DIR where the OGG library is installed.], - [ ac_ogg_libraries="$withval" - ogg_libs_given=yes - ]) - -ogg_incdirs="/usr/include /usr/lib/ogg/include /opt/include /usr/local/ogg/include /usr/include/ogg /usr/include /usr/local/include" -if test ! 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- 9000/[34678]??:HP-UX:*:*) - HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` - case "${UNAME_MACHINE}" in - 9000/31? ) HP_ARCH=m68000 ;; - 9000/[34]?? ) HP_ARCH=m68k ;; - 9000/[678][0-9][0-9]) - if [ -x /usr/bin/getconf ]; then - sc_cpu_version=`/usr/bin/getconf SC_CPU_VERSION 2>/dev/null` - sc_kernel_bits=`/usr/bin/getconf SC_KERNEL_BITS 2>/dev/null` - case "${sc_cpu_version}" in - 523) HP_ARCH="hppa1.0" ;; # CPU_PA_RISC1_0 - 528) HP_ARCH="hppa1.1" ;; # CPU_PA_RISC1_1 - 532) # CPU_PA_RISC2_0 - case "${sc_kernel_bits}" in - 32) HP_ARCH="hppa2.0n" ;; - 64) HP_ARCH="hppa2.0w" ;; - '') HP_ARCH="hppa2.0" ;; # HP-UX 10.20 - esac ;; - esac - fi - if [ "${HP_ARCH}" = "" ]; then - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - - #define _HPUX_SOURCE - #include - #include - - int main () - { - #if defined(_SC_KERNEL_BITS) - long bits = sysconf(_SC_KERNEL_BITS); - #endif - long cpu = sysconf (_SC_CPU_VERSION); - - switch (cpu) - { - case CPU_PA_RISC1_0: puts ("hppa1.0"); break; - case CPU_PA_RISC1_1: puts ("hppa1.1"); break; - case CPU_PA_RISC2_0: - #if defined(_SC_KERNEL_BITS) - switch (bits) - { - case 64: puts ("hppa2.0w"); break; - case 32: puts ("hppa2.0n"); break; - default: puts ("hppa2.0"); break; - } break; - #else /* !defined(_SC_KERNEL_BITS) */ - puts ("hppa2.0"); break; - #endif - default: puts ("hppa1.0"); break; - } - exit (0); - } -EOF - (CCOPTS= $CC_FOR_BUILD -o $dummy $dummy.c 2>/dev/null) && HP_ARCH=`$dummy` - if test -z "$HP_ARCH"; then HP_ARCH=hppa; fi - rm -f $dummy.c $dummy && rmdir $tmpdir - fi ;; - esac - echo ${HP_ARCH}-hp-hpux${HPUX_REV} - exit 0 ;; - ia64:HP-UX:*:*) - HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` - echo ia64-hp-hpux${HPUX_REV} - exit 0 ;; - 3050*:HI-UX:*:*) - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - #include - int - main () - { - long cpu = sysconf (_SC_CPU_VERSION); - /* The order matters, because CPU_IS_HP_MC68K erroneously returns - true for CPU_PA_RISC1_0. CPU_IS_PA_RISC returns correct - results, however. */ - if (CPU_IS_PA_RISC (cpu)) - { - switch (cpu) - { - case CPU_PA_RISC1_0: puts ("hppa1.0-hitachi-hiuxwe2"); break; - case CPU_PA_RISC1_1: puts ("hppa1.1-hitachi-hiuxwe2"); break; - case CPU_PA_RISC2_0: puts ("hppa2.0-hitachi-hiuxwe2"); break; - default: puts ("hppa-hitachi-hiuxwe2"); break; - } - } - else if (CPU_IS_HP_MC68K (cpu)) - puts ("m68k-hitachi-hiuxwe2"); - else puts ("unknown-hitachi-hiuxwe2"); - exit (0); - } -EOF - $CC_FOR_BUILD -o $dummy $dummy.c && $dummy && rm -f $dummy.c $dummy && rmdir $tmpdir && exit 0 - rm -f $dummy.c $dummy && rmdir $tmpdir - echo unknown-hitachi-hiuxwe2 - exit 0 ;; - 9000/7??:4.3bsd:*:* | 9000/8?[79]:4.3bsd:*:* ) - echo hppa1.1-hp-bsd - exit 0 ;; - 9000/8??:4.3bsd:*:*) - echo hppa1.0-hp-bsd - exit 0 ;; - *9??*:MPE/iX:*:* | *3000*:MPE/iX:*:*) - echo hppa1.0-hp-mpeix - exit 0 ;; - hp7??:OSF1:*:* | hp8?[79]:OSF1:*:* ) - echo hppa1.1-hp-osf - exit 0 ;; - hp8??:OSF1:*:*) - echo hppa1.0-hp-osf - exit 0 ;; - i*86:OSF1:*:*) - if [ -x /usr/sbin/sysversion ] ; then - echo ${UNAME_MACHINE}-unknown-osf1mk - else - echo ${UNAME_MACHINE}-unknown-osf1 - fi - exit 0 ;; - parisc*:Lites*:*:*) - echo hppa1.1-hp-lites - exit 0 ;; - C1*:ConvexOS:*:* | convex:ConvexOS:C1*:*) - echo c1-convex-bsd - exit 0 ;; - C2*:ConvexOS:*:* | convex:ConvexOS:C2*:*) - if getsysinfo -f scalar_acc - then echo c32-convex-bsd - else echo c2-convex-bsd - fi - exit 0 ;; - C34*:ConvexOS:*:* | convex:ConvexOS:C34*:*) - echo c34-convex-bsd - exit 0 ;; - C38*:ConvexOS:*:* | convex:ConvexOS:C38*:*) - echo c38-convex-bsd - exit 0 ;; - C4*:ConvexOS:*:* | convex:ConvexOS:C4*:*) - echo c4-convex-bsd - exit 0 ;; - CRAY*Y-MP:*:*:*) - echo ymp-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*[A-Z]90:*:*:*) - echo ${UNAME_MACHINE}-cray-unicos${UNAME_RELEASE} \ - | sed -e 's/CRAY.*\([A-Z]90\)/\1/' \ - -e y/ABCDEFGHIJKLMNOPQRSTUVWXYZ/abcdefghijklmnopqrstuvwxyz/ \ - -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*TS:*:*:*) - echo t90-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*T3D:*:*:*) - echo alpha-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*T3E:*:*:*) - echo alphaev5-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*SV1:*:*:*) - echo sv1-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - F30[01]:UNIX_System_V:*:* | F700:UNIX_System_V:*:*) - FUJITSU_PROC=`uname -m | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz'` - FUJITSU_SYS=`uname -p | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz' | sed -e 's/\///'` - FUJITSU_REL=`echo ${UNAME_RELEASE} | sed -e 's/ /_/'` - echo "${FUJITSU_PROC}-fujitsu-${FUJITSU_SYS}${FUJITSU_REL}" - exit 0 ;; - i*86:BSD/386:*:* | i*86:BSD/OS:*:* | *:Ascend\ Embedded/OS:*:*) - echo ${UNAME_MACHINE}-pc-bsdi${UNAME_RELEASE} - exit 0 ;; - sparc*:BSD/OS:*:*) - echo sparc-unknown-bsdi${UNAME_RELEASE} - exit 0 ;; - *:BSD/OS:*:*) - echo ${UNAME_MACHINE}-unknown-bsdi${UNAME_RELEASE} - exit 0 ;; - *:FreeBSD:*:*) - # Determine whether the default compiler uses glibc. - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - #include - #if __GLIBC__ >= 2 - LIBC=gnu - #else - LIBC= - #endif -EOF - eval `$CC_FOR_BUILD -E $dummy.c 2>/dev/null | grep ^LIBC=` - rm -f $dummy.c && rmdir $tmpdir - echo ${UNAME_MACHINE}-unknown-freebsd`echo ${UNAME_RELEASE}|sed -e 's/[-(].*//'`${LIBC:+-$LIBC} - exit 0 ;; - i*:CYGWIN*:*) - echo ${UNAME_MACHINE}-pc-cygwin - exit 0 ;; - i*:MINGW*:*) - echo ${UNAME_MACHINE}-pc-mingw32 - exit 0 ;; - i*:PW*:*) - echo ${UNAME_MACHINE}-pc-pw32 - exit 0 ;; - x86:Interix*:3*) - echo i386-pc-interix3 - exit 0 ;; - i*:Windows_NT*:* | Pentium*:Windows_NT*:*) - # How do we know it's Interix rather than the generic POSIX subsystem? - # It also conflicts with pre-2.0 versions of AT&T UWIN. Should we - # UNAME_MACHINE based on the output of uname instead of i386? - echo i386-pc-interix - exit 0 ;; - i*:UWIN*:*) - echo ${UNAME_MACHINE}-pc-uwin - exit 0 ;; - p*:CYGWIN*:*) - echo powerpcle-unknown-cygwin - exit 0 ;; - prep*:SunOS:5.*:*) - echo powerpcle-unknown-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - *:GNU:*:*) - echo `echo ${UNAME_MACHINE}|sed -e 's,[-/].*$,,'`-unknown-gnu`echo ${UNAME_RELEASE}|sed -e 's,/.*$,,'` - exit 0 ;; - i*86:Minix:*:*) - echo ${UNAME_MACHINE}-pc-minix - exit 0 ;; - arm*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - ia64:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - m68*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - mips:Linux:*:*) - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - #undef CPU - #undef mips - #undef mipsel - #if defined(__MIPSEL__) || defined(__MIPSEL) || defined(_MIPSEL) || defined(MIPSEL) - CPU=mipsel - #else - #if defined(__MIPSEB__) || defined(__MIPSEB) || defined(_MIPSEB) || defined(MIPSEB) - CPU=mips - #else - CPU= - #endif - #endif -EOF - eval `$CC_FOR_BUILD -E $dummy.c 2>/dev/null | grep ^CPU=` - rm -f $dummy.c && rmdir $tmpdir - test x"${CPU}" != x && echo "${CPU}-pc-linux-gnu" && exit 0 - ;; - ppc:Linux:*:*) - echo powerpc-unknown-linux-gnu - exit 0 ;; - ppc64:Linux:*:*) - echo powerpc64-unknown-linux-gnu - exit 0 ;; - alpha:Linux:*:*) - case `sed -n '/^cpu model/s/^.*: \(.*\)/\1/p' < /proc/cpuinfo` in - EV5) UNAME_MACHINE=alphaev5 ;; - EV56) UNAME_MACHINE=alphaev56 ;; - PCA56) UNAME_MACHINE=alphapca56 ;; - PCA57) UNAME_MACHINE=alphapca56 ;; - EV6) UNAME_MACHINE=alphaev6 ;; - EV67) UNAME_MACHINE=alphaev67 ;; - EV68*) UNAME_MACHINE=alphaev68 ;; - esac - objdump --private-headers /bin/sh | grep ld.so.1 >/dev/null - if test "$?" = 0 ; then LIBC="libc1" ; else LIBC="" ; fi - echo ${UNAME_MACHINE}-unknown-linux-gnu${LIBC} - exit 0 ;; - parisc:Linux:*:* | hppa:Linux:*:*) - # Look for CPU level - case `grep '^cpu[^a-z]*:' /proc/cpuinfo 2>/dev/null | cut -d' ' -f2` in - PA7*) echo hppa1.1-unknown-linux-gnu ;; - PA8*) echo hppa2.0-unknown-linux-gnu ;; - *) echo hppa-unknown-linux-gnu ;; - esac - exit 0 ;; - parisc64:Linux:*:* | hppa64:Linux:*:*) - echo hppa64-unknown-linux-gnu - exit 0 ;; - s390:Linux:*:* | s390x:Linux:*:*) - echo ${UNAME_MACHINE}-ibm-linux - exit 0 ;; - sh*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - sparc:Linux:*:* | sparc64:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - x86_64:Linux:*:*) - echo x86_64-unknown-linux-gnu - exit 0 ;; - i*86:Linux:*:*) - # The BFD linker knows what the default object file format is, so - # first see if it will tell us. cd to the root directory to prevent - # problems with other programs or directories called `ld' in the path. - # Set LC_ALL=C to ensure ld outputs messages in English. - ld_supported_targets=`cd /; LC_ALL=C ld --help 2>&1 \ - | sed -ne '/supported targets:/!d - s/[ ][ ]*/ /g - s/.*supported targets: *// - s/ .*// - p'` - case "$ld_supported_targets" in - elf32-i386) - TENTATIVE="${UNAME_MACHINE}-pc-linux-gnu" - ;; - a.out-i386-linux) - echo "${UNAME_MACHINE}-pc-linux-gnuaout" - exit 0 ;; - coff-i386) - echo "${UNAME_MACHINE}-pc-linux-gnucoff" - exit 0 ;; - "") - # Either a pre-BFD a.out linker (linux-gnuoldld) or - # one that does not give us useful --help. - echo "${UNAME_MACHINE}-pc-linux-gnuoldld" - exit 0 ;; - esac - # Determine whether the default compiler is a.out or elf - eval $set_cc_for_build - sed 's/^ //' << EOF >$dummy.c - #include - #ifdef __ELF__ - # ifdef __GLIBC__ - # if __GLIBC__ >= 2 - LIBC=gnu - # else - LIBC=gnulibc1 - # endif - # else - LIBC=gnulibc1 - # endif - #else - #ifdef __INTEL_COMPILER - LIBC=gnu - #else - LIBC=gnuaout - #endif - #endif -EOF - eval `$CC_FOR_BUILD -E $dummy.c 2>/dev/null | grep ^LIBC=` - rm -f $dummy.c && rmdir $tmpdir - test x"${LIBC}" != x && echo "${UNAME_MACHINE}-pc-linux-${LIBC}" && exit 0 - test x"${TENTATIVE}" != x && echo "${TENTATIVE}" && exit 0 - ;; - i*86:DYNIX/ptx:4*:*) - # ptx 4.0 does uname -s correctly, with DYNIX/ptx in there. - # earlier versions are messed up and put the nodename in both - # sysname and nodename. - echo i386-sequent-sysv4 - exit 0 ;; - i*86:UNIX_SV:4.2MP:2.*) - # Unixware is an offshoot of SVR4, but it has its own version - # number series starting with 2... - # I am not positive that other SVR4 systems won't match this, - # I just have to hope. -- rms. - # Use sysv4.2uw... so that sysv4* matches it. - echo ${UNAME_MACHINE}-pc-sysv4.2uw${UNAME_VERSION} - exit 0 ;; - i*86:*:4.*:* | i*86:SYSTEM_V:4.*:*) - UNAME_REL=`echo ${UNAME_RELEASE} | sed 's/\/MP$//'` - if grep Novell /usr/include/link.h >/dev/null 2>/dev/null; then - echo ${UNAME_MACHINE}-univel-sysv${UNAME_REL} - else - echo ${UNAME_MACHINE}-pc-sysv${UNAME_REL} - fi - exit 0 ;; - i*86:*:5:[78]*) - case `/bin/uname -X | grep "^Machine"` in - *486*) UNAME_MACHINE=i486 ;; - *Pentium) UNAME_MACHINE=i586 ;; - *Pent*|*Celeron) UNAME_MACHINE=i686 ;; - esac - echo ${UNAME_MACHINE}-unknown-sysv${UNAME_RELEASE}${UNAME_SYSTEM}${UNAME_VERSION} - exit 0 ;; - i*86:*:3.2:*) - if test -f /usr/options/cb.name; then - UNAME_REL=`sed -n 's/.*Version //p' /dev/null >/dev/null ; then - UNAME_REL=`(/bin/uname -X|grep Release|sed -e 's/.*= //')` - (/bin/uname -X|grep i80486 >/dev/null) && UNAME_MACHINE=i486 - (/bin/uname -X|grep '^Machine.*Pentium' >/dev/null) \ - && UNAME_MACHINE=i586 - (/bin/uname -X|grep '^Machine.*Pent *II' >/dev/null) \ - && UNAME_MACHINE=i686 - (/bin/uname -X|grep '^Machine.*Pentium Pro' >/dev/null) \ - && UNAME_MACHINE=i686 - echo ${UNAME_MACHINE}-pc-sco$UNAME_REL - else - echo ${UNAME_MACHINE}-pc-sysv32 - fi - exit 0 ;; - i*86:*DOS:*:*) - echo ${UNAME_MACHINE}-pc-msdosdjgpp - exit 0 ;; - pc:*:*:*) - # Left here for compatibility: - # uname -m prints for DJGPP always 'pc', but it prints nothing about - # the processor, so we play safe by assuming i386. - echo i386-pc-msdosdjgpp - exit 0 ;; - Intel:Mach:3*:*) - echo i386-pc-mach3 - exit 0 ;; - paragon:*:*:*) - echo i860-intel-osf1 - exit 0 ;; - i860:*:4.*:*) # i860-SVR4 - if grep Stardent /usr/include/sys/uadmin.h >/dev/null 2>&1 ; then - echo i860-stardent-sysv${UNAME_RELEASE} # Stardent Vistra i860-SVR4 - else # Add other i860-SVR4 vendors below as they are discovered. - echo i860-unknown-sysv${UNAME_RELEASE} # Unknown i860-SVR4 - fi - exit 0 ;; - mini*:CTIX:SYS*5:*) - # "miniframe" - echo m68010-convergent-sysv - exit 0 ;; - mc68k:UNIX:SYSTEM5:3.51m) - echo m68k-convergent-sysv - exit 0 ;; - M68*:*:R3V[567]*:*) - test -r /sysV68 && echo 'm68k-motorola-sysv' && exit 0 ;; - 3[34]??:*:4.0:3.0 | 3[34]??A:*:4.0:3.0 | 3[34]??,*:*:4.0:3.0 | 3[34]??/*:*:4.0:3.0 | 4400:*:4.0:3.0 | 4850:*:4.0:3.0 | SKA40:*:4.0:3.0) - OS_REL='' - test -r /etc/.relid \ - && OS_REL=.`sed -n 's/[^ ]* [^ ]* \([0-9][0-9]\).*/\1/p' < /etc/.relid` - /bin/uname -p 2>/dev/null | grep 86 >/dev/null \ - && echo i486-ncr-sysv4.3${OS_REL} && exit 0 - /bin/uname -p 2>/dev/null | /bin/grep entium >/dev/null \ - && echo i586-ncr-sysv4.3${OS_REL} && exit 0 ;; - 3[34]??:*:4.0:* | 3[34]??,*:*:4.0:*) - /bin/uname -p 2>/dev/null | grep 86 >/dev/null \ - && echo i486-ncr-sysv4 && exit 0 ;; - m68*:LynxOS:2.*:* | m68*:LynxOS:3.0*:*) - echo m68k-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - mc68030:UNIX_System_V:4.*:*) - echo m68k-atari-sysv4 - exit 0 ;; - i*86:LynxOS:2.*:* | i*86:LynxOS:3.[01]*:* | i*86:LynxOS:4.0*:*) - echo i386-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - TSUNAMI:LynxOS:2.*:*) - echo sparc-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - rs6000:LynxOS:2.*:*) - echo rs6000-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - PowerPC:LynxOS:2.*:* | PowerPC:LynxOS:3.[01]*:* | PowerPC:LynxOS:4.0*:*) - echo powerpc-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - SM[BE]S:UNIX_SV:*:*) - echo mips-dde-sysv${UNAME_RELEASE} - exit 0 ;; - RM*:ReliantUNIX-*:*:*) - echo mips-sni-sysv4 - exit 0 ;; - RM*:SINIX-*:*:*) - echo mips-sni-sysv4 - exit 0 ;; - *:SINIX-*:*:*) - if uname -p 2>/dev/null >/dev/null ; then - UNAME_MACHINE=`(uname -p) 2>/dev/null` - echo ${UNAME_MACHINE}-sni-sysv4 - else - echo ns32k-sni-sysv - fi - exit 0 ;; - PENTIUM:*:4.0*:*) # Unisys `ClearPath HMP IX 4000' SVR4/MP effort - # says - echo i586-unisys-sysv4 - exit 0 ;; - *:UNIX_System_V:4*:FTX*) - # From Gerald Hewes . - # How about differentiating between stratus architectures? -djm - echo hppa1.1-stratus-sysv4 - exit 0 ;; - *:*:*:FTX*) - # From seanf@swdc.stratus.com. - echo i860-stratus-sysv4 - exit 0 ;; - *:VOS:*:*) - # From Paul.Green@stratus.com. - echo hppa1.1-stratus-vos - exit 0 ;; - mc68*:A/UX:*:*) - echo m68k-apple-aux${UNAME_RELEASE} - exit 0 ;; - news*:NEWS-OS:6*:*) - echo mips-sony-newsos6 - exit 0 ;; - R[34]000:*System_V*:*:* | R4000:UNIX_SYSV:*:* | R*000:UNIX_SV:*:*) - if [ -d /usr/nec ]; then - echo mips-nec-sysv${UNAME_RELEASE} - else - echo mips-unknown-sysv${UNAME_RELEASE} - fi - exit 0 ;; - BeBox:BeOS:*:*) # BeOS running on hardware made by Be, PPC only. - echo powerpc-be-beos - exit 0 ;; - BeMac:BeOS:*:*) # BeOS running on Mac or Mac clone, PPC only. - echo powerpc-apple-beos - exit 0 ;; - BePC:BeOS:*:*) # BeOS running on Intel PC compatible. - echo i586-pc-beos - exit 0 ;; - SX-4:SUPER-UX:*:*) - echo sx4-nec-superux${UNAME_RELEASE} - exit 0 ;; - SX-5:SUPER-UX:*:*) - echo sx5-nec-superux${UNAME_RELEASE} - exit 0 ;; - SX-6:SUPER-UX:*:*) - echo sx6-nec-superux${UNAME_RELEASE} - exit 0 ;; - Power*:Rhapsody:*:*) - echo powerpc-apple-rhapsody${UNAME_RELEASE} - exit 0 ;; - *:Rhapsody:*:*) - echo ${UNAME_MACHINE}-apple-rhapsody${UNAME_RELEASE} - exit 0 ;; - *:Darwin:*:*) - echo `uname -p`-apple-darwin${UNAME_RELEASE} - exit 0 ;; - *:procnto*:*:* | *:QNX:[0123456789]*:*) - UNAME_PROCESSOR=`uname -p` - if test "$UNAME_PROCESSOR" = "x86"; then - UNAME_PROCESSOR=i386 - UNAME_MACHINE=pc - fi - echo ${UNAME_PROCESSOR}-${UNAME_MACHINE}-nto-qnx${UNAME_RELEASE} - exit 0 ;; - *:QNX:*:4*) - echo i386-pc-qnx - exit 0 ;; - NSR-[DGKLNPTVW]:NONSTOP_KERNEL:*:*) - echo nsr-tandem-nsk${UNAME_RELEASE} - exit 0 ;; - *:NonStop-UX:*:*) - echo mips-compaq-nonstopux - exit 0 ;; - BS2000:POSIX*:*:*) - echo bs2000-siemens-sysv - exit 0 ;; - DS/*:UNIX_System_V:*:*) - echo ${UNAME_MACHINE}-${UNAME_SYSTEM}-${UNAME_RELEASE} - exit 0 ;; - *:Plan9:*:*) - # "uname -m" is not consistent, so use $cputype instead. 386 - # is converted to i386 for consistency with other x86 - # operating systems. - if test "$cputype" = "386"; then - UNAME_MACHINE=i386 - else - UNAME_MACHINE="$cputype" - fi - echo ${UNAME_MACHINE}-unknown-plan9 - exit 0 ;; - i*86:OS/2:*:*) - # If we were able to find `uname', then EMX Unix compatibility - # is probably installed. - echo ${UNAME_MACHINE}-pc-os2-emx - exit 0 ;; - *:TOPS-10:*:*) - echo pdp10-unknown-tops10 - exit 0 ;; - *:TENEX:*:*) - echo pdp10-unknown-tenex - exit 0 ;; - KS10:TOPS-20:*:* | KL10:TOPS-20:*:* | TYPE4:TOPS-20:*:*) - echo pdp10-dec-tops20 - exit 0 ;; - XKL-1:TOPS-20:*:* | TYPE5:TOPS-20:*:*) - echo pdp10-xkl-tops20 - exit 0 ;; - *:TOPS-20:*:*) - echo pdp10-unknown-tops20 - exit 0 ;; - *:ITS:*:*) - echo pdp10-unknown-its - exit 0 ;; - i*86:XTS-300:*:STOP) - echo ${UNAME_MACHINE}-unknown-stop - exit 0 ;; - i*86:atheos:*:*) - echo ${UNAME_MACHINE}-unknown-atheos - exit 0 ;; -esac - -#echo '(No uname command or uname output not recognized.)' 1>&2 -#echo "${UNAME_MACHINE}:${UNAME_SYSTEM}:${UNAME_RELEASE}:${UNAME_VERSION}" 1>&2 - -eval $set_cc_for_build -cat >$dummy.c < -# include -#endif -main () -{ -#if defined (sony) -#if defined (MIPSEB) - /* BFD wants "bsd" instead of "newsos". Perhaps BFD should be changed, - I don't know.... */ - printf ("mips-sony-bsd\n"); exit (0); -#else -#include - printf ("m68k-sony-newsos%s\n", -#ifdef NEWSOS4 - "4" -#else - "" -#endif - ); exit (0); -#endif -#endif - -#if defined (__arm) && defined (__acorn) && defined (__unix) - printf ("arm-acorn-riscix"); exit (0); -#endif - -#if defined (hp300) && !defined (hpux) - printf ("m68k-hp-bsd\n"); exit (0); -#endif - -#if defined (NeXT) -#if !defined (__ARCHITECTURE__) -#define __ARCHITECTURE__ "m68k" -#endif - int version; - version=`(hostinfo | sed -n 's/.*NeXT Mach \([0-9]*\).*/\1/p') 2>/dev/null`; - if (version < 4) - printf ("%s-next-nextstep%d\n", __ARCHITECTURE__, version); - else - printf ("%s-next-openstep%d\n", __ARCHITECTURE__, version); - exit (0); -#endif - -#if defined (MULTIMAX) || defined (n16) -#if defined (UMAXV) - printf ("ns32k-encore-sysv\n"); exit (0); -#else -#if defined (CMU) - printf ("ns32k-encore-mach\n"); exit (0); -#else - printf ("ns32k-encore-bsd\n"); exit (0); -#endif -#endif -#endif - -#if defined (__386BSD__) - printf ("i386-pc-bsd\n"); exit (0); -#endif - -#if defined (sequent) -#if defined (i386) - printf ("i386-sequent-dynix\n"); exit (0); -#endif -#if defined (ns32000) - printf ("ns32k-sequent-dynix\n"); exit (0); -#endif -#endif - -#if defined (_SEQUENT_) - struct utsname un; - - uname(&un); - - if (strncmp(un.version, "V2", 2) == 0) { - printf ("i386-sequent-ptx2\n"); exit (0); - } - if (strncmp(un.version, "V1", 2) == 0) { /* XXX is V1 correct? */ - printf ("i386-sequent-ptx1\n"); exit (0); - } - printf ("i386-sequent-ptx\n"); exit (0); - -#endif - -#if defined (vax) -# if !defined (ultrix) -# include -# if defined (BSD) -# if BSD == 43 - printf ("vax-dec-bsd4.3\n"); exit (0); -# else -# if BSD == 199006 - printf ("vax-dec-bsd4.3reno\n"); exit (0); -# else - printf ("vax-dec-bsd\n"); exit (0); -# endif -# endif -# else - printf ("vax-dec-bsd\n"); exit (0); -# endif -# else - printf ("vax-dec-ultrix\n"); exit (0); -# endif -#endif - -#if defined (alliant) && defined (i860) - printf ("i860-alliant-bsd\n"); exit (0); -#endif - - exit (1); -} -EOF - -$CC_FOR_BUILD -o $dummy $dummy.c 2>/dev/null && $dummy && rm -f $dummy.c $dummy && rmdir $tmpdir && exit 0 -rm -f $dummy.c $dummy && rmdir $tmpdir - -# Apollos put the system type in the environment. - -test -d /usr/apollo && { echo ${ISP}-apollo-${SYSTYPE}; exit 0; } - -# Convex versions that predate uname can use getsysinfo(1) - -if [ -x /usr/convex/getsysinfo ] -then - case `getsysinfo -f cpu_type` in - c1*) - echo c1-convex-bsd - exit 0 ;; - c2*) - if getsysinfo -f scalar_acc - then echo c32-convex-bsd - else echo c2-convex-bsd - fi - exit 0 ;; - c34*) - echo c34-convex-bsd - exit 0 ;; - c38*) - echo c38-convex-bsd - exit 0 ;; - c4*) - echo c4-convex-bsd - exit 0 ;; - esac -fi - -cat >&2 < in order to provide the needed -information to handle your system. - -config.guess timestamp = $timestamp - -uname -m = `(uname -m) 2>/dev/null || echo unknown` -uname -r = `(uname -r) 2>/dev/null || echo unknown` -uname -s = `(uname -s) 2>/dev/null || echo unknown` -uname -v = `(uname -v) 2>/dev/null || echo unknown` - -/usr/bin/uname -p = `(/usr/bin/uname -p) 2>/dev/null` -/bin/uname -X = `(/bin/uname -X) 2>/dev/null` - -hostinfo = `(hostinfo) 2>/dev/null` -/bin/universe = `(/bin/universe) 2>/dev/null` -/usr/bin/arch -k = `(/usr/bin/arch -k) 2>/dev/null` -/bin/arch = `(/bin/arch) 2>/dev/null` -/usr/bin/oslevel = `(/usr/bin/oslevel) 2>/dev/null` -/usr/convex/getsysinfo = `(/usr/convex/getsysinfo) 2>/dev/null` - -UNAME_MACHINE = ${UNAME_MACHINE} -UNAME_RELEASE = ${UNAME_RELEASE} -UNAME_SYSTEM = ${UNAME_SYSTEM} -UNAME_VERSION = ${UNAME_VERSION} -EOF - -exit 1 - -# Local variables: -# eval: (add-hook 'write-file-hooks 'time-stamp) -# time-stamp-start: "timestamp='" -# time-stamp-format: "%:y-%02m-%02d" -# time-stamp-end: "'" -# End: diff --git a/engine/voice_codecs/speex/source/config.sub b/engine/voice_codecs/speex/source/config.sub deleted file mode 100644 index 1dea9b79..00000000 --- a/engine/voice_codecs/speex/source/config.sub +++ /dev/null @@ -1,1470 +0,0 @@ -#! /bin/sh -# Configuration validation subroutine script. -# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, -# 2000, 2001, 2002 Free Software Foundation, Inc. - -timestamp='2002-09-05' - -# This file is (in principle) common to ALL GNU software. -# The presence of a machine in this file suggests that SOME GNU software -# can handle that machine. It does not imply ALL GNU software can. -# -# This file is free software; you can redistribute it and/or modify -# it under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 2 of the License, or -# (at your option) any later version. -# -# This program is distributed in the hope that it will be useful, -# but WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -# GNU General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 59 Temple Place - Suite 330, -# Boston, MA 02111-1307, USA. - -# As a special exception to the GNU General Public License, if you -# distribute this file as part of a program that contains a -# configuration script generated by Autoconf, you may include it under -# the same distribution terms that you use for the rest of that program. - -# Please send patches to . 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- -# Tell versions [3.59,3.63) of GNU make to not export all variables. -# Otherwise a system limit (for SysV at least) may be exceeded. -.NOEXPORT: diff --git a/engine/voice_codecs/speex/source/libspeex/bits.c b/engine/voice_codecs/speex/source/libspeex/bits.c deleted file mode 100644 index 68285e04..00000000 --- a/engine/voice_codecs/speex/source/libspeex/bits.c +++ /dev/null @@ -1,352 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: speex_bits.c - - Handles bit packing/unpacking - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - -#include "speex_bits.h" -#include "misc.h" - -void speex_bits_init(SpeexBits *bits) -{ - int i; - bits->bytes = (char*)speex_alloc(MAX_BYTES_PER_FRAME); - bits->buf_size = MAX_BYTES_PER_FRAME; - - for (i=0;ibuf_size;i++) - bits->bytes[i]=0; - bits->nbBits=0; - bits->bytePtr=0; - bits->bitPtr=0; - bits->owner=1; - bits->overflow=0; -} - -void speex_bits_init_buffer(SpeexBits *bits, void *buff, int buf_size) -{ - int i; - bits->bytes = (char*)buff; - bits->buf_size = buf_size; - - for (i=0;ibytes[i]=0; - bits->nbBits=0; - bits->bytePtr=0; - bits->bitPtr=0; - bits->owner=0; - bits->overflow=0; -} - -void speex_bits_destroy(SpeexBits *bits) -{ - if (bits->owner) - speex_free(bits->bytes); - /* Will do something once the allocation is dynamic */ -} - -void speex_bits_reset(SpeexBits *bits) -{ - int i; - for (i=0;ibuf_size;i++) - bits->bytes[i]=0; - bits->nbBits=0; - bits->bytePtr=0; - bits->bitPtr=0; - bits->overflow=0; -} - -void speex_bits_rewind(SpeexBits *bits) -{ - bits->bytePtr=0; - bits->bitPtr=0; - bits->overflow=0; -} - -void speex_bits_read_from(SpeexBits *bits, char *bytes, int len) -{ - int i; - if (len > bits->buf_size) - { - speex_warning_int("Packet if larger than allocated buffer: ", len); - if (bits->owner) - { - char *tmp = (char*)speex_realloc(bits->bytes, len); - if (tmp) - { - bits->buf_size=len; - bits->bytes=tmp; - } else { - len=bits->buf_size; - speex_warning("Could not resize input buffer: truncating input"); - } - } else { - speex_warning("Do not own input buffer: truncating input"); - len=bits->buf_size; - } - } - for (i=0;ibytes[i]=bytes[i]; - bits->nbBits=len<<3; - bits->bytePtr=0; - bits->bitPtr=0; - bits->overflow=0; -} - -static void speex_bits_flush(SpeexBits *bits) -{ - int i; - if (bits->bytePtr>0) - { - for (i=bits->bytePtr;i<((bits->nbBits+7)>>3);i++) - bits->bytes[i-bits->bytePtr]=bits->bytes[i]; - } - bits->nbBits -= bits->bytePtr<<3; - bits->bytePtr=0; -} - -void speex_bits_read_whole_bytes(SpeexBits *bits, char *bytes, int len) -{ - int i,pos; - - if ((bits->nbBits>>3)+len+1 > bits->buf_size) - { - speex_warning_int("Packet if larger than allocated buffer: ", len); - if (bits->owner) - { - char *tmp = (char*)speex_realloc(bits->bytes, (bits->nbBits>>3)+len+1); - if (tmp) - { - bits->buf_size=(bits->nbBits>>3)+len+1; - bits->bytes=tmp; - } else { - len=bits->buf_size-(bits->nbBits>>3)-1; - speex_warning("Could not resize input buffer: truncating input"); - } - } else { - speex_warning("Do not own input buffer: truncating input"); - len=bits->buf_size; - } - } - - speex_bits_flush(bits); - pos=bits->nbBits>>3; - for (i=0;ibytes[pos+i]=bytes[i]; - bits->nbBits+=len<<3; -} - -int speex_bits_write(SpeexBits *bits, char *bytes, int max_len) -{ - int i; - if (max_len > ((bits->nbBits+7)>>3)) - max_len = ((bits->nbBits+7)>>3); - for (i=0;ibytes[i]; - return max_len; -} - -int speex_bits_write_whole_bytes(SpeexBits *bits, char *bytes, int max_len) -{ - int i; - if (max_len > ((bits->nbBits)>>3)) - max_len = ((bits->nbBits)>>3); - for (i=0;ibytes[i]; - - if (bits->bitPtr>0) - bits->bytes[0]=bits->bytes[max_len]; - else - bits->bytes[0]=0; - for (i=1;i<((bits->nbBits)>>3)+1;i++) - bits->bytes[i]=0; - bits->bytePtr=0; - bits->nbBits &= 7; - return max_len; -} - - -void speex_bits_pack(SpeexBits *bits, int data, int nbBits) -{ - int i; - unsigned int d=data; - - if (bits->bytePtr+((nbBits+bits->bitPtr)>>3) >= bits->buf_size) - { - speex_warning("Buffer too small to pack bits"); - if (bits->owner) - { - char *tmp = (char*)speex_realloc(bits->bytes, ((bits->buf_size+5)*3)>>1); - if (tmp) - { - for (i=bits->buf_size;i<(((bits->buf_size+5)*3)>>1);i++) - tmp[i]=0; - bits->buf_size=((bits->buf_size+5)*3)>>1; - bits->bytes=tmp; - } else { - speex_warning("Could not resize input buffer: not packing"); - return; - } - } else { - speex_warning("Do not own input buffer: not packing"); - return; - } - } - - while(nbBits) - { - int bit; - bit = (d>>(nbBits-1))&1; - bits->bytes[bits->bytePtr] |= bit<<(7-bits->bitPtr); - bits->bitPtr++; - - if (bits->bitPtr==8) - { - bits->bitPtr=0; - bits->bytePtr++; - } - bits->nbBits++; - nbBits--; - } -} - -int speex_bits_unpack_signed(SpeexBits *bits, int nbBits) -{ - unsigned int d=speex_bits_unpack_unsigned(bits,nbBits); - /* If number is negative */ - if (d>>(nbBits-1)) - { - d |= (-1)<bytePtr<<3)+bits->bitPtr+nbBits>bits->nbBits) - bits->overflow=1; - if (bits->overflow) - return 0; - while(nbBits) - { - d<<=1; - d |= (bits->bytes[bits->bytePtr]>>(7-bits->bitPtr))&1; - bits->bitPtr++; - if (bits->bitPtr==8) - { - bits->bitPtr=0; - bits->bytePtr++; - } - nbBits--; - } - return d; -} - -unsigned int speex_bits_peek_unsigned(SpeexBits *bits, int nbBits) -{ - unsigned int d=0; - int bitPtr, bytePtr; - char *bytes; - - if ((bits->bytePtr<<3)+bits->bitPtr+nbBits>bits->nbBits) - bits->overflow=1; - if (bits->overflow) - return 0; - - bitPtr=bits->bitPtr; - bytePtr=bits->bytePtr; - bytes = bits->bytes; - while(nbBits) - { - d<<=1; - d |= (bytes[bytePtr]>>(7-bitPtr))&1; - bitPtr++; - if (bitPtr==8) - { - bitPtr=0; - bytePtr++; - } - nbBits--; - } - return d; -} - -int speex_bits_peek(SpeexBits *bits) -{ - if ((bits->bytePtr<<3)+bits->bitPtr+1>bits->nbBits) - bits->overflow=1; - if (bits->overflow) - return 0; - return (bits->bytes[bits->bytePtr]>>(7-bits->bitPtr))&1; -} - -void speex_bits_advance(SpeexBits *bits, int n) -{ - int nbytes, nbits; - - if ((bits->bytePtr<<3)+bits->bitPtr+n>bits->nbBits) - bits->overflow=1; - if (bits->overflow) - return; - - nbytes = n >> 3; - nbits = n & 7; - - bits->bytePtr += nbytes; - bits->bitPtr += nbits; - - if (bits->bitPtr>7) - { - bits->bitPtr-=8; - bits->bytePtr++; - } -} - -int speex_bits_remaining(SpeexBits *bits) -{ - if (bits->overflow) - return -1; - else - return bits->nbBits-((bits->bytePtr<<3)+bits->bitPtr); -} - -int speex_bits_nbytes(SpeexBits *bits) -{ - return ((bits->nbBits+7)>>3); -} - -void speex_bits_insert_terminator(SpeexBits *bits) -{ - if (bits->bitPtr<7) - speex_bits_pack(bits, 0, 1); - while (bits->bitPtr<7) - speex_bits_pack(bits, 1, 1); -} diff --git a/engine/voice_codecs/speex/source/libspeex/cb_search.c b/engine/voice_codecs/speex/source/libspeex/cb_search.c deleted file mode 100644 index 785bb49b..00000000 --- a/engine/voice_codecs/speex/source/libspeex/cb_search.c +++ /dev/null @@ -1,387 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: cb_search.c - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - - -#include "cb_search.h" -#include "filters.h" -#include "stack_alloc.h" -#include "vq.h" -#include "misc.h" - -void split_cb_search_shape_sign( -float target[], /* target vector */ -float ak[], /* LPCs for this subframe */ -float awk1[], /* Weighted LPCs for this subframe */ -float awk2[], /* Weighted LPCs for this subframe */ -void *par, /* Codebook/search parameters*/ -int p, /* number of LPC coeffs */ -int nsf, /* number of samples in subframe */ -float *exc, -float *r, -SpeexBits *bits, -char *stack, -int complexity -) -{ - int i,j,k,m,n,q; - float *resp; - float *t, *e, *E, *r2; - float *tmp; - float *ndist, *odist; - int *itmp; - float **ot, **nt; - int **nind, **oind; - int *ind; - signed char *shape_cb; - int shape_cb_size, subvect_size, nb_subvect; - split_cb_params *params; - int N=2; - int *best_index; - float *best_dist; - int have_sign; - - N=complexity; - if (N>10) - N=10; - - ot=PUSH(stack, N, float*); - nt=PUSH(stack, N, float*); - oind=PUSH(stack, N, int*); - nind=PUSH(stack, N, int*); - - params = (split_cb_params *) par; - subvect_size = params->subvect_size; - nb_subvect = params->nb_subvect; - shape_cb_size = 1<shape_bits; - shape_cb = params->shape_cb; - have_sign = params->have_sign; - resp = PUSH(stack, shape_cb_size*subvect_size, float); - t = PUSH(stack, nsf, float); - e = PUSH(stack, nsf, float); - r2 = PUSH(stack, nsf, float); - E = PUSH(stack, shape_cb_size, float); - ind = PUSH(stack, nb_subvect, int); - - tmp = PUSH(stack, 2*N*nsf, float); - for (i=0;i=shape_cb_size) - { - sign=-1; - rind-=shape_cb_size; - } - res = resp+rind*subvect_size; - if (sign>0) - for (m=0;m=shape_cb_size) - { - sign=-1; - rind-=shape_cb_size; - } - - g=sign*0.03125*shape_cb[rind*subvect_size+m]; - q=subvect_size-m; - for (n=subvect_size*(i+1);nm;n--) - { - for (q=(i+1)*subvect_size;qshape_bits+have_sign); - } - - /* Put everything back together */ - for (i=0;i=shape_cb_size) - { - sign=-1; - rind-=shape_cb_size; - } - - for (j=0;jsubvect_size; - nb_subvect = params->nb_subvect; - shape_cb_size = 1<shape_bits; - shape_cb = params->shape_cb; - have_sign = params->have_sign; - - ind = PUSH(stack, nb_subvect, int); - signs = PUSH(stack, nb_subvect, int); - - /* Decode codewords and gains */ - for (i=0;ishape_bits); - } - /* Compute decoded excitation */ - for (i=0;i - - -void bw_lpc(float gamma, float *lpc_in, float *lpc_out, int order) -{ - int i; - float tmp=1; - for (i=0;i>1; - for (i=0;ilast_pitch=0; - mem->last_pitch_gain[0]=mem->last_pitch_gain[1]=mem->last_pitch_gain[2]=0; - mem->smooth_gain=1; -} - -void comb_filter( -float *exc, /*decoded excitation*/ -float *new_exc, /*enhanced excitation*/ -float *ak, /*LPC filter coefs*/ -int p, /*LPC order*/ -int nsf, /*sub-frame size*/ -int pitch, /*pitch period*/ -float *pitch_gain, /*pitch gain (3-tap)*/ -float comb_gain, /*gain of comb filter*/ -CombFilterMem *mem -) -{ - int i; - float exc_energy=0, new_exc_energy=0; - float gain; - float step; - float fact; - /*Compute excitation energy prior to enhancement*/ - for (i=0;ilast_pitch_gain[0] + mem->last_pitch_gain[1] + mem->last_pitch_gain[2]); - if (g>1.3) - comb_gain*=1.3/g; - if (g<.5) - comb_gain*=2*g; - } - step = 1.0/nsf; - fact=0; - /*Apply pitch comb-filter (filter out noise between pitch harmonics)*/ - for (i=0;ilast_pitch_gain[0]*exc[i-mem->last_pitch+1] + - mem->last_pitch_gain[1]*exc[i-mem->last_pitch] + - mem->last_pitch_gain[2]*exc[i-mem->last_pitch-1] - ); - } - - mem->last_pitch_gain[0] = pitch_gain[0]; - mem->last_pitch_gain[1] = pitch_gain[1]; - mem->last_pitch_gain[2] = pitch_gain[2]; - mem->last_pitch = pitch; - - /*Gain after enhancement*/ - for (i=0;i1) - gain=1; - - for (i=0;ismooth_gain = .96*mem->smooth_gain + .04*gain; - new_exc[i] *= mem->smooth_gain; - } -} diff --git a/engine/voice_codecs/speex/source/libspeex/filters.h b/engine/voice_codecs/speex/source/libspeex/filters.h deleted file mode 100644 index ad6dc022..00000000 --- a/engine/voice_codecs/speex/source/libspeex/filters.h +++ /dev/null @@ -1,79 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: filters.h - Various analysis/synthesis filters - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef FILTERS_H -#define FILTERS_H - - -typedef struct CombFilterMem { - int last_pitch; - float last_pitch_gain[3]; - float smooth_gain; -} CombFilterMem; - - -void qmf_decomp(float *xx, float *aa, float *y1, float *y2, int N, int M, float *mem, char *stack); -void fir_mem_up(float *x, float *a, float *y, int N, int M, float *mem, char *stack); - - -void filter_mem2(float *x, float *num, float *den, float *y, int N, int ord, float *mem); -void fir_mem2(float *x, float *num, float *y, int N, int ord, float *mem); -void iir_mem2(float *x, float *den, float *y, int N, int ord, float *mem); - -/* Apply bandwidth expansion on LPC coef */ -void bw_lpc(float gamma, float *lpc_in, float *lpc_out, int order); - - - -/* FIR filter */ -void fir_decim_mem(float *x, float *a, float *y, int N, int M, float *mem); - -void syn_percep_zero(float *x, float *ak, float *awk1, float *awk2, float *y, int N, int ord, char *stack); - -void residue_percep_zero(float *xx, float *ak, float *awk1, float *awk2, float *y, int N, int ord, char *stack); - -void comp_filter_mem_init (CombFilterMem *mem); - -void comb_filter( -float *exc, /*decoded excitation*/ -float *new_exc, /*enhanced excitation*/ -float *ak, /*LPC filter coefs*/ -int p, /*LPC order*/ -int nsf, /*sub-frame size*/ -int pitch, /*pitch period*/ -float *pitch_gain, /*pitch gain (3-tap)*/ -float comb_gain, /*gain of comb filter*/ -CombFilterMem *mem -); - - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/filters_sse.h b/engine/voice_codecs/speex/source/libspeex/filters_sse.h deleted file mode 100644 index 3fba6c8d..00000000 --- a/engine/voice_codecs/speex/source/libspeex/filters_sse.h +++ /dev/null @@ -1,289 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: filters.c - Various analysis/synthesis filters - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -void filter_mem2(float *x, float *_num, float *_den, float *y, int N, int ord, float *_mem) -{ - float __num[20], __den[20], __mem[20]; - float *num, *den, *mem; - int i; - - num = (float*)(((int)(__num+4))&0xfffffff0)-1; - den = (float*)(((int)(__den+4))&0xfffffff0)-1; - mem = (float*)(((int)(__mem+4))&0xfffffff0)-1; - for (i=0;i<=10;i++) - num[i]=den[i]=0; - for (i=0;i<10;i++) - mem[i]=0; - - for (i=0;i x(n) * x(n-i) for all n - * `--' - * for lags between 0 and lag-1, and x == 0 outside 0...n-1 - */ -void _spx_autocorr( - const float * x, /* in: [0...n-1] samples x */ - float *ac, /* out: [0...lag-1] ac values */ - int lag, int n) -{ - float d; int i; - while (lag--) { - for (i = lag, d = 0; i < n; i++) d += x[i] * x[i-lag]; - ac[lag] = d; - } -} diff --git a/engine/voice_codecs/speex/source/libspeex/lpc.h b/engine/voice_codecs/speex/source/libspeex/lpc.h deleted file mode 100644 index e3f70b5a..00000000 --- a/engine/voice_codecs/speex/source/libspeex/lpc.h +++ /dev/null @@ -1,50 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: lpc.h - Functions for LPC (Linear Prediction Coefficients) analysis - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef LPC_H -#define LPC_H - -void _spx_autocorr( - const float * x, /* in: [0...n-1] samples x */ - float *ac, /* out: [0...lag-1] ac values */ - int lag, int n); - -float /* returns minimum mean square error */ -wld( - float * lpc, /* [0...p-1] LPC coefficients */ - const float * ac, /* in: [0...p] autocorrelation values */ - float * ref, /* out: [0...p-1] reflection coef's */ - int p - ); - - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/lsp.c b/engine/voice_codecs/speex/source/libspeex/lsp.c deleted file mode 100644 index 1af762cd..00000000 --- a/engine/voice_codecs/speex/source/libspeex/lsp.c +++ /dev/null @@ -1,328 +0,0 @@ -/*---------------------------------------------------------------------------*\ -Original copyright - FILE........: AKSLSPD.C - TYPE........: Turbo C - COMPANY.....: Voicetronix - AUTHOR......: David Rowe - DATE CREATED: 24/2/93 - -Modified by Jean-Marc Valin - - This file contains functions for converting Linear Prediction - Coefficients (LPC) to Line Spectral Pair (LSP) and back. Note that the - LSP coefficients are not in radians format but in the x domain of the - unit circle. - - Speex License: - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include -#include "lsp.h" -#include "stack_alloc.h" - - -#ifndef M_PI -#define M_PI 3.14159265358979323846 /* pi */ -#endif - -#ifndef NULL -#define NULL 0 -#endif - -/*---------------------------------------------------------------------------*\ - - FUNCTION....: cheb_poly_eva() - - AUTHOR......: David Rowe - DATE CREATED: 24/2/93 - - This function evaluates a series of Chebyshev polynomials - -\*---------------------------------------------------------------------------*/ - - - -static float cheb_poly_eva(float *coef,float x,int m,char *stack) -/* float coef[] coefficients of the polynomial to be evaluated */ -/* float x the point where polynomial is to be evaluated */ -/* int m order of the polynomial */ -{ - int i; - float *T,sum; - int m2=m>>1; - - /* Allocate memory for Chebyshev series formulation */ - T=PUSH(stack, m2+1, float); - - /* Initialise values */ - T[0]=1; - T[1]=x; - - /* Evaluate Chebyshev series formulation using iterative approach */ - /* Evaluate polynomial and return value also free memory space */ - sum = coef[m2] + coef[m2-1]*x; - x *= 2; - for(i=2;i<=m2;i++) - { - T[i] = x*T[i-1] - T[i-2]; - sum += coef[m2-i] * T[i]; - } - - return sum; -} - - -/*---------------------------------------------------------------------------*\ - - FUNCTION....: lpc_to_lsp() - - AUTHOR......: David Rowe - DATE CREATED: 24/2/93 - - This function converts LPC coefficients to LSP - coefficients. - -\*---------------------------------------------------------------------------*/ - - -int lpc_to_lsp (float *a,int lpcrdr,float *freq,int nb,float delta, char *stack) -/* float *a lpc coefficients */ -/* int lpcrdr order of LPC coefficients (10) */ -/* float *freq LSP frequencies in the x domain */ -/* int nb number of sub-intervals (4) */ -/* float delta grid spacing interval (0.02) */ - - -{ - - float psuml,psumr,psumm,temp_xr,xl,xr,xm=0; - float temp_psumr/*,temp_qsumr*/; - int i,j,m,flag,k; - float *Q; /* ptrs for memory allocation */ - float *P; - float *px; /* ptrs of respective P'(z) & Q'(z) */ - float *qx; - float *p; - float *q; - float *pt; /* ptr used for cheb_poly_eval() - whether P' or Q' */ - int roots=0; /* DR 8/2/94: number of roots found */ - flag = 1; /* program is searching for a root when, - 1 else has found one */ - m = lpcrdr/2; /* order of P'(z) & Q'(z) polynomials */ - - - /* Allocate memory space for polynomials */ - Q = PUSH(stack, (m+1), float); - P = PUSH(stack, (m+1), float); - - /* determine P'(z)'s and Q'(z)'s coefficients where - P'(z) = P(z)/(1 + z^(-1)) and Q'(z) = Q(z)/(1-z^(-1)) */ - - px = P; /* initialise ptrs */ - qx = Q; - p = px; - q = qx; - *px++ = 1.0; - *qx++ = 1.0; - for(i=1;i<=m;i++){ - *px++ = a[i]+a[lpcrdr+1-i]-*p++; - *qx++ = a[i]-a[lpcrdr+1-i]+*q++; - } - px = P; - qx = Q; - for(i=0;i= -1.0)){ - float dd; - /* Modified by JMV to provide smaller steps around x=+-1 */ - dd=(delta*(1-.9*xl*xl)); - if (fabs(psuml)<.2) - dd *= .5; - - xr = xl - dd; /* interval spacing */ - psumr = cheb_poly_eva(pt,xr,lpcrdr,stack);/* poly(xl-delta_x) */ - temp_psumr = psumr; - temp_xr = xr; - - /* if no sign change increment xr and re-evaluate poly(xr). Repeat til - sign change. - if a sign change has occurred the interval is bisected and then - checked again for a sign change which determines in which - interval the zero lies in. - If there is no sign change between poly(xm) and poly(xl) set interval - between xm and xr else set interval between xl and xr and repeat till - root is located within the specified limits */ - - if((psumr*psuml)<0.0){ - roots++; - - psumm=psuml; - for(k=0;k<=nb;k++){ - xm = (xl+xr)/2; /* bisect the interval */ - psumm=cheb_poly_eva(pt,xm,lpcrdr,stack); - if(psumm*psuml>0.){ - psuml=psumm; - xl=xm; - } - else{ - psumr=psumm; - xr=xm; - } - } - - /* once zero is found, reset initial interval to xr */ - freq[j] = (xm); - xl = xm; - flag = 0; /* reset flag for next search */ - } - else{ - psuml=temp_psumr; - xl=temp_xr; - } - } - } - return(roots); -} - - -/*---------------------------------------------------------------------------*\ - - FUNCTION....: lsp_to_lpc() - - AUTHOR......: David Rowe - DATE CREATED: 24/2/93 - - lsp_to_lpc: This function converts LSP coefficients to LPC - coefficients. - -\*---------------------------------------------------------------------------*/ - - -void lsp_to_lpc(float *freq,float *ak,int lpcrdr, char *stack) -/* float *freq array of LSP frequencies in the x domain */ -/* float *ak array of LPC coefficients */ -/* int lpcrdr order of LPC coefficients */ - - -{ - int i,j; - float xout1,xout2,xin1,xin2; - float *Wp; - float *pw,*n1,*n2,*n3,*n4=NULL; - int m = lpcrdr/2; - - Wp = PUSH(stack, 4*m+2, float); - pw = Wp; - - /* initialise contents of array */ - - for(i=0;i<=4*m+1;i++){ /* set contents of buffer to 0 */ - *pw++ = 0.0; - } - - /* Set pointers up */ - - pw = Wp; - xin1 = 1.0; - xin2 = 1.0; - - /* reconstruct P(z) and Q(z) by cascading second order - polynomials in form 1 - 2xz(-1) +z(-2), where x is the - LSP coefficient */ - - for(j=0;j<=lpcrdr;j++){ - int i2=0; - for(i=0;iM_PI-margin) - lsp[len-1]=M_PI-margin; - for (i=1;ilsp[i+1]-margin) - lsp[i]= .5* (lsp[i] + lsp[i+1]-margin); - } -} diff --git a/engine/voice_codecs/speex/source/libspeex/lsp.h b/engine/voice_codecs/speex/source/libspeex/lsp.h deleted file mode 100644 index 25404954..00000000 --- a/engine/voice_codecs/speex/source/libspeex/lsp.h +++ /dev/null @@ -1,57 +0,0 @@ -/*---------------------------------------------------------------------------*\ -Original Copyright - FILE........: AK2LSPD.H - TYPE........: Turbo C header file - COMPANY.....: Voicetronix - AUTHOR......: James Whitehall - DATE CREATED: 21/11/95 - -Modified by Jean-Marc Valin - - This file contains functions for converting Linear Prediction - Coefficients (LPC) to Line Spectral Pair (LSP) and back. Note that the - LSP coefficients are not in radians format but in the x domain of the - unit circle. - -\*---------------------------------------------------------------------------*/ -/* Speex License: - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef __AK2LSPD__ -#define __AK2LSPD__ - -int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta, char *stack); -void lsp_to_lpc(float *freq, float *ak, int lpcrdr, char *stack); - -/*Added by JMV*/ -void lsp_enforce_margin(float *lsp, int len, float margin); - - -#endif /* __AK2LSPD__ */ diff --git a/engine/voice_codecs/speex/source/libspeex/lsp_tables_nb.c b/engine/voice_codecs/speex/source/libspeex/lsp_tables_nb.c deleted file mode 100644 index a68ce994..00000000 --- a/engine/voice_codecs/speex/source/libspeex/lsp_tables_nb.c +++ /dev/null @@ -1,360 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: lsp_tables_nb.c - Codebooks for LSPs in narrowband CELP mode - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions are - met: - - 1. Redistributions of source code must retain the above copyright notice, - this list of conditions and the following disclaimer. - - 2. Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - 3. The name of the author may not be used to endorse or promote products - derived from this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR - IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES - OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE - DISCLAIMED. IN NO EVENT SHALL THE AUTHOR 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. -*/ - -signed char cdbk_nb[640]={ -30,19,38,34,40,32,46,43,58,43, -5,-18,-25,-40,-33,-55,-52,20,34,28, --20,-63,-97,-92,61,53,47,49,53,75, --14,-53,-77,-79,0,-3,-5,19,22,26, --9,-53,-55,66,90,72,85,68,74,52, --4,-41,-58,-31,-18,-31,27,32,30,18, -24,3,8,5,-12,-3,26,28,74,63, --2,-39,-67,-77,-106,-74,59,59,73,65, -44,40,71,72,82,83,98,88,89,60, --6,-31,-47,-48,-13,-39,-9,7,2,79, --1,-39,-60,-17,87,81,65,50,45,19, --21,-67,-91,-87,-41,-50,7,18,39,74, -10,-31,-28,39,24,13,23,5,56,45, -29,10,-5,-13,-11,-35,-18,-8,-10,-8, --25,-71,-77,-21,2,16,50,63,87,87, -5,-32,-40,-51,-68,0,12,6,54,34, -5,-12,32,52,68,64,69,59,65,45, -14,-16,-31,-40,-65,-67,41,49,47,37, --11,-52,-75,-84,-4,57,48,42,42,33, --11,-51,-68,-6,13,0,8,-8,26,32, --23,-53,0,36,56,76,97,105,111,97, --1,-28,-39,-40,-43,-54,-44,-40,-18,35, -16,-20,-19,-28,-42,29,47,38,74,45, -3,-29,-48,-62,-80,-104,-33,56,59,59, -10,17,46,72,84,101,117,123,123,106, --7,-33,-49,-51,-70,-67,-27,-31,70,67, --16,-62,-85,-20,82,71,86,80,85,74, --19,-58,-75,-45,-29,-33,-18,-25,45,57, --12,-42,-5,12,28,36,52,64,81,82, -13,-9,-27,-28,22,3,2,22,26,6, --6,-44,-51,2,15,10,48,43,49,34, --19,-62,-84,-89,-102,-24,8,17,61,68, -39,24,23,19,16,-5,12,15,27,15, --8,-44,-49,-60,-18,-32,-28,52,54,62, --8,-48,-77,-70,66,101,83,63,61,37, --12,-50,-75,-64,33,17,13,25,15,77, -1,-42,-29,72,64,46,49,31,61,44, --8,-47,-54,-46,-30,19,20,-1,-16,0, -16,-12,-18,-9,-26,-27,-10,-22,53,45, --10,-47,-75,-82,-105,-109,8,25,49,77, -50,65,114,117,124,118,115,96,90,61, --9,-45,-63,-60,-75,-57,8,11,20,29, -0,-35,-49,-43,40,47,35,40,55,38, --24,-76,-103,-112,-27,3,23,34,52,75, -8,-29,-43,12,63,38,35,29,24,8, -25,11,1,-15,-18,-43,-7,37,40,21, --20,-56,-19,-19,-4,-2,11,29,51,63, --2,-44,-62,-75,-89,30,57,51,74,51, -50,46,68,64,65,52,63,55,65,43, -18,-9,-26,-35,-55,-69,3,6,8,17, --15,-61,-86,-97,1,86,93,74,78,67, --1,-38,-66,-48,48,39,29,25,17,-1, -13,13,29,39,50,51,69,82,97,98, --2,-36,-46,-27,-16,-30,-13,-4,-7,-4, -25,-5,-11,-6,-25,-21,33,12,31,29, --8,-38,-52,-63,-68,-89,-33,-1,10,74, --2,-15,59,91,105,105,101,87,84,62, --7,-33,-50,-35,-54,-47,25,17,82,81, --13,-56,-83,21,58,31,42,25,72,65, --24,-66,-91,-56,9,-2,21,10,69,75, -2,-24,11,22,25,28,38,34,48,33, -7,-29,-26,17,15,-1,14,0,-2,0, --6,-41,-67,6,-2,-9,19,2,85,74, --22,-67,-84,-71,-50,3,11,-9,2,62}; - -signed char cdbk_nb_low1[320]={ --34,-52,-15,45,2, -23,21,52,24,-33, --9,-1,9,-44,-41, --13,-17,44,22,-17, --6,-4,-1,22,38, -26,16,2,50,27, --35,-34,-9,-41,6, -0,-16,-34,51,8, --14,-31,-49,15,-33, -45,49,33,-11,-37, --62,-54,45,11,-5, --72,11,-1,-12,-11, -24,27,-11,-43,46, -43,33,-12,-9,-1, -1,-4,-23,-57,-71, -11,8,16,17,-8, --20,-31,-41,53,48, --16,3,65,-24,-8, --23,-32,-37,-32,-49, --10,-17,6,38,5, --9,-17,-46,8,52, -3,6,45,40,39, --7,-6,-34,-74,31, -8,1,-16,43,68, --11,-19,-31,4,6, -0,-6,-17,-16,-38, --16,-30,2,9,-39, --16,-1,43,-10,48, -3,3,-16,-31,-3, -62,68,43,13,3, --10,8,20,-56,12, -12,-2,-18,22,-15, --40,-36,1,7,41, -0,1,46,-6,-62, --4,-12,-2,-11,-83, --13,-2,91,33,-10, -0,4,-11,-16,79, -32,37,14,9,51, --21,-28,-56,-34,0, -21,9,-26,11,28, --42,-54,-23,-2,-15, -31,30,8,-39,-66, --39,-36,31,-28,-40, --46,35,40,22,24, -33,48,23,-34,14, -40,32,17,27,-3, -25,26,-13,-61,-17, -11,4,31,60,-6, --26,-41,-64,13,16, --26,54,31,-11,-23, --9,-11,-34,-71,-21, --34,-35,55,50,29, --22,-27,-50,-38,57, -33,42,57,48,26, -11,0,-49,-31,26, --4,-14,5,78,37, -17,0,-49,-12,-23, -26,14,2,2,-43, --17,-12,10,-8,-4, -8,18,12,-6,20, --12,-6,-13,-25,34, -15,40,49,7,8, -13,20,20,-19,-22, --2,-8,2,51,-51}; - -signed char cdbk_nb_low2[320]={ --6,53,-21,-24,4, -26,17,-4,-37,25, -17,-36,-13,31,3, --6,27,15,-10,31, -28,26,-10,-10,-40, -16,-7,15,13,41, --9,0,-4,50,-6, --7,14,38,22,0, --48,2,1,-13,-19, -32,-3,-60,11,-17, --1,-24,-34,-1,35, --5,-27,28,44,13, -25,15,42,-11,15, -51,35,-36,20,8, --4,-12,-29,19,-47, -49,-15,-4,16,-29, --39,14,-30,4,25, --9,-5,-51,-14,-3, --40,-32,38,5,-9, --8,-4,-1,-22,71, --3,14,26,-18,-22, -24,-41,-25,-24,6, -23,19,-10,39,-26, --27,65,45,2,-7, --26,-8,22,-12,16, -15,16,-35,-5,33, --21,-8,0,23,33, -34,6,21,36,6, --7,-22,8,-37,-14, -31,38,11,-4,-3, --39,-32,-8,32,-23, --6,-12,16,20,-28, --4,23,13,-52,-1, -22,6,-33,-40,-6, -4,-62,13,5,-26, -35,39,11,2,57, --11,9,-20,-28,-33, -52,-5,-6,-2,22, --14,-16,-48,35,1, --58,20,13,33,-1, --74,56,-18,-22,-31, -12,6,-14,4,-2, --9,-47,10,-3,29, --17,-5,61,14,47, --12,2,72,-39,-17, -92,64,-53,-51,-15, --30,-38,-41,-29,-28, -27,9,36,9,-35, --42,81,-21,20,25, --16,-5,-17,-35,21, -15,-28,48,2,-2, -9,-19,29,-40,30, --18,-18,18,-16,-57, -15,-20,-12,-15,-37, --15,33,-39,21,-22, --13,35,11,13,-38, --63,29,23,-27,32, -18,3,-26,42,33, --64,-66,-17,16,56, -2,36,3,31,21, --41,-39,8,-57,14, -37,-2,19,-36,-19, --23,-29,-16,1,-3, --8,-10,31,64,-65}; - -signed char cdbk_nb_high1[320]={ --26,-8,29,21,4, -19,-39,33,-7,-36, -56,54,48,40,29, --4,-24,-42,-66,-43, --60,19,-2,37,41, --10,-37,-60,-64,18, --22,77,73,40,25, -4,19,-19,-66,-2, -11,5,21,14,26, --25,-86,-4,18,1, -26,-37,10,37,-1, -24,-12,-59,-11,20, --6,34,-16,-16,42, -19,-28,-51,53,32, -4,10,62,21,-12, --34,27,4,-48,-48, --50,-49,31,-7,-21, --42,-25,-4,-43,-22, -59,2,27,12,-9, --6,-16,-8,-32,-58, --16,-29,-5,41,23, --30,-33,-46,-13,-10, --38,52,52,1,-17, --9,10,26,-25,-6, -33,-20,53,55,25, --32,-5,-42,23,21, -66,5,-28,20,9, -75,29,-7,-42,-39, -15,3,-23,21,6, -11,1,-29,14,63, -10,54,26,-24,-51, --49,7,-23,-51,15, --66,1,60,25,10, -0,-30,-4,-15,17, -19,59,40,4,-5, -33,6,-22,-58,-70, --5,23,-6,60,44, --29,-16,-47,-29,52, --19,50,28,16,35, -31,36,0,-21,6, -21,27,22,42,7, --66,-40,-8,7,19, -46,0,-4,60,36, -45,-7,-29,-6,-32, --39,2,6,-9,33, -20,-51,-34,18,-6, -19,6,11,5,-19, --29,-2,42,-11,-45, --21,-55,57,37,2, --14,-67,-16,-27,-38, -69,48,19,2,-17, -20,-20,-16,-34,-17, --25,-61,10,73,45, -16,-40,-64,-17,-29, --22,56,17,-39,8, --11,8,-25,-18,-13, --19,8,54,57,36, --17,-26,-4,6,-21, -40,42,-4,20,31, -53,10,-34,-53,31, --17,35,0,15,-6, --20,-63,-73,22,25, -29,17,8,-29,-39, --69,18,15,-15,-5}; - -signed char cdbk_nb_high2[320]={ -11,47,16,-9,-46, --32,26,-64,34,-5, -38,-7,47,20,2, --73,-99,-3,-45,20, -70,-52,15,-6,-7, --82,31,21,47,51, -39,-3,9,0,-41, --7,-15,-54,2,0, -27,-31,9,-45,-22, --38,-24,-24,8,-33, -23,5,50,-36,-17, --18,-51,-2,13,19, -43,12,-15,-12,61, -38,38,7,13,0, -6,-1,3,62,9, -27,22,-33,38,-35, --9,30,-43,-9,-32, --1,4,-4,1,-5, --11,-8,38,31,11, --10,-42,-21,-37,1, -43,15,-13,-35,-19, --18,15,23,-26,59, -1,-21,53,8,-41, --50,-14,-28,4,21, -25,-28,-40,5,-40, --41,4,51,-33,-8, --8,1,17,-60,12, -25,-41,17,34,43, -19,45,7,-37,24, --15,56,-2,35,-10, -48,4,-47,-2,5, --5,-54,5,-3,-33, --10,30,-2,-44,-24, --38,9,-9,42,4, -6,-56,44,-16,9, --40,-26,18,-20,10, -28,-41,-21,-4,13, --18,32,-30,-3,37, -15,22,28,50,-40, -3,-29,-64,7,51, --19,-11,17,-27,-40, --64,24,-12,-7,-27, -3,37,48,-1,2, --9,-38,-34,46,1, -27,-6,19,-13,26, -10,34,20,25,40, -50,-6,-7,30,9, --24,0,-23,71,-61, -22,58,-34,-4,2, --49,-33,25,30,-8, --6,-16,77,2,38, --8,-35,-6,-30,56, -78,31,33,-20,13, --39,20,22,4,21, --8,4,-6,10,-83, --41,9,-25,-43,15, --7,-12,-34,-39,-37, --33,19,30,16,-33, -42,-25,25,-68,44, --15,-11,-4,23,50, -14,4,-39,-43,20, --30,60,9,-20,7, -16,19,-33,37,29, -16,-35,7,38,-27}; diff --git a/engine/voice_codecs/speex/source/libspeex/ltp.c b/engine/voice_codecs/speex/source/libspeex/ltp.c deleted file mode 100644 index f95d2152..00000000 --- a/engine/voice_codecs/speex/source/libspeex/ltp.c +++ /dev/null @@ -1,548 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: ltp.c - Long-Term Prediction functions - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include -#include "ltp.h" -#include "stack_alloc.h" -#include "filters.h" -#include "speex_bits.h" - -#ifdef _USE_SSE -#include "ltp_sse.h" -#else -static float inner_prod(float *x, float *y, int len) -{ - int i; - float sum1=0,sum2=0,sum3=0,sum4=0; - for (i=0;ibest_score[N-1]) - { - float g1, g; - g1 = corr[i-start]/(energy[i-start]+10); - g = sqrt(g1*corr[i-start]/(e0+10)); - if (g>g1) - g=g1; - if (g<0) - g=0; - for (j=0;j best_score[j]) - { - for (k=N-1;k>j;k--) - { - best_score[k]=best_score[k-1]; - pitch[k]=pitch[k-1]; - gain[k] = gain[k-1]; - } - best_score[j]=score[i-start]; - pitch[j]=i; - gain[j]=g; - break; - } - } - } - } - -} - - - - -/** Finds the best quantized 3-tap pitch predictor by analysis by synthesis */ -float pitch_gain_search_3tap( -float target[], /* Target vector */ -float ak[], /* LPCs for this subframe */ -float awk1[], /* Weighted LPCs #1 for this subframe */ -float awk2[], /* Weighted LPCs #2 for this subframe */ -float exc[], /* Excitation */ -void *par, -int pitch, /* Pitch value */ -int p, /* Number of LPC coeffs */ -int nsf, /* Number of samples in subframe */ -SpeexBits *bits, -char *stack, -float *exc2, -float *r, -int *cdbk_index -) -{ - int i,j; - float *tmp, *tmp2; - float *x[3]; - float *e[3]; - float corr[3]; - float A[3][3]; - float gain[3]; - int gain_cdbk_size; - signed char *gain_cdbk; - float err1,err2; - - ltp_params *params; - params = (ltp_params*) par; - gain_cdbk=params->gain_cdbk; - gain_cdbk_size=1<gain_bits; - tmp = PUSH(stack, 3*nsf, float); - tmp2 = PUSH(stack, 3*nsf, float); - - x[0]=tmp; - x[1]=tmp+nsf; - x[2]=tmp+2*nsf; - - e[0]=tmp2; - e[1]=tmp2+nsf; - e[2]=tmp2+2*nsf; - - for (i=2;i>=0;i--) - { - int pp=pitch+1-i; - for (j=0;j0) - tot+=ptr[0]; - if (ptr[2]>0) - tot+=ptr[2]; - if (tot>1) - continue; - } - - if (sum>best_sum || i==0) - { - best_sum=sum; - best_cdbk=i; - } - } - gain[0] = 0.015625*gain_cdbk[best_cdbk*3] + .5; - gain[1] = 0.015625*gain_cdbk[best_cdbk*3+1]+ .5; - gain[2] = 0.015625*gain_cdbk[best_cdbk*3+2]+ .5; - - *cdbk_index=best_cdbk; - } - - for (i=0;i10) - N=10; - - nbest=PUSH(stack, N, int); - gains = PUSH(stack, N, float); - params = (ltp_params*) par; - - if (N==0 || endpitch_bits); - speex_bits_pack(bits, 0, params->gain_bits); - for (i=0;iend-start+1) - N=end-start+1; - open_loop_nbest_pitch(sw, start, end, nsf, nbest, gains, N, stack); - for (i=0;ipitch_bits); - speex_bits_pack(bits, best_gain_index, params->gain_bits); - /*printf ("encode pitch: %d %d\n", best_pitch, best_gain_index);*/ - for (i=0;igain_cdbk; - - pitch = speex_bits_unpack_unsigned(bits, params->pitch_bits); - pitch += start; - gain_index = speex_bits_unpack_unsigned(bits, params->gain_bits); - /*printf ("decode pitch: %d %d\n", pitch, gain_index);*/ - gain[0] = 0.015625*gain_cdbk[gain_index*3]+.5; - gain[1] = 0.015625*gain_cdbk[gain_index*3+1]+.5; - gain[2] = 0.015625*gain_cdbk[gain_index*3+2]+.5; - - if (count_lost && pitch > subframe_offset) - { - float gain_sum; - - if (1) { - float tmp = count_lost < 4 ? last_pitch_gain : 0.4 * last_pitch_gain; - if (tmp>.95) - tmp=.95; - gain_sum = fabs(gain[1]); - if (gain[0]>0) - gain_sum += gain[0]; - else - gain_sum -= .5*gain[0]; - if (gain[2]>0) - gain_sum += gain[2]; - else - gain_sum -= .5*gain[2]; - if (gain_sum > tmp) { - float fact = tmp/gain_sum; - for (i=0;i<3;i++) - gain[i]*=fact; - - } - - } - - if (0) { - gain_sum = fabs(gain[0])+fabs(gain[1])+fabs(gain[2]); - if (gain_sum>.95) { - float fact = .95/gain_sum; - for (i=0;i<3;i++) - gain[i]*=fact; - } - } - } - - *pitch_val = pitch; - /**gain_val = gain[0]+gain[1]+gain[2];*/ - gain_val[0]=gain[0]; - gain_val[1]=gain[1]; - gain_val[2]=gain[2]; - - { - float *e[3]; - float *tmp2; - tmp2=PUSH(stack, 3*nsf, float); - e[0]=tmp2; - e[1]=tmp2+nsf; - e[2]=tmp2+2*nsf; - - for (i=0;i<3;i++) - { - int j; - int pp=pitch+1-i; -#if 0 - for (j=0;jpp) - tmp1=pp; - for (j=0;jpp+pitch) - tmp3=pp+pitch; - for (j=tmp1;j.99) - pitch_coef=.99; - for (i=0;i.99) - pitch_coef=.99; - for (i=0;i -#include "math_approx.h" - -#ifdef SLOW_TRIG - -float cos_sin[102] = { - 1.00000000, 0.00000000, - 0.99804751, 0.06245932, - 0.99219767, 0.12467473, - 0.98247331, 0.18640330, - 0.96891242, 0.24740396, - 0.95156795, 0.30743851, - 0.93050762, 0.36627253, - 0.90581368, 0.42367626, - 0.87758256, 0.47942554, - 0.84592450, 0.53330267, - 0.81096312, 0.58509727, - 0.77283495, 0.63460708, - 0.73168887, 0.68163876, - 0.68768556, 0.72600866, - 0.64099686, 0.76754350, - 0.59180508, 0.80608111, - 0.54030231, 0.84147098, - 0.48668967, 0.87357494, - 0.43117652, 0.90226759, - 0.37397963, 0.92743692, - 0.31532236, 0.94898462, - 0.25543377, 0.96682656, - 0.19454771, 0.98089306, - 0.13290194, 0.99112919, - 0.07073720, 0.99749499, - 0.00829623, 0.99996559, - -0.05417714, 0.99853134, - -0.11643894, 0.99319785, - -0.17824606, 0.98398595, - -0.23935712, 0.97093160, - -0.29953351, 0.95408578, - -0.35854022, 0.93351428, - -0.41614684, 0.90929743, - -0.47212841, 0.88152979, - -0.52626633, 0.85031979, - -0.57834920, 0.81578931, - -0.62817362, 0.77807320, - -0.67554504, 0.73731872, - -0.72027847, 0.69368503, - -0.76219923, 0.64734252, - -0.80114362, 0.59847214, - -0.83695955, 0.54726475, - -0.86950718, 0.49392030, - -0.89865940, 0.43864710, - -0.92430238, 0.38166099, - -0.94633597, 0.32318451, - -0.96467415, 0.26344599, - -0.97924529, 0.20267873, - -0.98999250, 0.14112001, - -0.99687381, 0.07901022, - -0.99986235, 0.01659189 -}; - -float speex_cos(float x) -{ - int ind; - float delta; - ind = (int)floor(x*16+.5); - delta = x-0.062500*ind; - ind <<= 1; - return cos_sin[ind] - delta*(cos_sin[ind+1] + - .5*delta*(cos_sin[ind] - - .3333333*delta*cos_sin[ind+1])); -} - -#endif - diff --git a/engine/voice_codecs/speex/source/libspeex/math_approx.h b/engine/voice_codecs/speex/source/libspeex/math_approx.h deleted file mode 100644 index c917ffd7..00000000 --- a/engine/voice_codecs/speex/source/libspeex/math_approx.h +++ /dev/null @@ -1,39 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: math_approx.c - Various math approximation functions for Speex - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef MATH_APPROX_H -#define MATH_APPROX_H - -float speex_cos(float x); - - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/misc.c b/engine/voice_codecs/speex/source/libspeex/misc.c deleted file mode 100644 index f84e0f61..00000000 --- a/engine/voice_codecs/speex/source/libspeex/misc.c +++ /dev/null @@ -1,145 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: mics.c - Various utility routines for Speex - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include "misc.h" -#include -#include -#include - -#ifndef RELEASE -void print_vec(float *vec, int len, char *name) -{ - int i; - printf ("%s ", name); - for (i=0;i>24; - ret += (i>>8)&0x0000ff00; - ret += (i<<8)&0x00ff0000; - ret += (i<<24); -#endif - return ret; -} - -unsigned int le_int(unsigned int i) -{ - unsigned int ret=i; -#ifdef WORDS_BIGENDIAN - ret = i>>24; - ret += (i>>8)&0x0000ff00; - ret += (i<<8)&0x00ff0000; - ret += (i<<24); -#endif - return ret; -} - -unsigned short be_short(unsigned short s) -{ - unsigned short ret=s; -#ifndef WORDS_BIGENDIAN - ret = s>>8; - ret += s<<8; -#endif - return ret; -} - -unsigned short le_short(unsigned short s) -{ - unsigned short ret=s; -#ifdef WORDS_BIGENDIAN - ret = s>>8; - ret += s<<8; -#endif - return ret; -} - -void *speex_alloc (int size) -{ - return calloc(size,1); -} - -void *speex_realloc (void *ptr, int size) -{ - return realloc(ptr, size); -} - -void speex_free (void *ptr) -{ - free(ptr); -} - -void *speex_move (void *dest, void *src, int n) -{ - return memmove(dest,src,n); -} - -void speex_error(char *str) -{ - fprintf (stderr, "Fatal error: %s\n", str); - exit(1); -} - -void speex_warning(char *str) -{ - fprintf (stderr, "warning: %s\n", str); -} - -void speex_warning_int(char *str, int val) -{ - fprintf (stderr, "warning: %s %d\n", str, val); -} - -void speex_rand_vec(float std, float *data, int len) -{ - int i; - for (i=0;ienc_init(mode); -} - -void *speex_decoder_init(SpeexMode *mode) -{ - return mode->dec_init(mode); -} - -void speex_encoder_destroy(void *state) -{ - (*((SpeexMode**)state))->enc_destroy(state); -} - -int speex_encode(void *state, float *in, SpeexBits *bits) -{ - return (*((SpeexMode**)state))->enc(state, in, bits); -} - -void speex_decoder_destroy(void *state) -{ - (*((SpeexMode**)state))->dec_destroy(state); -} - -int speex_decode(void *state, SpeexBits *bits, float *out) -{ - return (*((SpeexMode**)state))->dec(state, bits, out); -} - - -int speex_encoder_ctl(void *state, int request, void *ptr) -{ - return (*((SpeexMode**)state))->enc_ctl(state, request, ptr); -} - -int speex_decoder_ctl(void *state, int request, void *ptr) -{ - return (*((SpeexMode**)state))->dec_ctl(state, request, ptr); -} - - - -static int nb_mode_query(void *mode, int request, void *ptr) -{ - SpeexNBMode *m = (SpeexNBMode*)mode; - - switch (request) - { - case SPEEX_MODE_FRAME_SIZE: - *((int*)ptr)=m->frameSize; - break; - case SPEEX_SUBMODE_BITS_PER_FRAME: - if (*((int*)ptr)==0) - *((int*)ptr) = NB_SUBMODE_BITS+1; - else if (m->submodes[*((int*)ptr)]==NULL) - *((int*)ptr) = -1; - else - *((int*)ptr) = m->submodes[*((int*)ptr)]->bits_per_frame; - break; - default: - speex_warning_int("Unknown nb_mode_query request: ", request); - return -1; - } - return 0; -} - -static int wb_mode_query(void *mode, int request, void *ptr) -{ - SpeexSBMode *m = (SpeexSBMode*)mode; - - switch (request) - { - case SPEEX_MODE_FRAME_SIZE: - *((int*)ptr)=2*m->frameSize; - break; - case SPEEX_SUBMODE_BITS_PER_FRAME: - if (*((int*)ptr)==0) - *((int*)ptr) = SB_SUBMODE_BITS+1; - else if (m->submodes[*((int*)ptr)]==NULL) - *((int*)ptr) = -1; - else - *((int*)ptr) = m->submodes[*((int*)ptr)]->bits_per_frame; - break; - default: - speex_warning_int("Unknown wb_mode_query request: ", request); - return -1; - } - return 0; -} - - -int speex_mode_query(SpeexMode *mode, int request, void *ptr) -{ - return mode->query(mode->mode, request, ptr); -} diff --git a/engine/voice_codecs/speex/source/libspeex/modes.h b/engine/voice_codecs/speex/source/libspeex/modes.h deleted file mode 100644 index 0a94357c..00000000 --- a/engine/voice_codecs/speex/source/libspeex/modes.h +++ /dev/null @@ -1,146 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin */ -/** - @file modes.h - @brief Describes the different modes of the codec -*/ -/* - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - -#ifndef MODES_H -#define MODES_H - -#include "speex.h" -#include "speex_bits.h" - - -#define NB_SUBMODES 16 -#define NB_SUBMODE_BITS 4 - -#define SB_SUBMODES 8 -#define SB_SUBMODE_BITS 3 - - -/** Quantizes LSPs */ -typedef void (*lsp_quant_func)(float *, float *, int, SpeexBits *); - -/** Decodes quantized LSPs */ -typedef void (*lsp_unquant_func)(float *, int, SpeexBits *); - - -/** Long-term predictor quantization */ -typedef int (*ltp_quant_func)(float *, float *, float *, float *, - float *, float *, void *, int, int, float, - int, int, SpeexBits*, char *, float *, float *, int); - -/** Long-term un-quantize */ -typedef void (*ltp_unquant_func)(float *, int, int, float, void *, int, int *, - float *, SpeexBits*, char*, int, int, float); - - -/** Innovation quantization function */ -typedef void (*innovation_quant_func)(float *, float *, float *, float *, void *, int, int, - float *, float *, SpeexBits *, char *, int); - -/** Innovation unquantization function */ -typedef void (*innovation_unquant_func)(float *, void *, int, SpeexBits*, char *); - -/** Description of a Speex sub-mode (wither narrowband or wideband */ -typedef struct SpeexSubmode { - int lbr_pitch; /**< Set to -1 for "normal" modes, otherwise encode pitch using a global pitch and allowing a +- lbr_pitch variation (for low not-rates)*/ - int forced_pitch_gain; /**< Use the same (forced) pitch gain for all sub-frames */ - int have_subframe_gain; /**< Number of bits to use as sub-frame innovation gain */ - int double_codebook; /**< Apply innovation quantization twice for higher quality (and higher bit-rate)*/ - /*LSP functions*/ - lsp_quant_func lsp_quant; /**< LSP quantization function */ - lsp_unquant_func lsp_unquant; /**< LSP unquantization function */ - - /*Lont-term predictor functions*/ - ltp_quant_func ltp_quant; /**< Long-term predictor (pitch) quantizer */ - ltp_unquant_func ltp_unquant; /**< Long-term predictor (pitch) un-quantizer */ - void *ltp_params; /**< Pitch parameters (options) */ - - /*Quantization of innovation*/ - innovation_quant_func innovation_quant; /**< Innovation quantization */ - innovation_unquant_func innovation_unquant; /**< Innovation un-quantization */ - void *innovation_params; /**< Innovation quantization parameters*/ - - /*Synthesis filter enhancement*/ - float lpc_enh_k1; /**< Enhancer constant */ - float lpc_enh_k2; /**< Enhancer constant */ - float comb_gain; /**< Gain of enhancer comb filter */ - - int bits_per_frame; /**< Number of bits per frame after encoding*/ -} SpeexSubmode; - -/** Struct defining the encoding/decoding mode*/ -typedef struct SpeexNBMode { - int frameSize; /**< Size of frames used for encoding */ - int subframeSize; /**< Size of sub-frames used for encoding */ - int lpcSize; /**< Order of LPC filter */ - int bufSize; /**< Size of signal buffer to use in encoder */ - int pitchStart; /**< Smallest pitch value allowed */ - int pitchEnd; /**< Largest pitch value allowed */ - - float gamma1; /**< Perceptual filter parameter #1 */ - float gamma2; /**< Perceptual filter parameter #2 */ - float lag_factor; /**< Lag-windowing parameter */ - float lpc_floor; /**< Noise floor for LPC analysis */ - float preemph; /**< Pre-emphasis */ - - SpeexSubmode *submodes[NB_SUBMODES]; /**< Sub-mode data for the mode */ - int defaultSubmode; /**< Default sub-mode to use when encoding */ - int quality_map[11]; /**< Mode corresponding to each quality setting */ -} SpeexNBMode; - - -/** Struct defining the encoding/decoding mode for SB-CELP (wideband) */ -typedef struct SpeexSBMode { - SpeexMode *nb_mode; /**< Embedded narrowband mode */ - int frameSize; /**< Size of frames used for encoding */ - int subframeSize; /**< Size of sub-frames used for encoding */ - int lpcSize; /**< Order of LPC filter */ - int bufSize; /**< Signal buffer size in encoder */ - float gamma1; /**< Perceptual filter parameter #1 */ - float gamma2; /**< Perceptual filter parameter #1 */ - float lag_factor; /**< Lag-windowing parameter */ - float lpc_floor; /**< Noise floor for LPC analysis */ - float preemph; /**< Pre-emphasis */ - float folding_gain; - - SpeexSubmode *submodes[SB_SUBMODES]; /**< Sub-mode data for the mode */ - int defaultSubmode; /**< Default sub-mode to use when encoding */ - int low_quality_map[11]; /**< Mode corresponding to each quality setting */ - int quality_map[11]; /**< Mode corresponding to each quality setting */ - float (*vbr_thresh)[11]; - int nb_modes; -} SpeexSBMode; - - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/nb_celp.c b/engine/voice_codecs/speex/source/libspeex/nb_celp.c deleted file mode 100644 index 837d2f5d..00000000 --- a/engine/voice_codecs/speex/source/libspeex/nb_celp.c +++ /dev/null @@ -1,1715 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: nb_celp.c - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include -#include "nb_celp.h" -#include "lpc.h" -#include "lsp.h" -#include "ltp.h" -#include "quant_lsp.h" -#include "cb_search.h" -#include "filters.h" -#include "stack_alloc.h" -#include "vq.h" -#include "speex_bits.h" -#include "vbr.h" -#include "misc.h" -#include "speex_callbacks.h" - -#ifdef SLOW_TRIG -#include "math_approx.h" -#define cos speex_cos -#endif - -#ifndef M_PI -#define M_PI 3.14159265358979323846 /* pi */ -#endif - -#ifndef NULL -#define NULL 0 -#endif - -#define SUBMODE(x) st->submodes[st->submodeID]->x - -float exc_gain_quant_scal3[8]={-2.794750, -1.810660, -1.169850, -0.848119, -0.587190, -0.329818, -0.063266, 0.282826}; - -float exc_gain_quant_scal1[2]={-0.35, 0.05}; - -#define sqr(x) ((x)*(x)) - -void *nb_encoder_init(SpeexMode *m) -{ - EncState *st; - SpeexNBMode *mode; - int i; - - mode=(SpeexNBMode *)m->mode; - st = (EncState*)speex_alloc(sizeof(EncState)+8000*sizeof(float)); - if (!st) - return NULL; - - st->stack = ((char*)st) + sizeof(EncState); - - st->mode=m; - - st->frameSize = mode->frameSize; - st->windowSize = st->frameSize*3/2; - st->nbSubframes=mode->frameSize/mode->subframeSize; - st->subframeSize=mode->subframeSize; - st->lpcSize = mode->lpcSize; - st->bufSize = mode->bufSize; - st->gamma1=mode->gamma1; - st->gamma2=mode->gamma2; - st->min_pitch=mode->pitchStart; - st->max_pitch=mode->pitchEnd; - st->lag_factor=mode->lag_factor; - st->lpc_floor = mode->lpc_floor; - st->preemph = mode->preemph; - - st->submodes=mode->submodes; - st->submodeID=st->submodeSelect=mode->defaultSubmode; - st->pre_mem=0; - st->pre_mem2=0; - st->bounded_pitch = 1; - - /* Allocating input buffer */ - st->inBuf = PUSH(st->stack, st->bufSize, float); - st->frame = st->inBuf + st->bufSize - st->windowSize; - /* Allocating excitation buffer */ - st->excBuf = PUSH(st->stack, st->bufSize, float); - st->exc = st->excBuf + st->bufSize - st->windowSize; - st->swBuf = PUSH(st->stack, st->bufSize, float); - st->sw = st->swBuf + st->bufSize - st->windowSize; - - st->exc2Buf = PUSH(st->stack, st->bufSize, float); - st->exc2 = st->exc2Buf + st->bufSize - st->windowSize; - - st->innov = PUSH(st->stack, st->frameSize, float); - - /* Asymmetric "pseudo-Hamming" window */ - { - int part1, part2; - part1 = st->subframeSize*7/2; - part2 = st->subframeSize*5/2; - st->window = PUSH(st->stack, st->windowSize, float); - for (i=0;iwindow[i]=.54-.46*cos(M_PI*i/part1); - for (i=0;iwindow[part1+i]=.54+.46*cos(M_PI*i/part2); - } - /* Create the window for autocorrelation (lag-windowing) */ - st->lagWindow = PUSH(st->stack, st->lpcSize+1, float); - for (i=0;ilpcSize+1;i++) - st->lagWindow[i]=exp(-.5*sqr(2*M_PI*st->lag_factor*i)); - - st->autocorr = PUSH(st->stack, st->lpcSize+1, float); - - st->buf2 = PUSH(st->stack, st->windowSize, float); - - st->lpc = PUSH(st->stack, st->lpcSize+1, float); - st->interp_lpc = PUSH(st->stack, st->lpcSize+1, float); - st->interp_qlpc = PUSH(st->stack, st->lpcSize+1, float); - st->bw_lpc1 = PUSH(st->stack, st->lpcSize+1, float); - st->bw_lpc2 = PUSH(st->stack, st->lpcSize+1, float); - - st->lsp = PUSH(st->stack, st->lpcSize, float); - st->qlsp = PUSH(st->stack, st->lpcSize, float); - st->old_lsp = PUSH(st->stack, st->lpcSize, float); - st->old_qlsp = PUSH(st->stack, st->lpcSize, float); - st->interp_lsp = PUSH(st->stack, st->lpcSize, float); - st->interp_qlsp = PUSH(st->stack, st->lpcSize, float); - st->rc = PUSH(st->stack, st->lpcSize, float); - st->first = 1; - for (i=0;ilpcSize;i++) - { - st->lsp[i]=(M_PI*((float)(i+1)))/(st->lpcSize+1); - } - - st->mem_sp = PUSH(st->stack, st->lpcSize, float); - st->mem_sw = PUSH(st->stack, st->lpcSize, float); - st->mem_sw_whole = PUSH(st->stack, st->lpcSize, float); - st->mem_exc = PUSH(st->stack, st->lpcSize, float); - - st->pi_gain = PUSH(st->stack, st->nbSubframes, float); - - st->pitch = PUSH(st->stack, st->nbSubframes, int); - - st->vbr = PUSHS(st->stack, VBRState); - vbr_init(st->vbr); - st->vbr_quality = 8; - st->vbr_enabled = 0; - st->vad_enabled = 0; - st->dtx_enabled = 0; - st->abr_enabled = 0; - st->abr_drift = 0; - - st->complexity=2; - st->sampling_rate=8000; - st->dtx_count=0; - - return st; -} - -void nb_encoder_destroy(void *state) -{ - EncState *st=(EncState *)state; - /* Free all allocated memory */ - - vbr_destroy(st->vbr); - - /*Free state memory... should be last*/ - speex_free(st); -} - -int nb_encode(void *state, float *in, SpeexBits *bits) -{ - EncState *st; - int i, sub, roots; - int ol_pitch; - float ol_pitch_coef; - float ol_gain; - float *res, *target, *mem; - char *stack; - float *syn_resp; - float lsp_dist=0; - float *orig; - - st=(EncState *)state; - stack=st->stack; - - /* Copy new data in input buffer */ - speex_move(st->inBuf, st->inBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - st->inBuf[st->bufSize-st->frameSize] = in[0] - st->preemph*st->pre_mem; - for (i=1;iframeSize;i++) - st->inBuf[st->bufSize-st->frameSize+i] = in[i] - st->preemph*in[i-1]; - st->pre_mem = in[st->frameSize-1]; - - /* Move signals 1 frame towards the past */ - speex_move(st->exc2Buf, st->exc2Buf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - speex_move(st->excBuf, st->excBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - speex_move(st->swBuf, st->swBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - - - /* Window for analysis */ - for (i=0;iwindowSize;i++) - st->buf2[i] = st->frame[i] * st->window[i]; - - /* Compute auto-correlation */ - _spx_autocorr(st->buf2, st->autocorr, st->lpcSize+1, st->windowSize); - - st->autocorr[0] += 10; /* prevents NANs */ - st->autocorr[0] *= st->lpc_floor; /* Noise floor in auto-correlation domain */ - - /* Lag windowing: equivalent to filtering in the power-spectrum domain */ - for (i=0;ilpcSize+1;i++) - st->autocorr[i] *= st->lagWindow[i]; - - /* Levinson-Durbin */ - wld(st->lpc+1, st->autocorr, st->rc, st->lpcSize); - st->lpc[0]=1; - - /* LPC to LSPs (x-domain) transform */ - roots=lpc_to_lsp (st->lpc, st->lpcSize, st->lsp, 15, 0.2, stack); - /* Check if we found all the roots */ - if (roots==st->lpcSize) - { - /* LSP x-domain to angle domain*/ - for (i=0;ilpcSize;i++) - st->lsp[i] = acos(st->lsp[i]); - } else { - /* Search again if we can afford it */ - if (st->complexity>1) - roots = lpc_to_lsp (st->lpc, st->lpcSize, st->lsp, 11, 0.05, stack); - if (roots==st->lpcSize) - { - /* LSP x-domain to angle domain*/ - for (i=0;ilpcSize;i++) - st->lsp[i] = acos(st->lsp[i]); - } else { - /*If we can't find all LSP's, do some damage control and use previous filter*/ - for (i=0;ilpcSize;i++) - { - st->lsp[i]=st->old_lsp[i]; - } - } - } - - - lsp_dist=0; - for (i=0;ilpcSize;i++) - lsp_dist += (st->old_lsp[i] - st->lsp[i])*(st->old_lsp[i] - st->lsp[i]); - - /* Whole frame analysis (open-loop estimation of pitch and excitation gain) */ - { - if (st->first) - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = st->lsp[i]; - else - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = .375*st->old_lsp[i] + .625*st->lsp[i]; - - lsp_enforce_margin(st->interp_lsp, st->lpcSize, .002); - - /* Compute interpolated LPCs (unquantized) for whole frame*/ - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = cos(st->interp_lsp[i]); - lsp_to_lpc(st->interp_lsp, st->interp_lpc, st->lpcSize,stack); - - - /*Open-loop pitch*/ - if (!st->submodes[st->submodeID] || st->vbr_enabled || st->vad_enabled || SUBMODE(forced_pitch_gain) || - SUBMODE(lbr_pitch) != -1) - { - int nol_pitch[6]; - float nol_pitch_coef[6]; - - bw_lpc(st->gamma1, st->interp_lpc, st->bw_lpc1, st->lpcSize); - bw_lpc(st->gamma2, st->interp_lpc, st->bw_lpc2, st->lpcSize); - - filter_mem2(st->frame, st->bw_lpc1, st->bw_lpc2, st->sw, st->frameSize, st->lpcSize, st->mem_sw_whole); - - open_loop_nbest_pitch(st->sw, st->min_pitch, st->max_pitch, st->frameSize, - nol_pitch, nol_pitch_coef, 6, stack); - ol_pitch=nol_pitch[0]; - ol_pitch_coef = nol_pitch_coef[0]; - /*Try to remove pitch multiples*/ - for (i=1;i<6;i++) - { - if ((nol_pitch_coef[i]>.85*ol_pitch_coef) && - (fabs(nol_pitch[i]-ol_pitch/2.0)<=1 || fabs(nol_pitch[i]-ol_pitch/3.0)<=1 || - fabs(nol_pitch[i]-ol_pitch/4.0)<=1 || fabs(nol_pitch[i]-ol_pitch/5.0)<=1)) - { - /*ol_pitch_coef=nol_pitch_coef[i];*/ - ol_pitch = nol_pitch[i]; - } - } - /*if (ol_pitch>50) - ol_pitch/=2;*/ - /*ol_pitch_coef = sqrt(ol_pitch_coef);*/ - } else { - ol_pitch=0; - ol_pitch_coef=0; - } - /*Compute "real" excitation*/ - fir_mem2(st->frame, st->interp_lpc, st->exc, st->frameSize, st->lpcSize, st->mem_exc); - - /* Compute open-loop excitation gain */ - ol_gain=0; - for (i=0;iframeSize;i++) - ol_gain += st->exc[i]*st->exc[i]; - - ol_gain=sqrt(1+ol_gain/st->frameSize); - } - - /*VBR stuff*/ - if (st->vbr && (st->vbr_enabled||st->vad_enabled)) - { - - if (st->abr_enabled) - { - float qual_change=0; - if (st->abr_drift2 * st->abr_drift > 0) - { - /* Only adapt if long-term and short-term drift are the same sign */ - qual_change = -.00001*st->abr_drift/(1+st->abr_count); - if (qual_change>.05) - qual_change=.05; - if (qual_change<-.05) - qual_change=-.05; - } - st->vbr_quality += qual_change; - if (st->vbr_quality>10) - st->vbr_quality=10; - if (st->vbr_quality<0) - st->vbr_quality=0; - } - - st->relative_quality = vbr_analysis(st->vbr, in, st->frameSize, ol_pitch, ol_pitch_coef); - /*if (delta_qual<0)*/ - /* delta_qual*=.1*(3+st->vbr_quality);*/ - if (st->vbr_enabled) - { - int mode; - int choice=0; - float min_diff=100; - mode = 8; - while (mode) - { - int v1; - float thresh; - v1=(int)floor(st->vbr_quality); - if (v1==10) - thresh = vbr_nb_thresh[mode][v1]; - else - thresh = (st->vbr_quality-v1)*vbr_nb_thresh[mode][v1+1] + (1+v1-st->vbr_quality)*vbr_nb_thresh[mode][v1]; - if (st->relative_quality > thresh && - st->relative_quality-threshrelative_quality-thresh; - } - mode--; - } - mode=choice; - if (mode==0) - { - if (st->dtx_count==0 || lsp_dist>.05 || !st->dtx_enabled || st->dtx_count>20) - { - mode=1; - st->dtx_count=1; - } else { - mode=0; - st->dtx_count++; - } - } else { - st->dtx_count=0; - } - - speex_encoder_ctl(state, SPEEX_SET_MODE, &mode); - - if (st->abr_enabled) - { - int bitrate; - speex_encoder_ctl(state, SPEEX_GET_BITRATE, &bitrate); - st->abr_drift+=(bitrate-st->abr_enabled); - st->abr_drift2 = .95*st->abr_drift2 + .05*(bitrate-st->abr_enabled); - st->abr_count += 1.0; - } - - } else { - /*VAD only case*/ - int mode; - if (st->relative_quality<2) - { - if (st->dtx_count==0 || lsp_dist>.05 || !st->dtx_enabled || st->dtx_count>20) - { - st->dtx_count=1; - mode=1; - } else { - mode=0; - st->dtx_count++; - } - } else { - st->dtx_count = 0; - mode=st->submodeSelect; - } - /*speex_encoder_ctl(state, SPEEX_SET_MODE, &mode);*/ - st->submodeID=mode; - } - } else { - st->relative_quality = -1; - } - - /* First, transmit a zero for narrowband */ - speex_bits_pack(bits, 0, 1); - - /* Transmit the sub-mode we use for this frame */ - speex_bits_pack(bits, st->submodeID, NB_SUBMODE_BITS); - - - /* If null mode (no transmission), just set a couple things to zero*/ - if (st->submodes[st->submodeID] == NULL) - { - for (i=0;iframeSize;i++) - st->exc[i]=st->exc2[i]=st->sw[i]=0; - - for (i=0;ilpcSize;i++) - st->mem_sw[i]=0; - st->first=1; - st->bounded_pitch = 1; - - /* Final signal synthesis from excitation */ - iir_mem2(st->exc, st->interp_qlpc, st->frame, st->frameSize, st->lpcSize, st->mem_sp); - - in[0] = st->frame[0] + st->preemph*st->pre_mem2; - for (i=1;iframeSize;i++) - in[i]=st->frame[i] + st->preemph*in[i-1]; - st->pre_mem2=in[st->frameSize-1]; - - return 0; - - } - - /* LSP Quantization */ - if (st->first) - { - for (i=0;ilpcSize;i++) - st->old_lsp[i] = st->lsp[i]; - } - - - /*Quantize LSPs*/ -#if 1 /*0 for unquantized*/ - SUBMODE(lsp_quant)(st->lsp, st->qlsp, st->lpcSize, bits); -#else - for (i=0;ilpcSize;i++) - st->qlsp[i]=st->lsp[i]; -#endif - - /*If we use low bit-rate pitch mode, transmit open-loop pitch*/ - if (SUBMODE(lbr_pitch)!=-1) - { - speex_bits_pack(bits, ol_pitch-st->min_pitch, 7); - } - - if (SUBMODE(forced_pitch_gain)) - { - int quant; - quant = (int)floor(.5+15*ol_pitch_coef); - if (quant>15) - quant=15; - if (quant<0) - quant=0; - speex_bits_pack(bits, quant, 4); - ol_pitch_coef=0.066667*quant; - } - - - /*Quantize and transmit open-loop excitation gain*/ - { - int qe = (int)(floor(3.5*log(ol_gain))); - if (qe<0) - qe=0; - if (qe>31) - qe=31; - ol_gain = exp(qe/3.5); - speex_bits_pack(bits, qe, 5); - } - - /* Special case for first frame */ - if (st->first) - { - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - } - - /* Filter response */ - res = PUSH(stack, st->subframeSize, float); - /* Target signal */ - target = PUSH(stack, st->subframeSize, float); - syn_resp = PUSH(stack, st->subframeSize, float); - mem = PUSH(stack, st->lpcSize, float); - orig = PUSH(stack, st->frameSize, float); - for (i=0;iframeSize;i++) - orig[i]=st->frame[i]; - - /* Loop on sub-frames */ - for (sub=0;subnbSubframes;sub++) - { - float tmp; - int offset; - float *sp, *sw, *exc, *exc2; - int pitch; - - /* Offset relative to start of frame */ - offset = st->subframeSize*sub; - /* Original signal */ - sp=st->frame+offset; - /* Excitation */ - exc=st->exc+offset; - /* Weighted signal */ - sw=st->sw+offset; - - exc2=st->exc2+offset; - - - /* LSP interpolation (quantized and unquantized) */ - tmp = (1.0 + sub)/st->nbSubframes; - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = (1-tmp)*st->old_lsp[i] + tmp*st->lsp[i]; - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = (1-tmp)*st->old_qlsp[i] + tmp*st->qlsp[i]; - - /* Make sure the filters are stable */ - lsp_enforce_margin(st->interp_lsp, st->lpcSize, .002); - lsp_enforce_margin(st->interp_qlsp, st->lpcSize, .002); - - /* Compute interpolated LPCs (quantized and unquantized) */ - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = cos(st->interp_lsp[i]); - lsp_to_lpc(st->interp_lsp, st->interp_lpc, st->lpcSize,stack); - - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = cos(st->interp_qlsp[i]); - lsp_to_lpc(st->interp_qlsp, st->interp_qlpc, st->lpcSize, stack); - - /* Compute analysis filter gain at w=pi (for use in SB-CELP) */ - tmp=1; - st->pi_gain[sub]=0; - for (i=0;i<=st->lpcSize;i++) - { - st->pi_gain[sub] += tmp*st->interp_qlpc[i]; - tmp = -tmp; - } - - /* Compute bandwidth-expanded (unquantized) LPCs for perceptual weighting */ - bw_lpc(st->gamma1, st->interp_lpc, st->bw_lpc1, st->lpcSize); - if (st->gamma2>=0) - bw_lpc(st->gamma2, st->interp_lpc, st->bw_lpc2, st->lpcSize); - else - { - st->bw_lpc2[0]=1; - st->bw_lpc2[1]=-st->preemph; - for (i=2;i<=st->lpcSize;i++) - st->bw_lpc2[i]=0; - } - - /* Compute impulse response of A(z/g1) / ( A(z)*A(z/g2) )*/ - for (i=0;isubframeSize;i++) - exc[i]=0; - exc[0]=1; - syn_percep_zero(exc, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, syn_resp, st->subframeSize, st->lpcSize, stack); - - /* Reset excitation */ - for (i=0;isubframeSize;i++) - exc[i]=0; - for (i=0;isubframeSize;i++) - exc2[i]=0; - - /* Compute zero response of A(z/g1) / ( A(z/g2) * A(z) ) */ - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sp[i]; - iir_mem2(exc, st->interp_qlpc, exc, st->subframeSize, st->lpcSize, mem); - - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sw[i]; - filter_mem2(exc, st->bw_lpc1, st->bw_lpc2, res, st->subframeSize, st->lpcSize, mem); - - /* Compute weighted signal */ - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sw[i]; - filter_mem2(sp, st->bw_lpc1, st->bw_lpc2, sw, st->subframeSize, st->lpcSize, mem); - - /* Compute target signal */ - for (i=0;isubframeSize;i++) - target[i]=sw[i]-res[i]; - - for (i=0;isubframeSize;i++) - exc[i]=exc2[i]=0; - - /* If we have a long-term predictor (otherwise, something's wrong) */ - if (SUBMODE(ltp_quant)) - { - int pit_min, pit_max; - /* Long-term prediction */ - if (SUBMODE(lbr_pitch) != -1) - { - /* Low bit-rate pitch handling */ - int margin; - margin = SUBMODE(lbr_pitch); - if (margin) - { - if (ol_pitch < st->min_pitch+margin-1) - ol_pitch=st->min_pitch+margin-1; - if (ol_pitch > st->max_pitch-margin) - ol_pitch=st->max_pitch-margin; - pit_min = ol_pitch-margin+1; - pit_max = ol_pitch+margin; - } else { - pit_min=pit_max=ol_pitch; - } - } else { - pit_min = st->min_pitch; - pit_max = st->max_pitch; - } - - /* Force pitch to use only the current frame if needed */ - if (st->bounded_pitch && pit_max>offset) - pit_max=offset; - - /* Perform pitch search */ - pitch = SUBMODE(ltp_quant)(target, sw, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, - exc, SUBMODE(ltp_params), pit_min, pit_max, ol_pitch_coef, - st->lpcSize, st->subframeSize, bits, stack, - exc2, syn_resp, st->complexity); - - st->pitch[sub]=pitch; - } else { - speex_error ("No pitch prediction, what's wrong"); - } - - /* Update target for adaptive codebook contribution */ - syn_percep_zero(exc, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, res, st->subframeSize, st->lpcSize, stack); - for (i=0;isubframeSize;i++) - target[i]-=res[i]; - - - /* Quantization of innovation */ - { - float *innov; - float ener=0, ener_1; - - innov = st->innov+sub*st->subframeSize; - for (i=0;isubframeSize;i++) - innov[i]=0; - - residue_percep_zero(target, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, st->buf2, st->subframeSize, st->lpcSize, stack); - for (i=0;isubframeSize;i++) - ener+=st->buf2[i]*st->buf2[i]; - ener=sqrt(.1+ener/st->subframeSize); - /*for (i=0;isubframeSize;i++) - printf ("%f\n", st->buf2[i]/ener); - */ - - ener /= ol_gain; - - /* Calculate gain correction for the sub-frame (if any) */ - if (SUBMODE(have_subframe_gain)) - { - int qe; - ener=log(ener); - if (SUBMODE(have_subframe_gain)==3) - { - qe = vq_index(&ener, exc_gain_quant_scal3, 1, 8); - speex_bits_pack(bits, qe, 3); - ener=exc_gain_quant_scal3[qe]; - } else { - qe = vq_index(&ener, exc_gain_quant_scal1, 1, 2); - speex_bits_pack(bits, qe, 1); - ener=exc_gain_quant_scal1[qe]; - } - ener=exp(ener); - } else { - ener=1; - } - - ener*=ol_gain; - - /*printf ("%f %f\n", ener, ol_gain);*/ - - ener_1 = 1/ener; - - /* Normalize innovation */ - for (i=0;isubframeSize;i++) - target[i]*=ener_1; - - /* Quantize innovation */ - if (SUBMODE(innovation_quant)) - { - /* Codebook search */ - SUBMODE(innovation_quant)(target, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, - SUBMODE(innovation_params), st->lpcSize, st->subframeSize, - innov, syn_resp, bits, stack, st->complexity); - - /* De-normalize innovation and update excitation */ - for (i=0;isubframeSize;i++) - innov[i]*=ener; - for (i=0;isubframeSize;i++) - exc[i] += innov[i]; - } else { - speex_error("No fixed codebook"); - } - - /* In some (rare) modes, we do a second search (more bits) to reduce noise even more */ - if (SUBMODE(double_codebook)) { - char *tmp_stack=stack; - float *innov2 = PUSH(tmp_stack, st->subframeSize, float); - for (i=0;isubframeSize;i++) - innov2[i]=0; - for (i=0;isubframeSize;i++) - target[i]*=2.2; - SUBMODE(innovation_quant)(target, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, - SUBMODE(innovation_params), st->lpcSize, st->subframeSize, - innov2, syn_resp, bits, tmp_stack, st->complexity); - for (i=0;isubframeSize;i++) - innov2[i]*=ener*(1/2.2); - for (i=0;isubframeSize;i++) - exc[i] += innov2[i]; - } - - for (i=0;isubframeSize;i++) - target[i]*=ener; - - } - - /*Keep the previous memory*/ - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sp[i]; - /* Final signal synthesis from excitation */ - iir_mem2(exc, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, st->mem_sp); - - /* Compute weighted signal again, from synthesized speech (not sure it's the right thing) */ - filter_mem2(sp, st->bw_lpc1, st->bw_lpc2, sw, st->subframeSize, st->lpcSize, st->mem_sw); - for (i=0;isubframeSize;i++) - exc2[i]=exc[i]; - } - - /* Store the LSPs for interpolation in the next frame */ - if (st->submodeID>=1) - { - for (i=0;ilpcSize;i++) - st->old_lsp[i] = st->lsp[i]; - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - } - - if (st->submodeID==1) - { - if (st->dtx_count) - speex_bits_pack(bits, 15, 4); - else - speex_bits_pack(bits, 0, 4); - } - - /* The next frame will not be the first (Duh!) */ - st->first = 0; - - { - float ener=0, err=0; - float snr; - for (i=0;iframeSize;i++) - { - ener+=st->frame[i]*st->frame[i]; - err += (st->frame[i]-orig[i])*(st->frame[i]-orig[i]); - } - snr = 10*log10((ener+1)/(err+1)); - /*printf ("%f %f %f\n", snr, ener, err);*/ - } - - /* Replace input by synthesized speech */ - in[0] = st->frame[0] + st->preemph*st->pre_mem2; - for (i=1;iframeSize;i++) - in[i]=st->frame[i] + st->preemph*in[i-1]; - st->pre_mem2=in[st->frameSize-1]; - - if (SUBMODE(innovation_quant) == noise_codebook_quant || st->submodeID==0) - st->bounded_pitch = 1; - else - st->bounded_pitch = 0; - - return 1; -} - - -void *nb_decoder_init(SpeexMode *m) -{ - DecState *st; - SpeexNBMode *mode; - int i; - - mode=(SpeexNBMode*)m->mode; - st = (DecState *)speex_alloc(sizeof(DecState)+4000*sizeof(float)); - st->mode=m; - - st->stack = ((char*)st) + sizeof(DecState); - - st->first=1; - /* Codec parameters, should eventually have several "modes"*/ - st->frameSize = mode->frameSize; - st->windowSize = st->frameSize*3/2; - st->nbSubframes=mode->frameSize/mode->subframeSize; - st->subframeSize=mode->subframeSize; - st->lpcSize = mode->lpcSize; - st->bufSize = mode->bufSize; - st->gamma1=mode->gamma1; - st->gamma2=mode->gamma2; - st->min_pitch=mode->pitchStart; - st->max_pitch=mode->pitchEnd; - st->preemph = mode->preemph; - - st->submodes=mode->submodes; - st->submodeID=mode->defaultSubmode; - - st->pre_mem=0; - st->lpc_enh_enabled=0; - - - st->inBuf = PUSH(st->stack, st->bufSize, float); - st->frame = st->inBuf + st->bufSize - st->windowSize; - st->excBuf = PUSH(st->stack, st->bufSize, float); - st->exc = st->excBuf + st->bufSize - st->windowSize; - for (i=0;ibufSize;i++) - st->inBuf[i]=0; - for (i=0;ibufSize;i++) - st->excBuf[i]=0; - st->innov = PUSH(st->stack, st->frameSize, float); - - st->interp_qlpc = PUSH(st->stack, st->lpcSize+1, float); - st->qlsp = PUSH(st->stack, st->lpcSize, float); - st->old_qlsp = PUSH(st->stack, st->lpcSize, float); - st->interp_qlsp = PUSH(st->stack, st->lpcSize, float); - st->mem_sp = PUSH(st->stack, 5*st->lpcSize, float); - st->comb_mem = PUSHS(st->stack, CombFilterMem); - comp_filter_mem_init (st->comb_mem); - - st->pi_gain = PUSH(st->stack, st->nbSubframes, float); - st->last_pitch = 40; - st->count_lost=0; - st->pitch_gain_buf[0] = st->pitch_gain_buf[1] = st->pitch_gain_buf[2] = 0; - st->pitch_gain_buf_idx = 0; - - st->sampling_rate=8000; - st->last_ol_gain = 0; - - st->user_callback.func = &speex_default_user_handler; - st->user_callback.data = NULL; - for (i=0;i<16;i++) - st->speex_callbacks[i].func = NULL; - - st->voc_m1=st->voc_m2=st->voc_mean=0; - st->voc_offset=0; - st->dtx_enabled=0; - return st; -} - -void nb_decoder_destroy(void *state) -{ - DecState *st; - st=(DecState*)state; - - speex_free(state); -} - -#define median3(a, b, c) ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) : ((c) < (b) ? (b) : ((c) < (a) ? (c) : (a)))) - -static void nb_decode_lost(DecState *st, float *out, char *stack) -{ - int i, sub; - float *awk1, *awk2, *awk3; - float pitch_gain, fact, gain_med; - - fact = exp(-.04*st->count_lost*st->count_lost); - gain_med = median3(st->pitch_gain_buf[0], st->pitch_gain_buf[1], st->pitch_gain_buf[2]); - if (gain_med < st->last_pitch_gain) - st->last_pitch_gain = gain_med; - - pitch_gain = st->last_pitch_gain; - if (pitch_gain>.95) - pitch_gain=.95; - - pitch_gain *= fact; - - /* Shift all buffers by one frame */ - speex_move(st->inBuf, st->inBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - speex_move(st->excBuf, st->excBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - - awk1=PUSH(stack, (st->lpcSize+1), float); - awk2=PUSH(stack, (st->lpcSize+1), float); - awk3=PUSH(stack, (st->lpcSize+1), float); - - for (sub=0;subnbSubframes;sub++) - { - int offset; - float *sp, *exc; - /* Offset relative to start of frame */ - offset = st->subframeSize*sub; - /* Original signal */ - sp=st->frame+offset; - /* Excitation */ - exc=st->exc+offset; - /* Excitation after post-filter*/ - - /* Calculate perceptually enhanced LPC filter */ - if (st->lpc_enh_enabled) - { - float r=.9; - - float k1,k2,k3; - if (st->submodes[st->submodeID] != NULL) - { - k1=SUBMODE(lpc_enh_k1); - k2=SUBMODE(lpc_enh_k2); - } else { - k1=k2=.7; - } - k3=(1-(1-r*k1)/(1-r*k2))/r; - if (!st->lpc_enh_enabled) - { - k1=k2; - k3=0; - } - bw_lpc(k1, st->interp_qlpc, awk1, st->lpcSize); - bw_lpc(k2, st->interp_qlpc, awk2, st->lpcSize); - bw_lpc(k3, st->interp_qlpc, awk3, st->lpcSize); - } - - /* Make up a plausible excitation */ - /* THIS CAN BE IMPROVED */ - /*if (pitch_gain>.95) - pitch_gain=.95;*/ - { - float innov_gain=0; - for (i=0;iframeSize;i++) - innov_gain += st->innov[i]*st->innov[i]; - innov_gain=sqrt(innov_gain/st->frameSize); - for (i=0;isubframeSize;i++) - { -#if 0 - exc[i] = pitch_gain * exc[i - st->last_pitch] + fact*sqrt(1-pitch_gain)*st->innov[i+offset]; - /*Just so it give the same lost packets as with if 0*/ - /*rand();*/ -#else - /*exc[i]=pitch_gain*exc[i-st->last_pitch] + fact*st->innov[i+offset];*/ - exc[i]=pitch_gain*exc[i-st->last_pitch] + - fact*sqrt(1-pitch_gain)*speex_rand(innov_gain); -#endif - } - } - for (i=0;isubframeSize;i++) - sp[i]=exc[i]; - - /* Signal synthesis */ - if (st->lpc_enh_enabled) - { - filter_mem2(sp, awk2, awk1, sp, st->subframeSize, st->lpcSize, - st->mem_sp+st->lpcSize); - filter_mem2(sp, awk3, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, - st->mem_sp); - } else { - for (i=0;ilpcSize;i++) - st->mem_sp[st->lpcSize+i] = 0; - iir_mem2(sp, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, - st->mem_sp); - } - } - - out[0] = st->frame[0] + st->preemph*st->pre_mem; - for (i=1;iframeSize;i++) - out[i]=st->frame[i] + st->preemph*out[i-1]; - st->pre_mem=out[st->frameSize-1]; - - st->first = 0; - st->count_lost++; - st->pitch_gain_buf[st->pitch_gain_buf_idx++] = pitch_gain; - if (st->pitch_gain_buf_idx > 2) /* rollover */ - st->pitch_gain_buf_idx = 0; -} - -int nb_decode(void *state, SpeexBits *bits, float *out) -{ - DecState *st; - int i, sub; - int pitch; - float pitch_gain[3]; - float ol_gain=0; - int ol_pitch=0; - float ol_pitch_coef=0; - int best_pitch=40; - float best_pitch_gain=0; - int wideband; - int m; - char *stack; - float *awk1, *awk2, *awk3; - float pitch_average=0; - - st=(DecState*)state; - stack=st->stack; - - /* Check if we're in DTX mode*/ - if (!bits && st->dtx_enabled) - { - st->submodeID=0; - } else - { - /* If bits is NULL, consider the packet to be lost (what could we do anyway) */ - if (!bits) - { - nb_decode_lost(st, out, stack); - return 0; - } - - /* Search for next narrowband block (handle requests, skip wideband blocks) */ - do { - wideband = speex_bits_unpack_unsigned(bits, 1); - if (wideband) /* Skip wideband block (for compatibility) */ - { - int submode; - int advance; - advance = submode = speex_bits_unpack_unsigned(bits, SB_SUBMODE_BITS); - speex_mode_query(&speex_wb_mode, SPEEX_SUBMODE_BITS_PER_FRAME, &advance); - if (advance < 0) - { - speex_warning ("Invalid wideband mode encountered. Corrupted stream?"); - return -2; - } - advance -= (SB_SUBMODE_BITS+1); - speex_bits_advance(bits, advance); - wideband = speex_bits_unpack_unsigned(bits, 1); - if (wideband) - { - advance = submode = speex_bits_unpack_unsigned(bits, SB_SUBMODE_BITS); - speex_mode_query(&speex_wb_mode, SPEEX_SUBMODE_BITS_PER_FRAME, &advance); - if (advance < 0) - { - speex_warning ("Invalid wideband mode encountered: corrupted stream?"); - return -2; - } - advance -= (SB_SUBMODE_BITS+1); - speex_bits_advance(bits, advance); - wideband = speex_bits_unpack_unsigned(bits, 1); - if (wideband) - { - speex_warning ("More than to wideband layers found: corrupted stream?"); - return -2; - } - - } - } - - /* FIXME: Check for overflow */ - m = speex_bits_unpack_unsigned(bits, 4); - if (m==15) /* We found a terminator */ - { - return -1; - } else if (m==14) /* Speex in-band request */ - { - int ret = speex_inband_handler(bits, st->speex_callbacks, state); - if (ret) - return ret; - } else if (m==13) /* User in-band request */ - { - int ret = st->user_callback.func(bits, state, st->user_callback.data); - if (ret) - return ret; - } else if (m>8) /* Invalid mode */ - { - speex_warning("Invalid mode encountered: corrupted stream?"); - return -2; - } - - } while (m>8); - - /* Get the sub-mode that was used */ - st->submodeID = m; - - } - - /* Shift all buffers by one frame */ - speex_move(st->inBuf, st->inBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - speex_move(st->excBuf, st->excBuf+st->frameSize, (st->bufSize-st->frameSize)*sizeof(float)); - - /* If null mode (no transmission), just set a couple things to zero*/ - if (st->submodes[st->submodeID] == NULL) - { - float *lpc; - lpc = PUSH(stack,11, float); - bw_lpc(.93, st->interp_qlpc, lpc, 10); - /*for (i=0;iframeSize;i++) - st->exc[i]=0;*/ - { - float innov_gain=0; - float pgain=st->last_pitch_gain; - if (pgain>.6) - pgain=.6; - for (i=0;iframeSize;i++) - innov_gain += st->innov[i]*st->innov[i]; - innov_gain=sqrt(innov_gain/st->frameSize); - for (i=0;iframeSize;i++) - st->exc[i]=0; - speex_rand_vec(innov_gain, st->exc, st->frameSize); - } - - - st->first=1; - - /* Final signal synthesis from excitation */ - iir_mem2(st->exc, lpc, st->frame, st->frameSize, st->lpcSize, st->mem_sp); - - out[0] = st->frame[0] + st->preemph*st->pre_mem; - for (i=1;iframeSize;i++) - out[i]=st->frame[i] + st->preemph*out[i-1]; - st->pre_mem=out[st->frameSize-1]; - st->count_lost=0; - return 0; - } - - /* Unquantize LSPs */ - SUBMODE(lsp_unquant)(st->qlsp, st->lpcSize, bits); - - /*Damp memory if a frame was lost and the LSP changed too much*/ - if (st->count_lost) - { - float lsp_dist=0, fact; - for (i=0;ilpcSize;i++) - lsp_dist += fabs(st->old_qlsp[i] - st->qlsp[i]); - fact = .6*exp(-.2*lsp_dist); - for (i=0;i<2*st->lpcSize;i++) - st->mem_sp[i] *= fact; - } - - - /* Handle first frame and lost-packet case */ - if (st->first || st->count_lost) - { - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - } - - /* Get open-loop pitch estimation for low bit-rate pitch coding */ - if (SUBMODE(lbr_pitch)!=-1) - { - ol_pitch = st->min_pitch+speex_bits_unpack_unsigned(bits, 7); - } - - if (SUBMODE(forced_pitch_gain)) - { - int quant; - quant = speex_bits_unpack_unsigned(bits, 4); - ol_pitch_coef=0.066667*quant; - } - - /* Get global excitation gain */ - { - int qe; - qe = speex_bits_unpack_unsigned(bits, 5); - ol_gain = exp(qe/3.5); - } - - awk1=PUSH(stack, st->lpcSize+1, float); - awk2=PUSH(stack, st->lpcSize+1, float); - awk3=PUSH(stack, st->lpcSize+1, float); - - if (st->submodeID==1) - { - int extra; - extra = speex_bits_unpack_unsigned(bits, 4); - - if (extra==15) - st->dtx_enabled=1; - else - st->dtx_enabled=0; - } - if (st->submodeID>1) - st->dtx_enabled=0; - - /*Loop on subframes */ - for (sub=0;subnbSubframes;sub++) - { - int offset; - float *sp, *exc, tmp; - - /* Offset relative to start of frame */ - offset = st->subframeSize*sub; - /* Original signal */ - sp=st->frame+offset; - /* Excitation */ - exc=st->exc+offset; - /* Excitation after post-filter*/ - - /* LSP interpolation (quantized and unquantized) */ - tmp = (1.0 + sub)/st->nbSubframes; - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = (1-tmp)*st->old_qlsp[i] + tmp*st->qlsp[i]; - - /* Make sure the LSP's are stable */ - lsp_enforce_margin(st->interp_qlsp, st->lpcSize, .002); - - - /* Compute interpolated LPCs (unquantized) */ - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = cos(st->interp_qlsp[i]); - lsp_to_lpc(st->interp_qlsp, st->interp_qlpc, st->lpcSize, stack); - - /* Compute enhanced synthesis filter */ - if (st->lpc_enh_enabled) - { - float r=.9; - - float k1,k2,k3; - k1=SUBMODE(lpc_enh_k1); - k2=SUBMODE(lpc_enh_k2); - k3=(1-(1-r*k1)/(1-r*k2))/r; - if (!st->lpc_enh_enabled) - { - k1=k2; - k3=0; - } - bw_lpc(k1, st->interp_qlpc, awk1, st->lpcSize); - bw_lpc(k2, st->interp_qlpc, awk2, st->lpcSize); - bw_lpc(k3, st->interp_qlpc, awk3, st->lpcSize); - - } - - /* Compute analysis filter at w=pi */ - tmp=1; - st->pi_gain[sub]=0; - for (i=0;i<=st->lpcSize;i++) - { - st->pi_gain[sub] += tmp*st->interp_qlpc[i]; - tmp = -tmp; - } - - /* Reset excitation */ - for (i=0;isubframeSize;i++) - exc[i]=0; - - /*Adaptive codebook contribution*/ - if (SUBMODE(ltp_unquant)) - { - int pit_min, pit_max; - /* Handle pitch constraints if any */ - if (SUBMODE(lbr_pitch) != -1) - { - int margin; - margin = SUBMODE(lbr_pitch); - if (margin) - { -/* GT - need optimization? - if (ol_pitch < st->min_pitch+margin-1) - ol_pitch=st->min_pitch+margin-1; - if (ol_pitch > st->max_pitch-margin) - ol_pitch=st->max_pitch-margin; - pit_min = ol_pitch-margin+1; - pit_max = ol_pitch+margin; -*/ - pit_min = ol_pitch-margin+1; - if (pit_min < st->min_pitch) - pit_min = st->min_pitch; - pit_max = ol_pitch+margin; - if (pit_max > st->max_pitch) - pit_max = st->max_pitch; - } else { - pit_min = pit_max = ol_pitch; - } - } else { - pit_min = st->min_pitch; - pit_max = st->max_pitch; - } - - /* Pitch synthesis */ - SUBMODE(ltp_unquant)(exc, pit_min, pit_max, ol_pitch_coef, SUBMODE(ltp_params), - st->subframeSize, &pitch, &pitch_gain[0], bits, stack, st->count_lost, offset, st->last_pitch_gain); - - /* If we had lost frames, check energy of last received frame */ - if (st->count_lost && ol_gain < st->last_ol_gain) - { - float fact = ol_gain/(st->last_ol_gain+1); - for (i=0;isubframeSize;i++) - exc[i]*=fact; - } - - tmp = fabs(pitch_gain[0]+pitch_gain[1]+pitch_gain[2]); - tmp = fabs(pitch_gain[1]); - if (pitch_gain[0]>0) - tmp += pitch_gain[0]; - else - tmp -= .5*pitch_gain[0]; - if (pitch_gain[2]>0) - tmp += pitch_gain[2]; - else - tmp -= .5*pitch_gain[0]; - - - pitch_average += tmp; - if (tmp>best_pitch_gain) - { - best_pitch = pitch; - best_pitch_gain = tmp; - /* best_pitch_gain = tmp*.9; - if (best_pitch_gain>.85) - best_pitch_gain=.85;*/ - } - } else { - speex_error("No pitch prediction, what's wrong"); - } - - /* Unquantize the innovation */ - { - int q_energy; - float ener; - float *innov; - - innov = st->innov+sub*st->subframeSize; - for (i=0;isubframeSize;i++) - innov[i]=0; - - /* Decode sub-frame gain correction */ - if (SUBMODE(have_subframe_gain)==3) - { - q_energy = speex_bits_unpack_unsigned(bits, 3); - ener = ol_gain*exp(exc_gain_quant_scal3[q_energy]); - } else if (SUBMODE(have_subframe_gain)==1) - { - q_energy = speex_bits_unpack_unsigned(bits, 1); - ener = ol_gain*exp(exc_gain_quant_scal1[q_energy]); - } else { - ener = ol_gain; - } - - if (SUBMODE(innovation_unquant)) - { - /*Fixed codebook contribution*/ - SUBMODE(innovation_unquant)(innov, SUBMODE(innovation_params), st->subframeSize, bits, stack); - } else { - speex_error("No fixed codebook"); - } - - /* De-normalize innovation and update excitation */ - for (i=0;isubframeSize;i++) - innov[i]*=ener; - - /*Vocoder mode*/ - if (st->submodeID==1) - { - float g=ol_pitch_coef; - - - for (i=0;isubframeSize;i++) - exc[i]=0; - while (st->voc_offsetsubframeSize) - { - if (st->voc_offset>=0) - exc[st->voc_offset]=sqrt(1.0*ol_pitch); - st->voc_offset+=ol_pitch; - } - st->voc_offset -= st->subframeSize; - - g=.5+2*(g-.6); - if (g<0) - g=0; - if (g>1) - g=1; - for (i=0;isubframeSize;i++) - { - float exci=exc[i]; - exc[i]=.8*g*exc[i]*ol_gain + .6*g*st->voc_m1*ol_gain + .5*g*innov[i] - .5*g*st->voc_m2 + (1-g)*innov[i]; - st->voc_m1 = exci; - st->voc_m2=innov[i]; - st->voc_mean = .95*st->voc_mean + .05*exc[i]; - exc[i]-=st->voc_mean; - } - } else { - for (i=0;isubframeSize;i++) - exc[i]+=innov[i]; - } - /* Decode second codebook (only for some modes) */ - if (SUBMODE(double_codebook)) - { - char *tmp_stack=stack; - float *innov2 = PUSH(tmp_stack, st->subframeSize, float); - for (i=0;isubframeSize;i++) - innov2[i]=0; - SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), st->subframeSize, bits, tmp_stack); - for (i=0;isubframeSize;i++) - innov2[i]*=ener*(1/2.2); - for (i=0;isubframeSize;i++) - exc[i] += innov2[i]; - } - - } - - for (i=0;isubframeSize;i++) - sp[i]=exc[i]; - - /* Signal synthesis */ - if (st->lpc_enh_enabled && SUBMODE(comb_gain)>0) - comb_filter(exc, sp, st->interp_qlpc, st->lpcSize, st->subframeSize, - pitch, pitch_gain, SUBMODE(comb_gain), st->comb_mem); - if (st->lpc_enh_enabled) - { - /* Use enhanced LPC filter */ - filter_mem2(sp, awk2, awk1, sp, st->subframeSize, st->lpcSize, - st->mem_sp+st->lpcSize); - filter_mem2(sp, awk3, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, - st->mem_sp); - } else { - /* Use regular filter */ - for (i=0;ilpcSize;i++) - st->mem_sp[st->lpcSize+i] = 0; - iir_mem2(sp, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, - st->mem_sp); - } - } - - /*Copy output signal*/ - out[0] = st->frame[0] + st->preemph*st->pre_mem; - for (i=1;iframeSize;i++) - out[i]=st->frame[i] + st->preemph*out[i-1]; - st->pre_mem=out[st->frameSize-1]; - - - /* Store the LSPs for interpolation in the next frame */ - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - - /* The next frame will not be the first (Duh!) */ - st->first = 0; - st->count_lost=0; - st->last_pitch = best_pitch; - st->last_pitch_gain = .25*pitch_average; - st->pitch_gain_buf[st->pitch_gain_buf_idx++] = st->last_pitch_gain; - if (st->pitch_gain_buf_idx > 2) /* rollover */ - st->pitch_gain_buf_idx = 0; - - st->last_ol_gain = ol_gain; - - return 0; -} - -int nb_encoder_ctl(void *state, int request, void *ptr) -{ - EncState *st; - st=(EncState*)state; - switch(request) - { - case SPEEX_GET_FRAME_SIZE: - (*(int*)ptr) = st->frameSize; - break; - case SPEEX_SET_LOW_MODE: - case SPEEX_SET_MODE: - st->submodeSelect = st->submodeID = (*(int*)ptr); - break; - case SPEEX_GET_LOW_MODE: - case SPEEX_GET_MODE: - (*(int*)ptr) = st->submodeID; - break; - case SPEEX_SET_VBR: - st->vbr_enabled = (*(int*)ptr); - break; - case SPEEX_GET_VBR: - (*(int*)ptr) = st->vbr_enabled; - break; - case SPEEX_SET_VAD: - st->vad_enabled = (*(int*)ptr); - break; - case SPEEX_GET_VAD: - (*(int*)ptr) = st->vad_enabled; - break; - case SPEEX_SET_DTX: - st->dtx_enabled = (*(int*)ptr); - break; - case SPEEX_GET_DTX: - (*(int*)ptr) = st->dtx_enabled; - break; - case SPEEX_SET_ABR: - st->abr_enabled = (*(int*)ptr); - st->vbr_enabled = 1; - { - int i=10, rate, target; - float vbr_qual; - target = (*(int*)ptr); - while (i>=0) - { - speex_encoder_ctl(st, SPEEX_SET_QUALITY, &i); - speex_encoder_ctl(st, SPEEX_GET_BITRATE, &rate); - if (rate <= target) - break; - i--; - } - vbr_qual=i; - if (vbr_qual<0) - vbr_qual=0; - speex_encoder_ctl(st, SPEEX_SET_VBR_QUALITY, &vbr_qual); - st->abr_count=0; - st->abr_drift=0; - st->abr_drift2=0; - } - - break; - case SPEEX_GET_ABR: - (*(int*)ptr) = st->abr_enabled; - break; - case SPEEX_SET_VBR_QUALITY: - st->vbr_quality = (*(float*)ptr); - break; - case SPEEX_GET_VBR_QUALITY: - (*(float*)ptr) = st->vbr_quality; - break; - case SPEEX_SET_QUALITY: - { - int quality = (*(int*)ptr); - if (quality < 0) - quality = 0; - if (quality > 10) - quality = 10; - st->submodeSelect = st->submodeID = ((SpeexNBMode*)(st->mode->mode))->quality_map[quality]; - } - break; - case SPEEX_SET_COMPLEXITY: - st->complexity = (*(int*)ptr); - if (st->complexity<1) - st->complexity=1; - break; - case SPEEX_GET_COMPLEXITY: - (*(int*)ptr) = st->complexity; - break; - case SPEEX_SET_BITRATE: - { - int i=10, rate, target; - target = (*(int*)ptr); - while (i>=0) - { - speex_encoder_ctl(st, SPEEX_SET_QUALITY, &i); - speex_encoder_ctl(st, SPEEX_GET_BITRATE, &rate); - if (rate <= target) - break; - i--; - } - } - break; - case SPEEX_GET_BITRATE: - if (st->submodes[st->submodeID]) - (*(int*)ptr) = st->sampling_rate*SUBMODE(bits_per_frame)/st->frameSize; - else - (*(int*)ptr) = st->sampling_rate*(NB_SUBMODE_BITS+1)/st->frameSize; - break; - case SPEEX_SET_SAMPLING_RATE: - st->sampling_rate = (*(int*)ptr); - break; - case SPEEX_GET_SAMPLING_RATE: - (*(int*)ptr)=st->sampling_rate; - break; - case SPEEX_RESET_STATE: - { - int i; - st->bounded_pitch = 1; - st->first = 1; - for (i=0;ilpcSize;i++) - st->lsp[i]=(M_PI*((float)(i+1)))/(st->lpcSize+1); - for (i=0;ilpcSize;i++) - st->mem_sw[i]=st->mem_sw_whole[i]=st->mem_sp[i]=st->mem_exc[i]=0; - for (i=0;ibufSize;i++) - st->excBuf[i]=st->swBuf[i]=st->inBuf[i]=st->exc2Buf[i]=0; - } - break; - case SPEEX_GET_PI_GAIN: - { - int i; - float *g = (float*)ptr; - for (i=0;inbSubframes;i++) - g[i]=st->pi_gain[i]; - } - break; - case SPEEX_GET_EXC: - { - int i; - float *e = (float*)ptr; - for (i=0;iframeSize;i++) - e[i]=st->exc[i]; - } - break; - case SPEEX_GET_INNOV: - { - int i; - float *e = (float*)ptr; - for (i=0;iframeSize;i++) - e[i]=st->innov[i]; - } - break; - case SPEEX_GET_RELATIVE_QUALITY: - (*(float*)ptr)=st->relative_quality; - break; - default: - speex_warning_int("Unknown nb_ctl request: ", request); - return -1; - } - return 0; -} - -int nb_decoder_ctl(void *state, int request, void *ptr) -{ - DecState *st; - st=(DecState*)state; - switch(request) - { - case SPEEX_GET_LOW_MODE: - case SPEEX_GET_MODE: - (*(int*)ptr) = st->submodeID; - break; - case SPEEX_SET_ENH: - st->lpc_enh_enabled = *((int*)ptr); - break; - case SPEEX_GET_ENH: - *((int*)ptr) = st->lpc_enh_enabled; - break; - case SPEEX_GET_FRAME_SIZE: - (*(int*)ptr) = st->frameSize; - break; - case SPEEX_GET_BITRATE: - if (st->submodes[st->submodeID]) - (*(int*)ptr) = st->sampling_rate*SUBMODE(bits_per_frame)/st->frameSize; - else - (*(int*)ptr) = st->sampling_rate*(NB_SUBMODE_BITS+1)/st->frameSize; - break; - case SPEEX_SET_SAMPLING_RATE: - st->sampling_rate = (*(int*)ptr); - break; - case SPEEX_GET_SAMPLING_RATE: - (*(int*)ptr)=st->sampling_rate; - break; - case SPEEX_SET_HANDLER: - { - SpeexCallback *c = (SpeexCallback*)ptr; - st->speex_callbacks[c->callback_id].func=c->func; - st->speex_callbacks[c->callback_id].data=c->data; - st->speex_callbacks[c->callback_id].callback_id=c->callback_id; - } - break; - case SPEEX_SET_USER_HANDLER: - { - SpeexCallback *c = (SpeexCallback*)ptr; - st->user_callback.func=c->func; - st->user_callback.data=c->data; - st->user_callback.callback_id=c->callback_id; - } - break; - case SPEEX_RESET_STATE: - { - int i; - for (i=0;i<2*st->lpcSize;i++) - st->mem_sp[i]=0; - for (i=0;ibufSize;i++) - st->excBuf[i]=st->inBuf[i]=0; - } - break; - case SPEEX_GET_PI_GAIN: - { - int i; - float *g = (float*)ptr; - for (i=0;inbSubframes;i++) - g[i]=st->pi_gain[i]; - } - break; - case SPEEX_GET_EXC: - { - int i; - float *e = (float*)ptr; - for (i=0;iframeSize;i++) - e[i]=st->exc[i]; - } - break; - case SPEEX_GET_INNOV: - { - int i; - float *e = (float*)ptr; - for (i=0;iframeSize;i++) - e[i]=st->innov[i]; - } - break; - case SPEEX_GET_DTX_STATUS: - *((int*)ptr) = st->dtx_enabled; - break; - default: - speex_warning_int("Unknown nb_ctl request: ", request); - return -1; - } - return 0; -} diff --git a/engine/voice_codecs/speex/source/libspeex/nb_celp.h b/engine/voice_codecs/speex/source/libspeex/nb_celp.h deleted file mode 100644 index 7439f1c7..00000000 --- a/engine/voice_codecs/speex/source/libspeex/nb_celp.h +++ /dev/null @@ -1,202 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin */ -/** - @file nb_celp.h - @brief Narrowband CELP encoder/decoder -*/ -/* - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - -#ifndef NB_CELP_H -#define NB_CELP_H - -#include "modes.h" -#include "speex_bits.h" -#include "speex_callbacks.h" -#include "vbr.h" -#include "filters.h" - -/**Structure representing the full state of the narrowband encoder*/ -typedef struct EncState { - SpeexMode *mode; /**< Mode corresponding to the state */ - int first; /**< Is this the first frame? */ - int frameSize; /**< Size of frames */ - int subframeSize; /**< Size of sub-frames */ - int nbSubframes; /**< Number of sub-frames */ - int windowSize; /**< Analysis (LPC) window length */ - int lpcSize; /**< LPC order */ - int bufSize; /**< Buffer size */ - int min_pitch; /**< Minimum pitch value allowed */ - int max_pitch; /**< Maximum pitch value allowed */ - - int safe_pitch; /**< Don't use too large values for pitch (in case we lose a packet) */ - int bounded_pitch; /**< Next frame should not rely on previous frames for pitch */ - int ol_pitch; /**< Open-loop pitch */ - int ol_voiced; /**< Open-loop voiced/non-voiced decision */ - int *pitch; - float gamma1; /**< Perceptual filter: A(z/gamma1) */ - float gamma2; /**< Perceptual filter: A(z/gamma2) */ - float lag_factor; /**< Lag windowing Gaussian width */ - float lpc_floor; /**< Noise floor multiplier for A[0] in LPC analysis*/ - float preemph; /**< Pre-emphasis: P(z) = 1 - a*z^-1*/ - float pre_mem; /**< 1-element memory for pre-emphasis */ - float pre_mem2; /**< 1-element memory for pre-emphasis */ - char *stack; /**< Pseudo-stack allocation for temporary memory */ - float *inBuf; /**< Input buffer (original signal) */ - float *frame; /**< Start of original frame */ - float *excBuf; /**< Excitation buffer */ - float *exc; /**< Start of excitation frame */ - float *exc2Buf; /**< "Pitch enhanced" excitation */ - float *exc2; /**< "Pitch enhanced" excitation */ - float *swBuf; /**< Weighted signal buffer */ - float *sw; /**< Start of weighted signal frame */ - float *innov; /**< Innovation for the frame */ - float *window; /**< Temporary (Hanning) window */ - float *buf2; /**< 2nd temporary buffer */ - float *autocorr; /**< auto-correlation */ - float *lagWindow; /**< Window applied to auto-correlation */ - float *lpc; /**< LPCs for current frame */ - float *lsp; /**< LSPs for current frame */ - float *qlsp; /**< Quantized LSPs for current frame */ - float *old_lsp; /**< LSPs for previous frame */ - float *old_qlsp; /**< Quantized LSPs for previous frame */ - float *interp_lsp; /**< Interpolated LSPs */ - float *interp_qlsp; /**< Interpolated quantized LSPs */ - float *interp_lpc; /**< Interpolated LPCs */ - float *interp_qlpc; /**< Interpolated quantized LPCs */ - float *bw_lpc1; /**< LPCs after bandwidth expansion by gamma1 for perceptual weighting*/ - float *bw_lpc2; /**< LPCs after bandwidth expansion by gamma2 for perceptual weighting*/ - float *rc; /**< Reflection coefficients */ - float *mem_sp; /**< Filter memory for signal synthesis */ - float *mem_sw; /**< Filter memory for perceptually-weighted signal */ - float *mem_sw_whole; /**< Filter memory for perceptually-weighted signal (whole frame)*/ - float *mem_exc; /**< Filter memory for excitation (whole frame) */ - float *pi_gain; /**< Gain of LPC filter at theta=pi (fe/2) */ - - VBRState *vbr; /**< State of the VBR data */ - float vbr_quality; /**< Quality setting for VBR encoding */ - float relative_quality; /**< Relative quality that will be needed by VBR */ - int vbr_enabled; /**< 1 for enabling VBR, 0 otherwise */ - int vad_enabled; /**< 1 for enabling VAD, 0 otherwise */ - int dtx_enabled; /**< 1 for enabling DTX, 0 otherwise */ - int dtx_count; /**< Number of consecutive DTX frames */ - int abr_enabled; /**< ABR setting (in bps), 0 if off */ - float abr_drift; - float abr_drift2; - float abr_count; - int complexity; /**< Complexity setting (0-10 from least complex to most complex) */ - int sampling_rate; - - SpeexSubmode **submodes; /**< Sub-mode data */ - int submodeID; /**< Activated sub-mode */ - int submodeSelect; /**< Mode chosen by the user (may differ from submodeID if VAD is on) */ -} EncState; - -/**Structure representing the full state of the narrowband decoder*/ -typedef struct DecState { - SpeexMode *mode; /**< Mode corresponding to the state */ - int first; /**< Is this the first frame? */ - int count_lost; /**< Was the last frame lost? */ - int frameSize; /**< Size of frames */ - int subframeSize; /**< Size of sub-frames */ - int nbSubframes; /**< Number of sub-frames */ - int windowSize; /**< Analysis (LPC) window length */ - int lpcSize; /**< LPC order */ - int bufSize; /**< Buffer size */ - int min_pitch; /**< Minimum pitch value allowed */ - int max_pitch; /**< Maximum pitch value allowed */ - int sampling_rate; - float last_ol_gain; /**< Open-loop gain for previous frame */ - - - float gamma1; /**< Perceptual filter: A(z/gamma1) */ - float gamma2; /**< Perceptual filter: A(z/gamma2) */ - float preemph; /**< Pre-emphasis: P(z) = 1 - a*z^-1*/ - float pre_mem; /**< 1-element memory for pre-emphasis */ - char *stack; /**< Pseudo-stack allocation for temporary memory */ - float *inBuf; /**< Input buffer (original signal) */ - float *frame; /**< Start of original frame */ - float *excBuf; /**< Excitation buffer */ - float *exc; /**< Start of excitation frame */ - float *innov; /**< Innovation for the frame */ - float *qlsp; /**< Quantized LSPs for current frame */ - float *old_qlsp; /**< Quantized LSPs for previous frame */ - float *interp_qlsp; /**< Interpolated quantized LSPs */ - float *interp_qlpc; /**< Interpolated quantized LPCs */ - float *mem_sp; /**< Filter memory for synthesis signal */ - float *pi_gain; /**< Gain of LPC filter at theta=pi (fe/2) */ - int last_pitch; /**< Pitch of last correctly decoded frame */ - float last_pitch_gain; /**< Pitch gain of last correctly decoded frame */ - float pitch_gain_buf[3]; /**< Pitch gain of last decoded frames */ - int pitch_gain_buf_idx; /**< Tail of the buffer */ - - SpeexSubmode **submodes; /**< Sub-mode data */ - int submodeID; /**< Activated sub-mode */ - int lpc_enh_enabled; /**< 1 when LPC enhancer is on, 0 otherwise */ - CombFilterMem *comb_mem; - SpeexCallback speex_callbacks[SPEEX_MAX_CALLBACKS]; - - SpeexCallback user_callback; - - /*Vocoder data*/ - float voc_m1; - float voc_m2; - float voc_mean; - int voc_offset; - - int dtx_enabled; -} DecState; - -/** Initializes encoder state*/ -void *nb_encoder_init(SpeexMode *m); - -/** De-allocates encoder state resources*/ -void nb_encoder_destroy(void *state); - -/** Encodes one frame*/ -int nb_encode(void *state, float *in, SpeexBits *bits); - - -/** Initializes decoder state*/ -void *nb_decoder_init(SpeexMode *m); - -/** De-allocates decoder state resources*/ -void nb_decoder_destroy(void *state); - -/** Decodes one frame*/ -int nb_decode(void *state, SpeexBits *bits, float *out); - -/** ioctl-like function for controlling a narrowband encoder */ -int nb_encoder_ctl(void *state, int request, void *ptr); - -/** ioctl-like function for controlling a narrowband decoder */ -int nb_decoder_ctl(void *state, int request, void *ptr); - - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/quant_lsp.c b/engine/voice_codecs/speex/source/libspeex/quant_lsp.c deleted file mode 100644 index c9d797e4..00000000 --- a/engine/voice_codecs/speex/source/libspeex/quant_lsp.c +++ /dev/null @@ -1,311 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: quant_lsp.c - LSP vector quantization - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include "quant_lsp.h" -#include - -extern int lsp_nb_vqid[64]; -static float quant_weight[MAX_LSP_SIZE]; - -/* Note: x is modified*/ -static int lsp_quant(float *x, signed char *cdbk, int nbVec, int nbDim) -{ - int i,j; - float dist, tmp; - float best_dist=0; - int best_id=0; - signed char *ptr=cdbk; - for (i=0;i tmp2 ? tmp1 : tmp2; - } - for (i=0;i tmp2 ? tmp1 : tmp2; - } - - for (i=0;i tmp2 ? tmp1 : tmp2; - } - - for (i=0;i -#include "sb_celp.h" -#include "stdlib.h" -#include "filters.h" -#include "lpc.h" -#include "lsp.h" -#include "stack_alloc.h" -#include "cb_search.h" -#include "quant_lsp.h" -#include "vq.h" -#include "ltp.h" -#include "misc.h" - -#ifndef M_PI -#define M_PI 3.14159265358979323846 /* pi */ -#endif - -#define sqr(x) ((x)*(x)) - -#define SUBMODE(x) st->submodes[st->submodeID]->x - -#define QMF_ORDER 64 -static float h0[64] = { - 3.596189e-05, -0.0001123515, - -0.0001104587, 0.0002790277, - 0.0002298438, -0.0005953563, - -0.0003823631, 0.00113826, - 0.0005308539, -0.001986177, - -0.0006243724, 0.003235877, - 0.0005743159, -0.004989147, - -0.0002584767, 0.007367171, - -0.0004857935, -0.01050689, - 0.001894714, 0.01459396, - -0.004313674, -0.01994365, - 0.00828756, 0.02716055, - -0.01485397, -0.03764973, - 0.026447, 0.05543245, - -0.05095487, -0.09779096, - 0.1382363, 0.4600981, - 0.4600981, 0.1382363, - -0.09779096, -0.05095487, - 0.05543245, 0.026447, - -0.03764973, -0.01485397, - 0.02716055, 0.00828756, - -0.01994365, -0.004313674, - 0.01459396, 0.001894714, - -0.01050689, -0.0004857935, - 0.007367171, -0.0002584767, - -0.004989147, 0.0005743159, - 0.003235877, -0.0006243724, - -0.001986177, 0.0005308539, - 0.00113826, -0.0003823631, - -0.0005953563, 0.0002298438, - 0.0002790277, -0.0001104587, - -0.0001123515, 3.596189e-05 -}; - -static float h1[64] = { - 3.596189e-05, 0.0001123515, - -0.0001104587, -0.0002790277, - 0.0002298438, 0.0005953563, - -0.0003823631, -0.00113826, - 0.0005308539, 0.001986177, - -0.0006243724, -0.003235877, - 0.0005743159, 0.004989147, - -0.0002584767, -0.007367171, - -0.0004857935, 0.01050689, - 0.001894714, -0.01459396, - -0.004313674, 0.01994365, - 0.00828756, -0.02716055, - -0.01485397, 0.03764973, - 0.026447, -0.05543245, - -0.05095487, 0.09779096, - 0.1382363, -0.4600981, - 0.4600981, -0.1382363, - -0.09779096, 0.05095487, - 0.05543245, -0.026447, - -0.03764973, 0.01485397, - 0.02716055, -0.00828756, - -0.01994365, 0.004313674, - 0.01459396, -0.001894714, - -0.01050689, 0.0004857935, - 0.007367171, 0.0002584767, - -0.004989147, -0.0005743159, - 0.003235877, 0.0006243724, - -0.001986177, -0.0005308539, - 0.00113826, 0.0003823631, - -0.0005953563, -0.0002298438, - 0.0002790277, 0.0001104587, - -0.0001123515, -3.596189e-05 -}; - -void *sb_encoder_init(SpeexMode *m) -{ - int i; - SBEncState *st; - SpeexSBMode *mode; - - st = (SBEncState*)speex_alloc(sizeof(SBEncState)+8000*sizeof(float)); - st->mode = m; - mode = (SpeexSBMode*)m->mode; - - st->stack = ((char*)st) + sizeof(SBEncState); - - st->st_low = speex_encoder_init(mode->nb_mode); - st->full_frame_size = 2*mode->frameSize; - st->frame_size = mode->frameSize; - st->subframeSize = mode->subframeSize; - st->nbSubframes = mode->frameSize/mode->subframeSize; - st->windowSize = st->frame_size*3/2; - st->lpcSize=mode->lpcSize; - st->bufSize=mode->bufSize; - - st->submodes=mode->submodes; - st->submodeSelect = st->submodeID=mode->defaultSubmode; - - i=9; - speex_encoder_ctl(st->st_low, SPEEX_SET_QUALITY, &i); - - st->lag_factor = mode->lag_factor; - st->lpc_floor = mode->lpc_floor; - st->gamma1=mode->gamma1; - st->gamma2=mode->gamma2; - st->first=1; - - st->x0d=PUSH(st->stack, st->frame_size, float); - st->x1d=PUSH(st->stack, st->frame_size, float); - st->high=PUSH(st->stack, st->full_frame_size, float); - st->y0=PUSH(st->stack, st->full_frame_size, float); - st->y1=PUSH(st->stack, st->full_frame_size, float); - - st->h0_mem=PUSH(st->stack, QMF_ORDER, float); - st->h1_mem=PUSH(st->stack, QMF_ORDER, float); - st->g0_mem=PUSH(st->stack, QMF_ORDER, float); - st->g1_mem=PUSH(st->stack, QMF_ORDER, float); - - st->buf=PUSH(st->stack, st->windowSize, float); - st->excBuf=PUSH(st->stack, st->bufSize, float); - st->exc = st->excBuf + st->bufSize - st->windowSize; - - st->res=PUSH(st->stack, st->frame_size, float); - st->sw=PUSH(st->stack, st->frame_size, float); - st->target=PUSH(st->stack, st->frame_size, float); - /*Asymmetric "pseudo-Hamming" window*/ - { - int part1, part2; - part1 = st->subframeSize*7/2; - part2 = st->subframeSize*5/2; - st->window = PUSH(st->stack, st->windowSize, float); - for (i=0;iwindow[i]=.54-.46*cos(M_PI*i/part1); - for (i=0;iwindow[part1+i]=.54+.46*cos(M_PI*i/part2); - } - - st->lagWindow = PUSH(st->stack, st->lpcSize+1, float); - for (i=0;ilpcSize+1;i++) - st->lagWindow[i]=exp(-.5*sqr(2*M_PI*st->lag_factor*i)); - - st->rc = PUSH(st->stack, st->lpcSize, float); - st->autocorr = PUSH(st->stack, st->lpcSize+1, float); - st->lpc = PUSH(st->stack, st->lpcSize+1, float); - st->bw_lpc1 = PUSH(st->stack, st->lpcSize+1, float); - st->bw_lpc2 = PUSH(st->stack, st->lpcSize+1, float); - st->lsp = PUSH(st->stack, st->lpcSize, float); - st->qlsp = PUSH(st->stack, st->lpcSize, float); - st->old_lsp = PUSH(st->stack, st->lpcSize, float); - st->old_qlsp = PUSH(st->stack, st->lpcSize, float); - st->interp_lsp = PUSH(st->stack, st->lpcSize, float); - st->interp_qlsp = PUSH(st->stack, st->lpcSize, float); - st->interp_lpc = PUSH(st->stack, st->lpcSize+1, float); - st->interp_qlpc = PUSH(st->stack, st->lpcSize+1, float); - st->pi_gain = PUSH(st->stack, st->nbSubframes, float); - - st->mem_sp = PUSH(st->stack, st->lpcSize, float); - st->mem_sp2 = PUSH(st->stack, st->lpcSize, float); - st->mem_sw = PUSH(st->stack, st->lpcSize, float); - - st->vbr_quality = 8; - st->vbr_enabled = 0; - st->vad_enabled = 0; - st->abr_enabled = 0; - st->relative_quality=0; - - st->complexity=2; - speex_decoder_ctl(st->st_low, SPEEX_GET_SAMPLING_RATE, &st->sampling_rate); - st->sampling_rate*=2; - - return st; -} - -void sb_encoder_destroy(void *state) -{ - SBEncState *st=(SBEncState*)state; - - speex_encoder_destroy(st->st_low); - - speex_free(st); -} - - -int sb_encode(void *state, float *in, SpeexBits *bits) -{ - SBEncState *st; - int i, roots, sub; - char *stack; - float *mem, *innov, *syn_resp; - float *low_pi_gain, *low_exc, *low_innov; - SpeexSBMode *mode; - int dtx; - - st = (SBEncState*)state; - stack=st->stack; - mode = (SpeexSBMode*)(st->mode->mode); - - /* Compute the two sub-bands by filtering with h0 and h1*/ - qmf_decomp(in, h0, st->x0d, st->x1d, st->full_frame_size, QMF_ORDER, st->h0_mem, stack); - - /* Encode the narrowband part*/ - speex_encode(st->st_low, st->x0d, bits); - - /* High-band buffering / sync with low band */ - for (i=0;iwindowSize-st->frame_size;i++) - st->high[i] = st->high[st->frame_size+i]; - for (i=0;iframe_size;i++) - st->high[st->windowSize-st->frame_size+i]=st->x1d[i]; - - speex_move(st->excBuf, st->excBuf+st->frame_size, (st->bufSize-st->frame_size)*sizeof(float)); - - - low_pi_gain = PUSH(stack, st->nbSubframes, float); - low_exc = PUSH(stack, st->frame_size, float); - low_innov = PUSH(stack, st->frame_size, float); - speex_encoder_ctl(st->st_low, SPEEX_GET_PI_GAIN, low_pi_gain); - speex_encoder_ctl(st->st_low, SPEEX_GET_EXC, low_exc); - speex_encoder_ctl(st->st_low, SPEEX_GET_INNOV, low_innov); - - speex_encoder_ctl(st->st_low, SPEEX_GET_LOW_MODE, &dtx); - - if (dtx==0) - dtx=1; - else - dtx=0; - - /* Start encoding the high-band */ - for (i=0;iwindowSize;i++) - st->buf[i] = st->high[i] * st->window[i]; - - /* Compute auto-correlation */ - _spx_autocorr(st->buf, st->autocorr, st->lpcSize+1, st->windowSize); - - st->autocorr[0] += 1; /* prevents NANs */ - st->autocorr[0] *= st->lpc_floor; /* Noise floor in auto-correlation domain */ - /* Lag windowing: equivalent to filtering in the power-spectrum domain */ - for (i=0;ilpcSize+1;i++) - st->autocorr[i] *= st->lagWindow[i]; - - /* Levinson-Durbin */ - wld(st->lpc+1, st->autocorr, st->rc, st->lpcSize); - st->lpc[0]=1; - - /* LPC to LSPs (x-domain) transform */ - roots=lpc_to_lsp (st->lpc, st->lpcSize, st->lsp, 15, 0.2, stack); - if (roots!=st->lpcSize) - { - roots = lpc_to_lsp (st->lpc, st->lpcSize, st->lsp, 11, 0.02, stack); - if (roots!=st->lpcSize) { - /*If we can't find all LSP's, do some damage control and use a flat filter*/ - for (i=0;ilpcSize;i++) - { - st->lsp[i]=cos(M_PI*((float)(i+1))/(st->lpcSize+1)); - } - } - } - - /* x-domain to angle domain*/ - for (i=0;ilpcSize;i++) - st->lsp[i] = acos(st->lsp[i]); - - /* VBR code */ - if ((st->vbr_enabled || st->vad_enabled) && !dtx) - { - float e_low=0, e_high=0; - float ratio; - if (st->abr_enabled) - { - float qual_change=0; - if (st->abr_drift2 * st->abr_drift > 0) - { - /* Only adapt if long-term and short-term drift are the same sign */ - qual_change = -.00001*st->abr_drift/(1+st->abr_count); - if (qual_change>.1) - qual_change=.1; - if (qual_change<-.1) - qual_change=-.1; - } - st->vbr_quality += qual_change; - if (st->vbr_quality>10) - st->vbr_quality=10; - if (st->vbr_quality<0) - st->vbr_quality=0; - } - - - for (i=0;iframe_size;i++) - { - e_low += st->x0d[i]* st->x0d[i]; - e_high += st->high[i]* st->high[i]; - } - ratio = log((1+e_high)/(1+e_low)); - speex_encoder_ctl(st->st_low, SPEEX_GET_RELATIVE_QUALITY, &st->relative_quality); - if (ratio<-4) - ratio=-4; - if (ratio>2) - ratio=2; - /*if (ratio>-2)*/ - if (st->vbr_enabled) - { - int modeid; - modeid = mode->nb_modes-1; - st->relative_quality+=1.0*(ratio+2); - if (st->relative_quality<-1) - st->relative_quality=-1; - while (modeid) - { - int v1; - float thresh; - v1=(int)floor(st->vbr_quality); - if (v1==10) - thresh = mode->vbr_thresh[modeid][v1]; - else - thresh = (st->vbr_quality-v1) * mode->vbr_thresh[modeid][v1+1] + - (1+v1-st->vbr_quality) * mode->vbr_thresh[modeid][v1]; - if (st->relative_quality >= thresh) - break; - modeid--; - } - speex_encoder_ctl(state, SPEEX_SET_HIGH_MODE, &modeid); - if (st->abr_enabled) - { - int bitrate; - speex_encoder_ctl(state, SPEEX_GET_BITRATE, &bitrate); - st->abr_drift+=(bitrate-st->abr_enabled); - st->abr_drift2 = .95*st->abr_drift2 + .05*(bitrate-st->abr_enabled); - st->abr_count += 1.0; - } - - } else { - /* VAD only */ - int modeid; - if (st->relative_quality<2.0) - modeid=1; - else - modeid=st->submodeSelect; - /*speex_encoder_ctl(state, SPEEX_SET_MODE, &mode);*/ - st->submodeID=modeid; - - } - /*fprintf (stderr, "%f %f\n", ratio, low_qual);*/ - } - - speex_bits_pack(bits, 1, 1); - if (dtx) - speex_bits_pack(bits, 0, SB_SUBMODE_BITS); - else - speex_bits_pack(bits, st->submodeID, SB_SUBMODE_BITS); - - /* If null mode (no transmission), just set a couple things to zero*/ - if (dtx || st->submodes[st->submodeID] == NULL) - { - for (i=0;iframe_size;i++) - st->exc[i]=st->sw[i]=0; - - for (i=0;ilpcSize;i++) - st->mem_sw[i]=0; - st->first=1; - - /* Final signal synthesis from excitation */ - iir_mem2(st->exc, st->interp_qlpc, st->high, st->subframeSize, st->lpcSize, st->mem_sp); - -#ifndef RELEASE - - /* Reconstruct the original */ - fir_mem_up(st->x0d, h0, st->y0, st->full_frame_size, QMF_ORDER, st->g0_mem, stack); - fir_mem_up(st->high, h1, st->y1, st->full_frame_size, QMF_ORDER, st->g1_mem, stack); - - for (i=0;ifull_frame_size;i++) - in[i]=2*(st->y0[i]-st->y1[i]); -#endif - - if (dtx) - return 0; - else - return 1; - } - - - /* LSP quantization */ - SUBMODE(lsp_quant)(st->lsp, st->qlsp, st->lpcSize, bits); - - if (st->first) - { - for (i=0;ilpcSize;i++) - st->old_lsp[i] = st->lsp[i]; - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - } - - mem=PUSH(stack, st->lpcSize, float); - syn_resp=PUSH(stack, st->subframeSize, float); - innov = PUSH(stack, st->subframeSize, float); - - for (sub=0;subnbSubframes;sub++) - { - float *exc, *sp, *res, *target, *sw, tmp, filter_ratio; - int offset; - float rl, rh, eh=0, el=0; - int fold; - - offset = st->subframeSize*sub; - sp=st->high+offset; - exc=st->exc+offset; - res=st->res+offset; - target=st->target+offset; - sw=st->sw+offset; - - /* LSP interpolation (quantized and unquantized) */ - tmp = (1.0 + sub)/st->nbSubframes; - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = (1-tmp)*st->old_lsp[i] + tmp*st->lsp[i]; - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = (1-tmp)*st->old_qlsp[i] + tmp*st->qlsp[i]; - - lsp_enforce_margin(st->interp_lsp, st->lpcSize, .05); - lsp_enforce_margin(st->interp_qlsp, st->lpcSize, .05); - - /* Compute interpolated LPCs (quantized and unquantized) */ - for (i=0;ilpcSize;i++) - st->interp_lsp[i] = cos(st->interp_lsp[i]); - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = cos(st->interp_qlsp[i]); - - lsp_to_lpc(st->interp_lsp, st->interp_lpc, st->lpcSize,stack); - lsp_to_lpc(st->interp_qlsp, st->interp_qlpc, st->lpcSize, stack); - - bw_lpc(st->gamma1, st->interp_lpc, st->bw_lpc1, st->lpcSize); - bw_lpc(st->gamma2, st->interp_lpc, st->bw_lpc2, st->lpcSize); - - /* Compute mid-band (4000 Hz for wideband) response of low-band and high-band - filters */ - rl=rh=0; - tmp=1; - st->pi_gain[sub]=0; - for (i=0;i<=st->lpcSize;i++) - { - rh += tmp*st->interp_qlpc[i]; - tmp = -tmp; - st->pi_gain[sub]+=st->interp_qlpc[i]; - } - rl = low_pi_gain[sub]; - rl=1/(fabs(rl)+.01); - rh=1/(fabs(rh)+.01); - /* Compute ratio, will help predict the gain */ - filter_ratio=fabs(.01+rh)/(.01+fabs(rl)); - - fold = filter_ratio<5; - /*printf ("filter_ratio %f\n", filter_ratio);*/ - fold=0; - - /* Compute "real excitation" */ - fir_mem2(sp, st->interp_qlpc, exc, st->subframeSize, st->lpcSize, st->mem_sp2); - /* Compute energy of low-band and high-band excitation */ - for (i=0;isubframeSize;i++) - eh+=sqr(exc[i]); - - if (!SUBMODE(innovation_quant)) {/* 1 for spectral folding excitation, 0 for stochastic */ - float g; - /*speex_bits_pack(bits, 1, 1);*/ - for (i=0;isubframeSize;i++) - el+=sqr(low_innov[offset+i]); - - /* Gain to use if we want to use the low-band excitation for high-band */ - g=eh/(.01+el); - g=sqrt(g); - - g *= filter_ratio; - /*print_vec(&g, 1, "gain factor");*/ - /* Gain quantization */ - { - int quant = (int) floor(.5 + 10 + 8.0 * log((g+.0001))); - /*speex_warning_int("tata", quant);*/ - if (quant<0) - quant=0; - if (quant>31) - quant=31; - speex_bits_pack(bits, quant, 5); - g= .1*exp(quant/9.4); - } - /*printf ("folding gain: %f\n", g);*/ - g /= filter_ratio; - - } else { - float gc, scale, scale_1; - - for (i=0;isubframeSize;i++) - el+=sqr(low_exc[offset+i]); - /*speex_bits_pack(bits, 0, 1);*/ - - gc = sqrt(1+eh)*filter_ratio/sqrt((1+el)*st->subframeSize); - { - int qgc = (int)floor(.5+3.7*(log(gc)+2)); - if (qgc<0) - qgc=0; - if (qgc>15) - qgc=15; - speex_bits_pack(bits, qgc, 4); - gc = exp((1/3.7)*qgc-2); - } - - scale = gc*sqrt(1+el)/filter_ratio; - scale_1 = 1/scale; - - for (i=0;isubframeSize;i++) - exc[i]=0; - exc[0]=1; - syn_percep_zero(exc, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, syn_resp, st->subframeSize, st->lpcSize, stack); - - /* Reset excitation */ - for (i=0;isubframeSize;i++) - exc[i]=0; - - /* Compute zero response (ringing) of A(z/g1) / ( A(z/g2) * Aq(z) ) */ - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sp[i]; - iir_mem2(exc, st->interp_qlpc, exc, st->subframeSize, st->lpcSize, mem); - - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sw[i]; - filter_mem2(exc, st->bw_lpc1, st->bw_lpc2, res, st->subframeSize, st->lpcSize, mem); - - /* Compute weighted signal */ - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sw[i]; - filter_mem2(sp, st->bw_lpc1, st->bw_lpc2, sw, st->subframeSize, st->lpcSize, mem); - - /* Compute target signal */ - for (i=0;isubframeSize;i++) - target[i]=sw[i]-res[i]; - - for (i=0;isubframeSize;i++) - exc[i]=0; - - - for (i=0;isubframeSize;i++) - target[i]*=scale_1; - - /* Reset excitation */ - for (i=0;isubframeSize;i++) - innov[i]=0; - - /*print_vec(target, st->subframeSize, "\ntarget");*/ - SUBMODE(innovation_quant)(target, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, - SUBMODE(innovation_params), st->lpcSize, st->subframeSize, - innov, syn_resp, bits, stack, (st->complexity+1)>>1); - /*print_vec(target, st->subframeSize, "after");*/ - - for (i=0;isubframeSize;i++) - exc[i] += innov[i]*scale; - - if (SUBMODE(double_codebook)) { - char *tmp_stack=stack; - float *innov2 = PUSH(tmp_stack, st->subframeSize, float); - for (i=0;isubframeSize;i++) - innov2[i]=0; - for (i=0;isubframeSize;i++) - target[i]*=2.5; - SUBMODE(innovation_quant)(target, st->interp_qlpc, st->bw_lpc1, st->bw_lpc2, - SUBMODE(innovation_params), st->lpcSize, st->subframeSize, - innov2, syn_resp, bits, tmp_stack, (st->complexity+1)>>1); - for (i=0;isubframeSize;i++) - innov2[i]*=scale*(1/2.5); - for (i=0;isubframeSize;i++) - exc[i] += innov2[i]; - } - - } - - /*Keep the previous memory*/ - for (i=0;ilpcSize;i++) - mem[i]=st->mem_sp[i]; - /* Final signal synthesis from excitation */ - iir_mem2(exc, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, st->mem_sp); - - /* Compute weighted signal again, from synthesized speech (not sure it's the right thing) */ - filter_mem2(sp, st->bw_lpc1, st->bw_lpc2, sw, st->subframeSize, st->lpcSize, st->mem_sw); - } - - -#ifndef RELEASE - - /* Reconstruct the original */ - fir_mem_up(st->x0d, h0, st->y0, st->full_frame_size, QMF_ORDER, st->g0_mem, stack); - fir_mem_up(st->high, h1, st->y1, st->full_frame_size, QMF_ORDER, st->g1_mem, stack); - - for (i=0;ifull_frame_size;i++) - in[i]=2*(st->y0[i]-st->y1[i]); -#endif - for (i=0;ilpcSize;i++) - st->old_lsp[i] = st->lsp[i]; - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - - st->first=0; - - return 1; -} - - - - - -void *sb_decoder_init(SpeexMode *m) -{ - SBDecState *st; - SpeexSBMode *mode; - st = (SBDecState*)speex_alloc(sizeof(SBDecState)+6000*sizeof(float)); - st->mode = m; - mode=(SpeexSBMode*)m->mode; - - st->stack = ((char*)st) + sizeof(SBDecState); - - - - st->st_low = speex_decoder_init(mode->nb_mode); - st->full_frame_size = 2*mode->frameSize; - st->frame_size = mode->frameSize; - st->subframeSize = mode->subframeSize; - st->nbSubframes = mode->frameSize/mode->subframeSize; - st->lpcSize=8; - speex_decoder_ctl(st->st_low, SPEEX_GET_SAMPLING_RATE, &st->sampling_rate); - st->sampling_rate*=2; - - st->submodes=mode->submodes; - st->submodeID=mode->defaultSubmode; - - st->first=1; - - - st->x0d=PUSH(st->stack, st->frame_size, float); - st->x1d=PUSH(st->stack, st->frame_size, float); - st->high=PUSH(st->stack, st->full_frame_size, float); - st->y0=PUSH(st->stack, st->full_frame_size, float); - st->y1=PUSH(st->stack, st->full_frame_size, float); - - st->h0_mem=PUSH(st->stack, QMF_ORDER, float); - st->h1_mem=PUSH(st->stack, QMF_ORDER, float); - st->g0_mem=PUSH(st->stack, QMF_ORDER, float); - st->g1_mem=PUSH(st->stack, QMF_ORDER, float); - - st->exc=PUSH(st->stack, st->frame_size, float); - - st->qlsp = PUSH(st->stack, st->lpcSize, float); - st->old_qlsp = PUSH(st->stack, st->lpcSize, float); - st->interp_qlsp = PUSH(st->stack, st->lpcSize, float); - st->interp_qlpc = PUSH(st->stack, st->lpcSize+1, float); - - st->pi_gain = PUSH(st->stack, st->nbSubframes, float); - st->mem_sp = PUSH(st->stack, 2*st->lpcSize, float); - - st->lpc_enh_enabled=0; - - return st; -} - -void sb_decoder_destroy(void *state) -{ - SBDecState *st; - st = (SBDecState*)state; - speex_decoder_destroy(st->st_low); - - speex_free(state); -} - -static void sb_decode_lost(SBDecState *st, float *out, int dtx, char *stack) -{ - int i; - float *awk1, *awk2, *awk3; - int saved_modeid=0; - - if (dtx) - { - saved_modeid=st->submodeID; - st->submodeID=1; - } else { - bw_lpc(0.99, st->interp_qlpc, st->interp_qlpc, st->lpcSize); - } - - st->first=1; - - awk1=PUSH(stack, st->lpcSize+1, float); - awk2=PUSH(stack, st->lpcSize+1, float); - awk3=PUSH(stack, st->lpcSize+1, float); - - if (st->lpc_enh_enabled) - { - float r=.9; - - float k1,k2,k3; - if (st->submodes[st->submodeID] != NULL) - { - k1=SUBMODE(lpc_enh_k1); - k2=SUBMODE(lpc_enh_k2); - } else { - k1=k2=.7; - } - k3=(1-(1-r*k1)/(1-r*k2))/r; - k3=k1-k2; - if (!st->lpc_enh_enabled) - { - k1=k2; - k3=0; - } - bw_lpc(k1, st->interp_qlpc, awk1, st->lpcSize); - bw_lpc(k2, st->interp_qlpc, awk2, st->lpcSize); - bw_lpc(k3, st->interp_qlpc, awk3, st->lpcSize); - /*fprintf (stderr, "%f %f %f\n", k1, k2, k3);*/ - } - - - /* Final signal synthesis from excitation */ - if (!dtx) - { - for (i=0;iframe_size;i++) - st->exc[i] *= .9; - } - - for (i=0;iframe_size;i++) - st->high[i]=st->exc[i]; - - if (st->lpc_enh_enabled) - { - /* Use enhanced LPC filter */ - filter_mem2(st->high, awk2, awk1, st->high, st->frame_size, st->lpcSize, - st->mem_sp+st->lpcSize); - filter_mem2(st->high, awk3, st->interp_qlpc, st->high, st->frame_size, st->lpcSize, - st->mem_sp); - } else { - /* Use regular filter */ - for (i=0;ilpcSize;i++) - st->mem_sp[st->lpcSize+i] = 0; - iir_mem2(st->high, st->interp_qlpc, st->high, st->frame_size, st->lpcSize, - st->mem_sp); - } - - /*iir_mem2(st->exc, st->interp_qlpc, st->high, st->frame_size, st->lpcSize, st->mem_sp);*/ - - /* Reconstruct the original */ - fir_mem_up(st->x0d, h0, st->y0, st->full_frame_size, QMF_ORDER, st->g0_mem, stack); - fir_mem_up(st->high, h1, st->y1, st->full_frame_size, QMF_ORDER, st->g1_mem, stack); - - for (i=0;ifull_frame_size;i++) - out[i]=2*(st->y0[i]-st->y1[i]); - - if (dtx) - { - st->submodeID=saved_modeid; - } - - return; -} - -int sb_decode(void *state, SpeexBits *bits, float *out) -{ - int i, sub; - SBDecState *st; - int wideband; - int ret; - char *stack; - float *low_pi_gain, *low_exc, *low_innov; - float *awk1, *awk2, *awk3; - int dtx; - SpeexSBMode *mode; - - st = (SBDecState*)state; - stack=st->stack; - mode = (SpeexSBMode*)(st->mode->mode); - - /* Decode the low-band */ - ret = speex_decode(st->st_low, bits, st->x0d); - - speex_decoder_ctl(st->st_low, SPEEX_GET_DTX_STATUS, &dtx); - - /* If error decoding the narrowband part, propagate error */ - if (ret!=0) - { - return ret; - } - - if (!bits) - { - sb_decode_lost(st, out, dtx, stack); - return 0; - } - - /*Check "wideband bit"*/ - if (speex_bits_remaining(bits)>0) - wideband = speex_bits_peek(bits); - else - wideband = 0; - if (wideband) - { - /*Regular wideband frame, read the submode*/ - wideband = speex_bits_unpack_unsigned(bits, 1); - st->submodeID = speex_bits_unpack_unsigned(bits, SB_SUBMODE_BITS); - } else - { - /*Was a narrowband frame, set "null submode"*/ - st->submodeID = 0; - } - if (st->submodeID != 0 && st->submodes[st->submodeID] == NULL) - { - speex_warning("Invalid mode encountered: corrupted stream?"); - return -2; - } - - /* If null mode (no transmission), just set a couple things to zero*/ - if (st->submodes[st->submodeID] == NULL) - { - if (dtx) - { - sb_decode_lost(st, out, 1, stack); - return 0; - } - - for (i=0;iframe_size;i++) - st->exc[i]=0; - - st->first=1; - - /* Final signal synthesis from excitation */ - iir_mem2(st->exc, st->interp_qlpc, st->high, st->frame_size, st->lpcSize, st->mem_sp); - - fir_mem_up(st->x0d, h0, st->y0, st->full_frame_size, QMF_ORDER, st->g0_mem, stack); - fir_mem_up(st->high, h1, st->y1, st->full_frame_size, QMF_ORDER, st->g1_mem, stack); - - for (i=0;ifull_frame_size;i++) - out[i]=2*(st->y0[i]-st->y1[i]); - - return 0; - - } - - for (i=0;iframe_size;i++) - st->exc[i]=0; - - low_pi_gain = PUSH(stack, st->nbSubframes, float); - low_exc = PUSH(stack, st->frame_size, float); - low_innov = PUSH(stack, st->frame_size, float); - speex_decoder_ctl(st->st_low, SPEEX_GET_PI_GAIN, low_pi_gain); - speex_decoder_ctl(st->st_low, SPEEX_GET_EXC, low_exc); - speex_decoder_ctl(st->st_low, SPEEX_GET_INNOV, low_innov); - - SUBMODE(lsp_unquant)(st->qlsp, st->lpcSize, bits); - - if (st->first) - { - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - } - - awk1=PUSH(stack, st->lpcSize+1, float); - awk2=PUSH(stack, st->lpcSize+1, float); - awk3=PUSH(stack, st->lpcSize+1, float); - - for (sub=0;subnbSubframes;sub++) - { - float *exc, *sp, tmp, filter_ratio, el=0; - int offset; - float rl=0,rh=0; - - offset = st->subframeSize*sub; - sp=st->high+offset; - exc=st->exc+offset; - - /* LSP interpolation */ - tmp = (1.0 + sub)/st->nbSubframes; - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = (1-tmp)*st->old_qlsp[i] + tmp*st->qlsp[i]; - - lsp_enforce_margin(st->interp_qlsp, st->lpcSize, .05); - - /* LSPs to x-domain */ - for (i=0;ilpcSize;i++) - st->interp_qlsp[i] = cos(st->interp_qlsp[i]); - - /* LSP to LPC */ - lsp_to_lpc(st->interp_qlsp, st->interp_qlpc, st->lpcSize, stack); - - - if (st->lpc_enh_enabled) - { - float r=.9; - - float k1,k2,k3; - k1=SUBMODE(lpc_enh_k1); - k2=SUBMODE(lpc_enh_k2); - k3=(1-(1-r*k1)/(1-r*k2))/r; - k3=k1-k2; - if (!st->lpc_enh_enabled) - { - k1=k2; - k3=0; - } - bw_lpc(k1, st->interp_qlpc, awk1, st->lpcSize); - bw_lpc(k2, st->interp_qlpc, awk2, st->lpcSize); - bw_lpc(k3, st->interp_qlpc, awk3, st->lpcSize); - /*fprintf (stderr, "%f %f %f\n", k1, k2, k3);*/ - } - - - /* Calculate reponse ratio between the low and high filter in the middle - of the band (4000 Hz) */ - - tmp=1; - st->pi_gain[sub]=0; - for (i=0;i<=st->lpcSize;i++) - { - rh += tmp*st->interp_qlpc[i]; - tmp = -tmp; - st->pi_gain[sub]+=st->interp_qlpc[i]; - } - rl = low_pi_gain[sub]; - rl=1/(fabs(rl)+.01); - rh=1/(fabs(rh)+.01); - filter_ratio=fabs(.01+rh)/(.01+fabs(rl)); - - - for (i=0;isubframeSize;i++) - exc[i]=0; - if (!SUBMODE(innovation_unquant)) - { - float g; - int quant; - - for (i=0;isubframeSize;i++) - el+=sqr(low_innov[offset+i]); - quant = speex_bits_unpack_unsigned(bits, 5); - g= exp(((float)quant-10)/8.0); - - /*printf ("unquant folding gain: %f\n", g);*/ - g /= filter_ratio; - - /* High-band excitation using the low-band excitation and a gain */ - for (i=0;isubframeSize;i++) - exc[i]=mode->folding_gain*g*low_innov[offset+i]; - /*speex_rand_vec(mode->folding_gain*g*sqrt(el/st->subframeSize), exc, st->subframeSize);*/ - } else { - float gc, scale; - int qgc = speex_bits_unpack_unsigned(bits, 4); - for (i=0;isubframeSize;i++) - el+=sqr(low_exc[offset+i]); - - - gc = exp((1/3.7)*qgc-2); - - scale = gc*sqrt(1+el)/filter_ratio; - - - SUBMODE(innovation_unquant)(exc, SUBMODE(innovation_params), st->subframeSize, - bits, stack); - for (i=0;isubframeSize;i++) - exc[i]*=scale; - - if (SUBMODE(double_codebook)) { - char *tmp_stack=stack; - float *innov2 = PUSH(tmp_stack, st->subframeSize, float); - for (i=0;isubframeSize;i++) - innov2[i]=0; - SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), st->subframeSize, - bits, tmp_stack); - for (i=0;isubframeSize;i++) - innov2[i]*=scale*(1/2.5); - for (i=0;isubframeSize;i++) - exc[i] += innov2[i]; - } - - } - - for (i=0;isubframeSize;i++) - sp[i]=exc[i]; - if (st->lpc_enh_enabled) - { - /* Use enhanced LPC filter */ - filter_mem2(sp, awk2, awk1, sp, st->subframeSize, st->lpcSize, - st->mem_sp+st->lpcSize); - filter_mem2(sp, awk3, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, - st->mem_sp); - } else { - /* Use regular filter */ - for (i=0;ilpcSize;i++) - st->mem_sp[st->lpcSize+i] = 0; - iir_mem2(sp, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, - st->mem_sp); - } - /*iir_mem2(exc, st->interp_qlpc, sp, st->subframeSize, st->lpcSize, st->mem_sp);*/ - - } - - fir_mem_up(st->x0d, h0, st->y0, st->full_frame_size, QMF_ORDER, st->g0_mem, stack); - fir_mem_up(st->high, h1, st->y1, st->full_frame_size, QMF_ORDER, st->g1_mem, stack); - - for (i=0;ifull_frame_size;i++) - out[i]=2*(st->y0[i]-st->y1[i]); - - for (i=0;ilpcSize;i++) - st->old_qlsp[i] = st->qlsp[i]; - - st->first=0; - - return 0; -} - - -int sb_encoder_ctl(void *state, int request, void *ptr) -{ - SBEncState *st; - st=(SBEncState*)state; - switch(request) - { - case SPEEX_GET_FRAME_SIZE: - (*(int*)ptr) = st->full_frame_size; - break; - case SPEEX_SET_HIGH_MODE: - st->submodeSelect = st->submodeID = (*(int*)ptr); - break; - case SPEEX_SET_LOW_MODE: - speex_encoder_ctl(st->st_low, SPEEX_SET_LOW_MODE, ptr); - break; - case SPEEX_SET_DTX: - speex_encoder_ctl(st->st_low, SPEEX_SET_DTX, ptr); - break; - case SPEEX_GET_DTX: - speex_encoder_ctl(st->st_low, SPEEX_GET_DTX, ptr); - break; - case SPEEX_GET_LOW_MODE: - speex_encoder_ctl(st->st_low, SPEEX_GET_LOW_MODE, ptr); - break; - case SPEEX_SET_MODE: - speex_encoder_ctl(st, SPEEX_SET_QUALITY, ptr); - break; - case SPEEX_SET_VBR: - st->vbr_enabled = (*(int*)ptr); - speex_encoder_ctl(st->st_low, SPEEX_SET_VBR, ptr); - break; - case SPEEX_GET_VBR: - (*(int*)ptr) = st->vbr_enabled; - break; - case SPEEX_SET_VAD: - st->vad_enabled = (*(int*)ptr); - speex_encoder_ctl(st->st_low, SPEEX_SET_VAD, ptr); - break; - case SPEEX_GET_VAD: - (*(int*)ptr) = st->vad_enabled; - break; - case SPEEX_SET_VBR_QUALITY: - { - int q; - float qual = (*(float*)ptr)+.6; - st->vbr_quality = (*(float*)ptr); - if (qual>10) - qual=10; - q=(int)floor(.5+*(float*)ptr); - if (q>10) - q=10; - speex_encoder_ctl(st->st_low, SPEEX_SET_VBR_QUALITY, &qual); - speex_encoder_ctl(state, SPEEX_SET_QUALITY, &q); - break; - } - case SPEEX_SET_ABR: - st->abr_enabled = (*(int*)ptr); - st->vbr_enabled = 1; - speex_encoder_ctl(st->st_low, SPEEX_SET_VBR, &st->vbr_enabled); - { - int i=10, rate, target; - float vbr_qual; - target = (*(int*)ptr); - while (i>=0) - { - speex_encoder_ctl(st, SPEEX_SET_QUALITY, &i); - speex_encoder_ctl(st, SPEEX_GET_BITRATE, &rate); - if (rate <= target) - break; - i--; - } - vbr_qual=i; - if (vbr_qual<0) - vbr_qual=0; - speex_encoder_ctl(st, SPEEX_SET_VBR_QUALITY, &vbr_qual); - st->abr_count=0; - st->abr_drift=0; - st->abr_drift2=0; - } - - break; - case SPEEX_GET_ABR: - (*(int*)ptr) = st->abr_enabled; - break; - case SPEEX_SET_QUALITY: - { - int nb_qual; - int quality = (*(int*)ptr); - if (quality < 0) - quality = 0; - if (quality > 10) - quality = 10; - st->submodeSelect = st->submodeID = ((SpeexSBMode*)(st->mode->mode))->quality_map[quality]; - nb_qual = ((SpeexSBMode*)(st->mode->mode))->low_quality_map[quality]; - speex_encoder_ctl(st->st_low, SPEEX_SET_MODE, &nb_qual); - } - break; - case SPEEX_SET_COMPLEXITY: - speex_encoder_ctl(st->st_low, SPEEX_SET_COMPLEXITY, ptr); - st->complexity = (*(int*)ptr); - if (st->complexity<1) - st->complexity=1; - break; - case SPEEX_GET_COMPLEXITY: - (*(int*)ptr) = st->complexity; - break; - case SPEEX_SET_BITRATE: - { - int i=10, rate, target; - target = (*(int*)ptr); - while (i>=0) - { - speex_encoder_ctl(st, SPEEX_SET_QUALITY, &i); - speex_encoder_ctl(st, SPEEX_GET_BITRATE, &rate); - if (rate <= target) - break; - i--; - } - } - break; - case SPEEX_GET_BITRATE: - speex_encoder_ctl(st->st_low, request, ptr); - /*fprintf (stderr, "before: %d\n", (*(int*)ptr));*/ - if (st->submodes[st->submodeID]) - (*(int*)ptr) += st->sampling_rate*SUBMODE(bits_per_frame)/st->full_frame_size; - else - (*(int*)ptr) += st->sampling_rate*(SB_SUBMODE_BITS+1)/st->full_frame_size; - /*fprintf (stderr, "after: %d\n", (*(int*)ptr));*/ - break; - case SPEEX_SET_SAMPLING_RATE: - { - int tmp=(*(int*)ptr); - st->sampling_rate = tmp; - tmp>>=1; - speex_encoder_ctl(st->st_low, SPEEX_SET_SAMPLING_RATE, &tmp); - } - break; - case SPEEX_GET_SAMPLING_RATE: - (*(int*)ptr)=st->sampling_rate; - break; - case SPEEX_RESET_STATE: - { - int i; - st->first = 1; - for (i=0;ilpcSize;i++) - st->lsp[i]=(M_PI*((float)(i+1)))/(st->lpcSize+1); - for (i=0;ilpcSize;i++) - st->mem_sw[i]=st->mem_sp[i]=st->mem_sp2[i]=0; - for (i=0;ibufSize;i++) - st->excBuf[i]=0; - for (i=0;ih0_mem[i]=st->h1_mem[i]=st->g0_mem[i]=st->g1_mem[i]=0; - } - break; - case SPEEX_GET_PI_GAIN: - { - int i; - float *g = (float*)ptr; - for (i=0;inbSubframes;i++) - g[i]=st->pi_gain[i]; - } - break; - case SPEEX_GET_EXC: - { - int i; - float *e = (float*)ptr; - for (i=0;ifull_frame_size;i++) - e[i]=0; - for (i=0;iframe_size;i++) - e[2*i]=2*st->exc[i]; - } - break; - case SPEEX_GET_INNOV: - { - int i; - float *e = (float*)ptr; - for (i=0;ifull_frame_size;i++) - e[i]=0; - for (i=0;iframe_size;i++) - e[2*i]=2*st->exc[i]; - } - break; - case SPEEX_GET_RELATIVE_QUALITY: - (*(float*)ptr)=st->relative_quality; - break; - default: - speex_warning_int("Unknown nb_ctl request: ", request); - return -1; - } - return 0; -} - -int sb_decoder_ctl(void *state, int request, void *ptr) -{ - SBDecState *st; - st=(SBDecState*)state; - switch(request) - { - case SPEEX_GET_LOW_MODE: - speex_decoder_ctl(st->st_low, SPEEX_GET_LOW_MODE, ptr); - break; - case SPEEX_GET_FRAME_SIZE: - (*(int*)ptr) = st->full_frame_size; - break; - case SPEEX_SET_ENH: - speex_decoder_ctl(st->st_low, request, ptr); - st->lpc_enh_enabled = *((int*)ptr); - break; - case SPEEX_GET_BITRATE: - speex_decoder_ctl(st->st_low, request, ptr); - if (st->submodes[st->submodeID]) - (*(int*)ptr) += st->sampling_rate*SUBMODE(bits_per_frame)/st->full_frame_size; - else - (*(int*)ptr) += st->sampling_rate*(SB_SUBMODE_BITS+1)/st->full_frame_size; - break; - case SPEEX_SET_SAMPLING_RATE: - { - int tmp=(*(int*)ptr); - st->sampling_rate = tmp; - tmp>>=1; - speex_decoder_ctl(st->st_low, SPEEX_SET_SAMPLING_RATE, &tmp); - } - break; - case SPEEX_GET_SAMPLING_RATE: - (*(int*)ptr)=st->sampling_rate; - break; - case SPEEX_SET_HANDLER: - speex_decoder_ctl(st->st_low, SPEEX_SET_HANDLER, ptr); - break; - case SPEEX_SET_USER_HANDLER: - speex_decoder_ctl(st->st_low, SPEEX_SET_USER_HANDLER, ptr); - break; - case SPEEX_RESET_STATE: - { - int i; - for (i=0;i<2*st->lpcSize;i++) - st->mem_sp[i]=0; - for (i=0;ih0_mem[i]=st->h1_mem[i]=st->g0_mem[i]=st->g1_mem[i]=0; - } - break; - case SPEEX_GET_PI_GAIN: - { - int i; - float *g = (float*)ptr; - for (i=0;inbSubframes;i++) - g[i]=st->pi_gain[i]; - } - break; - case SPEEX_GET_EXC: - { - int i; - float *e = (float*)ptr; - for (i=0;ifull_frame_size;i++) - e[i]=0; - for (i=0;iframe_size;i++) - e[2*i]=2*st->exc[i]; - } - break; - case SPEEX_GET_INNOV: - { - int i; - float *e = (float*)ptr; - for (i=0;ifull_frame_size;i++) - e[i]=0; - for (i=0;iframe_size;i++) - e[2*i]=2*st->exc[i]; - } - break; - case SPEEX_GET_DTX_STATUS: - speex_decoder_ctl(st->st_low, SPEEX_GET_DTX_STATUS, ptr); - break; - default: - speex_warning_int("Unknown nb_ctl request: ", request); - return -1; - } - return 0; -} diff --git a/engine/voice_codecs/speex/source/libspeex/sb_celp.h b/engine/voice_codecs/speex/source/libspeex/sb_celp.h deleted file mode 100644 index 64b23eea..00000000 --- a/engine/voice_codecs/speex/source/libspeex/sb_celp.h +++ /dev/null @@ -1,167 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin */ -/** - @file sb_celp.h - @brief Sub-band CELP mode used for wideband encoding -*/ -/* - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - -#ifndef SB_CELP_H -#define SB_CELP_H - -#include "modes.h" -#include "speex_bits.h" -#include "nb_celp.h" - -/**Structure representing the full state of the sub-band encoder*/ -typedef struct SBEncState { - SpeexMode *mode; /**< Pointer to the mode (containing for vtable info) */ - void *st_low; /**< State of the low-band (narrowband) encoder */ - int full_frame_size; /**< Length of full-band frames*/ - int frame_size; /**< Length of high-band frames*/ - int subframeSize; /**< Length of high-band sub-frames*/ - int nbSubframes; /**< Number of high-band sub-frames*/ - int windowSize; /**< Length of high-band LPC window*/ - int lpcSize; /**< Order of high-band LPC analysis */ - int bufSize; /**< Buffer size */ - int first; /**< First frame? */ - float lag_factor; /**< Lag-windowing control parameter */ - float lpc_floor; /**< Controls LPC analysis noise floor */ - float gamma1; /**< Perceptual weighting coef 1 */ - float gamma2; /**< Perceptual weighting coef 2 */ - - char *stack; /**< Temporary allocation stack */ - float *x0d, *x1d; /**< QMF filter signals*/ - float *high; /**< High-band signal (buffer) */ - float *y0, *y1; /**< QMF synthesis signals */ - float *h0_mem, *h1_mem, *g0_mem, *g1_mem; /**< QMF memories */ - - float *excBuf; /**< High-band excitation */ - float *exc; /**< High-band excitation (for QMF only)*/ - float *buf; /**< Temporary buffer */ - float *res; /**< Zero-input response (ringing) */ - float *sw; /**< Perceptually weighted signal */ - float *target; /**< Weighted target signal (analysis by synthesis) */ - float *window; /**< LPC analysis window */ - float *lagWindow; /**< Auto-correlation window */ - float *autocorr; /**< Auto-correlation (for LPC analysis) */ - float *rc; /**< Reflection coefficients (unused) */ - float *lpc; /**< LPC coefficients */ - float *lsp; /**< LSP coefficients */ - float *qlsp; /**< Quantized LSPs */ - float *old_lsp; /**< LSPs of previous frame */ - float *old_qlsp; /**< Quantized LSPs of previous frame */ - float *interp_lsp; /**< Interpolated LSPs for current sub-frame */ - float *interp_qlsp; /**< Interpolated quantized LSPs for current sub-frame */ - float *interp_lpc; /**< Interpolated LPCs for current sub-frame */ - float *interp_qlpc; /**< Interpolated quantized LPCs for current sub-frame */ - float *bw_lpc1; /**< Bandwidth-expanded version of LPCs (#1) */ - float *bw_lpc2; /**< Bandwidth-expanded version of LPCs (#2) */ - - float *mem_sp; /**< Synthesis signal memory */ - float *mem_sp2; - float *mem_sw; /**< Perceptual signal memory */ - float *pi_gain; - - float vbr_quality; /**< Quality setting for VBR encoding */ - int vbr_enabled; /**< 1 for enabling VBR, 0 otherwise */ - int abr_enabled; /**< ABR setting (in bps), 0 if off */ - float abr_drift; - float abr_drift2; - float abr_count; - int vad_enabled; /**< 1 for enabling VAD, 0 otherwise */ - float relative_quality; - - SpeexSubmode **submodes; - int submodeID; - int submodeSelect; - int complexity; - int sampling_rate; - -} SBEncState; - - -/**Structure representing the full state of the sub-band decoder*/ -typedef struct SBDecState { - SpeexMode *mode; /**< Pointer to the mode (containing for vtable info) */ - void *st_low; /**< State of the low-band (narrowband) encoder */ - int full_frame_size; - int frame_size; - int subframeSize; - int nbSubframes; - int lpcSize; - int first; - int sampling_rate; - int lpc_enh_enabled; - - char *stack; - float *x0d, *x1d; - float *high; - float *y0, *y1; - float *h0_mem, *h1_mem, *g0_mem, *g1_mem; - - float *exc; - float *qlsp; - float *old_qlsp; - float *interp_qlsp; - float *interp_qlpc; - - float *mem_sp; - float *pi_gain; - - SpeexSubmode **submodes; - int submodeID; -} SBDecState; - - -/**Initializes encoder state*/ -void *sb_encoder_init(SpeexMode *m); - -/**De-allocates encoder state resources*/ -void sb_encoder_destroy(void *state); - -/**Encodes one frame*/ -int sb_encode(void *state, float *in, SpeexBits *bits); - - -/**Initializes decoder state*/ -void *sb_decoder_init(SpeexMode *m); - -/**De-allocates decoder state resources*/ -void sb_decoder_destroy(void *state); - -/**Decodes one frame*/ -int sb_decode(void *state, SpeexBits *bits, float *out); - -int sb_encoder_ctl(void *state, int request, void *ptr); - -int sb_decoder_ctl(void *state, int request, void *ptr); - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/speex_callbacks.c b/engine/voice_codecs/speex/source/libspeex/speex_callbacks.c deleted file mode 100644 index 3b486965..00000000 --- a/engine/voice_codecs/speex/source/libspeex/speex_callbacks.c +++ /dev/null @@ -1,135 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File speex_callbacks.c - Callback handling and in-band signalling - - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - -#include "speex_callbacks.h" -#include "misc.h" - -int speex_inband_handler(SpeexBits *bits, SpeexCallback *callback_list, void *state) -{ - int id; - SpeexCallback *callback; - /*speex_bits_advance(bits, 5);*/ - id=speex_bits_unpack_unsigned(bits, 4); - callback = callback_list+id; - - if (callback->func) - { - return callback->func(bits, state, callback->data); - } else - /*If callback is not registered, skip the right number of bits*/ - { - int adv; - if (id<2) - adv = 1; - else if (id<8) - adv = 4; - else if (id<10) - adv = 8; - else if (id<12) - adv = 16; - else if (id<14) - adv = 32; - else - adv = 64; - speex_bits_advance(bits, adv); - } - return 0; -} - -int speex_std_mode_request_handler(SpeexBits *bits, void *state, void *data) -{ - int m; - m = speex_bits_unpack_unsigned(bits, 4); - speex_encoder_ctl(data, SPEEX_SET_MODE, &m); - return 0; -} - -int speex_std_low_mode_request_handler(SpeexBits *bits, void *state, void *data) -{ - int m; - m = speex_bits_unpack_unsigned(bits, 4); - speex_encoder_ctl(data, SPEEX_SET_LOW_MODE, &m); - return 0; -} - -int speex_std_high_mode_request_handler(SpeexBits *bits, void *state, void *data) -{ - int m; - m = speex_bits_unpack_unsigned(bits, 4); - speex_encoder_ctl(data, SPEEX_SET_HIGH_MODE, &m); - return 0; -} - -int speex_std_vbr_request_handler(SpeexBits *bits, void *state, void *data) -{ - int vbr; - vbr = speex_bits_unpack_unsigned(bits, 1); - speex_encoder_ctl(data, SPEEX_SET_VBR, &vbr); - return 0; -} - -int speex_std_enh_request_handler(SpeexBits *bits, void *state, void *data) -{ - int enh; - enh = speex_bits_unpack_unsigned(bits, 1); - speex_decoder_ctl(data, SPEEX_SET_ENH, &enh); - return 0; -} - -int speex_std_vbr_quality_request_handler(SpeexBits *bits, void *state, void *data) -{ - int qual; - qual = speex_bits_unpack_unsigned(bits, 4); - speex_encoder_ctl(data, SPEEX_SET_VBR_QUALITY, &qual); - return 0; -} - - -int speex_std_char_handler(SpeexBits *bits, void *state, void *data) -{ - unsigned char ch; - ch = speex_bits_unpack_unsigned(bits, 8); - _speex_putc(ch, data); - return 0; -} - - - -/* Default handler for user callbacks: skip it */ -int speex_default_user_handler(SpeexBits *bits, void *state, void *data) -{ - int req_size = speex_bits_unpack_unsigned(bits, 4); - speex_bits_advance(bits, 5+8*req_size); - return 0; -} diff --git a/engine/voice_codecs/speex/source/libspeex/speex_header.c b/engine/voice_codecs/speex/source/libspeex/speex_header.c deleted file mode 100644 index 7ee1ef34..00000000 --- a/engine/voice_codecs/speex/source/libspeex/speex_header.c +++ /dev/null @@ -1,162 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: speex_header.c - Describes the Speex header - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - -#include "speex_header.h" -#include "misc.h" -#include "speex.h" - -#ifndef NULL -#define NULL 0 -#endif - -#define ENDIAN_SWITCH(x) {x=le_int(x);} - - -/* -typedef struct SpeexHeader { - char speex_string[8]; - char speex_version[SPEEX_HEADER_VERSION_LENGTH]; - int speex_version_id; - int header_size; - int rate; - int mode; - int mode_bitstream_version; - int nb_channels; - int bitrate; - int frame_size; - int vbr; - int frames_per_packet; - int extra_headers; - int reserved1; - int reserved2; -} SpeexHeader; -*/ - -void speex_init_header(SpeexHeader *header, int rate, int nb_channels, SpeexMode *m) -{ - int i; - char *h="Speex "; - /* - strncpy(header->speex_string, "Speex ", 8); - strncpy(header->speex_version, VERSION, SPEEX_HEADER_VERSION_LENGTH-1); - header->speex_version[SPEEX_HEADER_VERSION_LENGTH-1]=0; - */ - for (i=0;i<8;i++) - header->speex_string[i]=h[i]; - for (i=0;ispeex_version[i]=VERSION[i]; - for (;ispeex_version[i]=0; - - header->speex_version_id = 1; - header->header_size = sizeof(SpeexHeader); - - header->rate = rate; - header->mode = m->modeID; - header->mode_bitstream_version = m->bitstream_version; - if (m->modeID<0) - speex_warning("This mode is meant to be used alone"); - header->nb_channels = nb_channels; - header->bitrate = -1; - speex_mode_query(m, SPEEX_MODE_FRAME_SIZE, &header->frame_size); - header->vbr = 0; - - header->frames_per_packet = 0; - header->extra_headers = 0; - header->reserved1 = 0; - header->reserved2 = 0; -} - -char *speex_header_to_packet(SpeexHeader *header, int *size) -{ - SpeexHeader *le_header; - le_header = (SpeexHeader*)speex_alloc(sizeof(SpeexHeader)); - - speex_move(le_header, header, sizeof(SpeexHeader)); - - /*Make sure everything is now little-endian*/ - ENDIAN_SWITCH(le_header->speex_version_id); - ENDIAN_SWITCH(le_header->header_size); - ENDIAN_SWITCH(le_header->rate); - ENDIAN_SWITCH(le_header->mode); - ENDIAN_SWITCH(le_header->mode_bitstream_version); - ENDIAN_SWITCH(le_header->nb_channels); - ENDIAN_SWITCH(le_header->bitrate); - ENDIAN_SWITCH(le_header->frame_size); - ENDIAN_SWITCH(le_header->vbr); - ENDIAN_SWITCH(le_header->frames_per_packet); - ENDIAN_SWITCH(le_header->extra_headers); - - *size = sizeof(SpeexHeader); - return (char *)le_header; -} - -SpeexHeader *speex_packet_to_header(char *packet, int size) -{ - int i; - SpeexHeader *le_header; - char *h = "Speex "; - for (i=0;i<8;i++) - if (packet[i]!=h[i]) - { - speex_warning ("This doesn't look like a Speex file"); - return NULL; - } - - /*FIXME: Do we allow larger headers?*/ - if (size < sizeof(SpeexHeader)) - { - speex_warning("Speex header too small"); - return NULL; - } - - le_header = (SpeexHeader*)speex_alloc(sizeof(SpeexHeader)); - - speex_move(le_header, packet, sizeof(SpeexHeader)); - - /*Make sure everything is converted correctly from little-endian*/ - ENDIAN_SWITCH(le_header->speex_version_id); - ENDIAN_SWITCH(le_header->header_size); - ENDIAN_SWITCH(le_header->rate); - ENDIAN_SWITCH(le_header->mode); - ENDIAN_SWITCH(le_header->mode_bitstream_version); - ENDIAN_SWITCH(le_header->nb_channels); - ENDIAN_SWITCH(le_header->bitrate); - ENDIAN_SWITCH(le_header->frame_size); - ENDIAN_SWITCH(le_header->vbr); - ENDIAN_SWITCH(le_header->frames_per_packet); - ENDIAN_SWITCH(le_header->extra_headers); - - return le_header; - -} diff --git a/engine/voice_codecs/speex/source/libspeex/stack_alloc.h b/engine/voice_codecs/speex/source/libspeex/stack_alloc.h deleted file mode 100644 index 2390b69d..00000000 --- a/engine/voice_codecs/speex/source/libspeex/stack_alloc.h +++ /dev/null @@ -1,62 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: stack_alloc.h - - Temporary memory allocation on stack - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef STACK_ALLOC_H -#define STACK_ALLOC_H - -#if 0 - -/*Aligns the stack to a 'size' boundary */ -#define ALIGN(stack, size) (stack=(void*)((((int)stack)+((size)-1)) & (-(size)))) -/*Aligns the stack to a 'size' boundary minus k */ -#define ALIGN_1(stack, size, k) (stack=(void*)(((((int)stack)+((size)-1+(k))) & (-(size)))-(k))) - -/* Allocates 'size' elements of type 'type' on the stack */ -#define PUSH(stack, size, type) (ALIGN(stack,sizeof(type)),stack=(void*)(((int)stack)+((size)*sizeof(type))),(type*)(((int)stack)-((size)*sizeof(type)))) - - -#else - -/*Aligns the stack to a 'size' boundary */ -#define ALIGN(stack, size) ((stack) += ((size) - (int)(stack)) & ((size) - 1)) - -/* Allocates 'size' elements of type 'type' on the stack */ -#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) - -/* Allocates a struct stack */ -#define PUSHS(stack, type) (ALIGN((stack),sizeof(long)),(stack)+=(sizeof(type)),(type*)((stack)-(sizeof(type)))) - - -#endif - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/stereo.c b/engine/voice_codecs/speex/source/libspeex/stereo.c deleted file mode 100644 index e033ce54..00000000 --- a/engine/voice_codecs/speex/source/libspeex/stereo.c +++ /dev/null @@ -1,122 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: stereo.c - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include "speex_stereo.h" -#include "speex_callbacks.h" -#include "vq.h" -#include - -/*float e_ratio_quant[4] = {1, 1.26, 1.587, 2};*/ -static float e_ratio_quant[4] = {.25, .315, .397, .5}; - -void speex_encode_stereo(float *data, int frame_size, SpeexBits *bits) -{ - int i, tmp; - float e_left=0, e_right=0, e_tot=0; - float balance, e_ratio; - for (i=0;i0) - speex_bits_pack(bits, 0, 1); - else - speex_bits_pack(bits, 1, 1); - balance=floor(.5+fabs(balance)); - if (balance>30) - balance=31; - - speex_bits_pack(bits, (int)balance, 5); - - /*Quantize energy ratio*/ - tmp=vq_index(&e_ratio, e_ratio_quant, 1, 4); - speex_bits_pack(bits, tmp, 2); -} - -void speex_decode_stereo(float *data, int frame_size, SpeexStereoState *stereo) -{ - float balance, e_ratio; - int i; - float e_tot=0, e_left, e_right, e_sum; - - balance=stereo->balance; - e_ratio=stereo->e_ratio; - for (i=frame_size-1;i>=0;i--) - { - e_tot += data[i]*data[i]; - } - e_sum=e_tot/e_ratio; - e_left = e_sum*balance / (1+balance); - e_right = e_sum-e_left; - - e_left = sqrt(e_left/(e_tot+.01)); - e_right = sqrt(e_right/(e_tot+.01)); - - for (i=frame_size-1;i>=0;i--) - { - float ftmp=data[i]; - stereo->smooth_left = .98*stereo->smooth_left + .02*e_left; - stereo->smooth_right = .98*stereo->smooth_right + .02*e_right; - data[2*i] = stereo->smooth_left*ftmp; - data[2*i+1] = stereo->smooth_right*ftmp; - } -} - -int speex_std_stereo_request_handler(SpeexBits *bits, void *state, void *data) -{ - SpeexStereoState *stereo; - float sign=1; - int tmp; - - stereo = (SpeexStereoState*)data; - if (speex_bits_unpack_unsigned(bits, 1)) - sign=-1; - tmp = speex_bits_unpack_unsigned(bits, 5); - stereo->balance = exp(sign*.25*tmp); - - tmp = speex_bits_unpack_unsigned(bits, 2); - stereo->e_ratio = e_ratio_quant[tmp]; - - return 0; -} diff --git a/engine/voice_codecs/speex/source/libspeex/testenc.c b/engine/voice_codecs/speex/source/libspeex/testenc.c deleted file mode 100644 index 7033ef4a..00000000 --- a/engine/voice_codecs/speex/source/libspeex/testenc.c +++ /dev/null @@ -1,127 +0,0 @@ -#include "speex.h" -#include -#include -#include "speex_callbacks.h" - -#define FRAME_SIZE 160 -#include -int main(int argc, char **argv) -{ - char *inFile, *outFile, *bitsFile; - FILE *fin, *fout, *fbits=NULL; - short in[FRAME_SIZE]; - float input[FRAME_SIZE], bak[FRAME_SIZE], bak2[FRAME_SIZE]; - char cbits[200]; - int nbBits; - int i; - void *st; - void *dec; - SpeexBits bits; - int tmp; - int bitCount=0; - SpeexCallback callback; - - for (i=0;i32000) - input[i]=32000; - else if (input[i]<-32000) - input[i]=-32000; - } - speex_bits_reset(&bits); - for (i=0;i -#include - -#define FRAME_SIZE 640 -#include -int main(int argc, char **argv) -{ - char *inFile, *outFile, *bitsFile; - FILE *fin, *fout, *fbits=NULL; - short in[FRAME_SIZE]; - float input[FRAME_SIZE], bak[FRAME_SIZE], bak2[FRAME_SIZE]; - char cbits[200]; - int nbBits; - int i; - void *st; - void *dec; - SpeexBits bits; - int tmp; - int bitCount=0; - - for (i=0;i32000) - input[i]=32000; - else if (input[i]<-32000) - input[i]=-32000; - } - speex_bits_reset(&bits); - for (i=0;i -#include - -#define FRAME_SIZE 320 -#include -int main(int argc, char **argv) -{ - char *inFile, *outFile, *bitsFile; - FILE *fin, *fout, *fbits=NULL; - short in[FRAME_SIZE]; - float input[FRAME_SIZE], bak[FRAME_SIZE], bak2[FRAME_SIZE]; - char cbits[200]; - int nbBits; - int i; - void *st; - void *dec; - SpeexBits bits; - int tmp; - int bitCount=0; - - for (i=0;i32000) - input[i]=32000; - else if (input[i]<-32000) - input[i]=-32000; - } - speex_bits_reset(&bits); - for (i=0;i - - -#define sqr(x) ((x)*(x)) - -#define MIN_ENERGY 6000 -#define NOISE_POW .3 - - -float vbr_nb_thresh[9][11]={ - {-1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0}, /* CNG */ - { 3.5, 2.5, 2.0, 1.2, 0.5, 0.0, -0.5, -0.7, -0.8, -0.9, -1.0}, /* 2 kbps */ - {10.0, 6.5, 5.2, 4.5, 3.9, 3.5, 3.0, 2.5, 2.3, 1.8, 1.0}, /* 6 kbps */ - {11.0, 8.8, 7.5, 6.5, 5.0, 3.9, 3.9, 3.9, 3.5, 3.0, 1.0}, /* 8 kbps */ - {11.0, 11.0, 9.9, 9.0, 8.0, 7.0, 6.5, 6.0, 5.0, 4.0, 2.0}, /* 11 kbps */ - {11.0, 11.0, 11.0, 11.0, 9.5, 9.0, 8.0, 7.0, 6.5, 5.0, 3.0}, /* 15 kbps */ - {11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 9.5, 8.5, 8.0, 6.5, 4.0}, /* 18 kbps */ - {11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 9.8, 7.5, 5.5}, /* 24 kbps */ - { 8.0, 5.0, 3.7, 3.0, 2.5, 2.0, 1.8, 1.5, 1.0, 0.0, 0.0} /* 4 kbps */ -}; - - -float vbr_hb_thresh[5][11]={ - {-1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0}, /* silence */ - {-1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0}, /* 2 kbps */ - {11.0, 11.0, 9.5, 8.5, 7.5, 6.0, 5.0, 3.9, 3.0, 2.0, 1.0}, /* 6 kbps */ - {11.0, 11.0, 11.0, 11.0, 11.0, 9.5, 8.7, 7.8, 7.0, 6.5, 4.0}, /* 10 kbps */ - {11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 11.0, 9.8, 7.5, 5.5} /* 18 kbps */ -}; - -float vbr_uhb_thresh[2][11]={ - {-1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0, -1.0}, /* silence */ - { 3.9, 2.5, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, -1.0} /* 2 kbps */ -}; - -void vbr_init(VBRState *vbr) -{ - int i; - - vbr->average_energy=0; - vbr->last_energy=1; - vbr->accum_sum=0; - vbr->energy_alpha=.1; - vbr->soft_pitch=0; - vbr->last_pitch_coef=0; - vbr->last_quality=0; - - vbr->noise_accum = .05*pow(MIN_ENERGY, NOISE_POW); - vbr->noise_accum_count=.05; - vbr->noise_level=vbr->noise_accum/vbr->noise_accum_count; - vbr->consec_noise=0; - - - for (i=0;ilast_log_energy[i] = log(MIN_ENERGY); -} - - -/* - This function should analyse the signal and decide how critical the - coding error will be perceptually. The following factors should be - taken into account: - - -Attacks (positive energy derivative) should be coded with more bits - - -Stationary voiced segments should receive more bits - - -Segments with (very) low absolute energy should receive less bits (maybe - only shaped noise?) - - -DTX for near-zero energy? - - -Stationary fricative segments should have less bits - - -Temporal masking: when energy slope is decreasing, decrease the bit-rate - - -Decrease bit-rate for males (low pitch)? - - -(wideband only) less bits in the high-band when signal is very - non-stationary (harder to notice high-frequency noise)??? - -*/ - -float vbr_analysis(VBRState *vbr, float *sig, int len, int pitch, float pitch_coef) -{ - int i; - float ener=0, ener1=0, ener2=0; - float qual=7; - int va; - float log_energy; - float non_st=0; - float voicing; - float pow_ener; - - for (i=0;i>1;i++) - ener1 += sig[i]*sig[i]; - - for (i=len>>1;ilast_log_energy[i]); - non_st = non_st/(30*VBR_MEMORY_SIZE); - if (non_st>1) - non_st=1; - - voicing = 3*(pitch_coef-.4)*fabs(pitch_coef-.4); - vbr->average_energy = (1-vbr->energy_alpha)*vbr->average_energy + vbr->energy_alpha*ener; - vbr->noise_level=vbr->noise_accum/vbr->noise_accum_count; - pow_ener = pow(ener,NOISE_POW); - if (vbr->noise_accum_count<.06 && ener>MIN_ENERGY) - vbr->noise_accum = .05*pow_ener; - - if ((voicing<.3 && non_st < .2 && pow_ener < 1.2*vbr->noise_level) - || (voicing<.3 && non_st < .05 && pow_ener < 1.5*vbr->noise_level) - || (voicing<.4 && non_st < .05 && pow_ener < 1.2*vbr->noise_level) - || (voicing<0 && non_st < .05)) - { - float tmp; - va = 0; - vbr->consec_noise++; - if (pow_ener > 3*vbr->noise_level) - tmp = 3*vbr->noise_level; - else - tmp = pow_ener; - if (vbr->consec_noise>=4) - { - vbr->noise_accum = .95*vbr->noise_accum + .05*tmp; - vbr->noise_accum_count = .95*vbr->noise_accum_count + .05; - } - } else { - va = 1; - vbr->consec_noise=0; - } - - if (pow_ener < vbr->noise_level && ener>MIN_ENERGY) - { - vbr->noise_accum = .95*vbr->noise_accum + .05*pow_ener; - vbr->noise_accum_count = .95*vbr->noise_accum_count + .05; - } - - /* Checking for very low absolute energy */ - if (ener < 30000) - { - qual -= .7; - if (ener < 10000) - qual-=.7; - if (ener < 3000) - qual-=.7; - } else { - float short_diff, long_diff; - short_diff = log((ener+1)/(1+vbr->last_energy)); - long_diff = log((ener+1)/(1+vbr->average_energy)); - /*fprintf (stderr, "%f %f\n", short_diff, long_diff);*/ - - if (long_diff<-5) - long_diff=-5; - if (long_diff>2) - long_diff=2; - - if (long_diff>0) - qual += .6*long_diff; - if (long_diff<0) - qual += .5*long_diff; - if (short_diff>0) - { - if (short_diff>5) - short_diff=5; - qual += .5*short_diff; - } - /* Checking for energy increases */ - if (ener2 > 1.6*ener1) - qual += .5; - } - vbr->last_energy = ener; - vbr->soft_pitch = .6*vbr->soft_pitch + .4*pitch_coef; - qual += 2.2*((pitch_coef-.4) + (vbr->soft_pitch-.4)); - - if (qual < vbr->last_quality) - qual = .5*qual + .5*vbr->last_quality; - if (qual<4) - qual=4; - if (qual>10) - qual=10; - - /* - if (vbr->consec_noise>=2) - qual-=1.3; - if (vbr->consec_noise>=5) - qual-=1.3; - if (vbr->consec_noise>=12) - qual-=1.3; - */ - if (vbr->consec_noise>=3) - qual=4; - - if (vbr->consec_noise) - qual -= 1.0 * (log(3.0 + vbr->consec_noise)-log(3)); - if (qual<0) - qual=0; - - if (ener<60000) - { - if (vbr->consec_noise>2) - qual-=0.5*(log(3.0 + vbr->consec_noise)-log(3)); - if (ener<10000&&vbr->consec_noise>2) - qual-=0.5*(log(3.0 + vbr->consec_noise)-log(3)); - if (qual<0) - qual=0; - qual += .3*log(ener/60000.0); - } - if (qual<-1) - qual=-1; - - /*printf ("%f %f %f %f %d\n", qual, voicing, non_st, pow_ener/(.01+vbr->noise_level), va);*/ - - vbr->last_pitch_coef = pitch_coef; - vbr->last_quality = qual; - - for (i=VBR_MEMORY_SIZE-1;i>0;i--) - vbr->last_log_energy[i] = vbr->last_log_energy[i-1]; - vbr->last_log_energy[0] = log_energy; - - /*printf ("VBR: %f %f %f %d %f\n", (float)(log_energy-log(vbr->average_energy+MIN_ENERGY)), non_st, voicing, va, vbr->noise_level);*/ - - return qual; -} - -void vbr_destroy(VBRState *vbr) -{ -} diff --git a/engine/voice_codecs/speex/source/libspeex/vbr.h b/engine/voice_codecs/speex/source/libspeex/vbr.h deleted file mode 100644 index 405b47f6..00000000 --- a/engine/voice_codecs/speex/source/libspeex/vbr.h +++ /dev/null @@ -1,66 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: vbr.h - - VBR-related routines - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - -*/ - - -#ifndef VBR_H -#define VBR_H - -#define VBR_MEMORY_SIZE 5 - -extern float vbr_nb_thresh[9][11]; -extern float vbr_hb_thresh[5][11]; -extern float vbr_uhb_thresh[2][11]; - -typedef struct VBRState { - float energy_alpha; - float average_energy; - float last_energy; - float last_log_energy[VBR_MEMORY_SIZE]; - float accum_sum; - float last_pitch_coef; - float soft_pitch; - float last_quality; - float noise_level; - float noise_accum; - float noise_accum_count; - int consec_noise; -} VBRState; - -void vbr_init(VBRState *vbr); - -float vbr_analysis(VBRState *vbr, float *sig, int len, int pitch, float pitch_coef); - -void vbr_destroy(VBRState *vbr); - -#endif diff --git a/engine/voice_codecs/speex/source/libspeex/vq.c b/engine/voice_codecs/speex/source/libspeex/vq.c deleted file mode 100644 index ed36f7d2..00000000 --- a/engine/voice_codecs/speex/source/libspeex/vq.c +++ /dev/null @@ -1,116 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: vq.c - Vector quantization - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include "vq.h" - -/*Finds the index of the entry in a codebook that best matches the input*/ -int vq_index(float *in, float *codebook, int len, int entries) -{ - int i,j; - float min_dist=0; - int best_index=0; - for (i=0;i= 1) && (k > used || dist < best_dist[k-1]); k--) - { - best_dist[k]=best_dist[k-1]; - nbest[k] = nbest[k-1]; - } - best_dist[k]=dist; - nbest[k]=i; - used++; - } - } -} - -/*Finds the indices of the n-best entries in a codebook with sign*/ -void vq_nbest_sign(float *in, float *codebook, int len, int entries, float *E, int N, int *nbest, float *best_dist) -{ - int i,j,k, sign, used; - used=0; - for (i=0;i0) - { - sign=1; - dist=-dist; - } else - { - sign=0; - } - dist += .5*E[i]; - if (i= 1) && (k > used || dist < best_dist[k-1]); k--) - { - best_dist[k]=best_dist[k-1]; - nbest[k] = nbest[k-1]; - } - best_dist[k]=dist; - nbest[k]=i; - used++; - if (sign) - nbest[k]+=entries; - } - } -} diff --git a/engine/voice_codecs/speex/source/libspeex/vq.h b/engine/voice_codecs/speex/source/libspeex/vq.h deleted file mode 100644 index e82ba6c0..00000000 --- a/engine/voice_codecs/speex/source/libspeex/vq.h +++ /dev/null @@ -1,42 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin - File: vq.h - Vector quantization - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef VQ_H -#define VQ_H - -int vq_index(float *in, float *codebook, int len, int entries); - -void vq_nbest(float *in, float *codebook, int len, int entries, float *E, int N, int *nbest, float *best_dist); - -void vq_nbest_sign(float *in, float *codebook, int len, int entries, float *E, int N, int *nbest, float *best_dist); - -#endif diff --git a/engine/voice_codecs/speex/source/linux/libspeex.a b/engine/voice_codecs/speex/source/linux/libspeex.a deleted file mode 100644 index 6a126717..00000000 Binary files a/engine/voice_codecs/speex/source/linux/libspeex.a and /dev/null differ diff --git a/engine/voice_codecs/speex/source/ltmain.sh b/engine/voice_codecs/speex/source/ltmain.sh deleted file mode 100644 index 239bdefb..00000000 --- a/engine/voice_codecs/speex/source/ltmain.sh +++ /dev/null @@ -1,5130 +0,0 @@ -# ltmain.sh - Provide generalized library-building support services. -# NOTE: Changing this file will not affect anything until you rerun configure. -# -# Copyright (C) 1996, 1997, 1998, 1999, 2000, 2001 -# Free Software Foundation, Inc. -# Originally by Gordon Matzigkeit , 1996 -# -# This program is free software; you can redistribute it and/or modify -# it under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 2 of the License, or -# (at your option) any later version. -# -# This program is distributed in the hope that it will be useful, but -# WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -# General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. -# -# As a special exception to the GNU General Public License, if you -# distribute this file as part of a program that contains a -# configuration script generated by Autoconf, you may include it under -# the same distribution terms that you use for the rest of that program. - -# Check that we have a working $echo. -if test "X$1" = X--no-reexec; then - # Discard the --no-reexec flag, and continue. - shift -elif test "X$1" = X--fallback-echo; then - # Avoid inline document here, it may be left over - : -elif test "X`($echo '\t') 2>/dev/null`" = 'X\t'; then - # Yippee, $echo works! - : -else - # Restart under the correct shell, and then maybe $echo will work. - exec $SHELL "$0" --no-reexec ${1+"$@"} -fi - -if test "X$1" = X--fallback-echo; then - # used as fallback echo - shift - cat <&2 - echo "Fatal configuration error. 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then - # Transform all the library objects into standard objects. - compile_command=`$echo "X$compile_command" | $SP2NL | $Xsed -e "$lo2o" | $NL2SP` - finalize_command=`$echo "X$finalize_command" | $SP2NL | $Xsed -e "$lo2o" | $NL2SP` - fi - - dlsyms= - if test -n "$dlfiles$dlprefiles" || test "$dlself" != no; then - if test -n "$NM" && test -n "$global_symbol_pipe"; then - dlsyms="${outputname}S.c" - else - $echo "$modename: not configured to extract global symbols from dlpreopened files" 1>&2 - fi - fi - - if test -n "$dlsyms"; then - case $dlsyms in - "") ;; - *.c) - # Discover the nlist of each of the dlfiles. - nlist="$output_objdir/${outputname}.nm" - - $show "$rm $nlist ${nlist}S ${nlist}T" - $run $rm "$nlist" "${nlist}S" "${nlist}T" - - # Parse the name list into a source file. - $show "creating $output_objdir/$dlsyms" - - test -z "$run" && $echo > "$output_objdir/$dlsyms" "\ -/* $dlsyms - symbol resolution table for \`$outputname' dlsym emulation. */ -/* Generated by $PROGRAM - GNU $PACKAGE $VERSION$TIMESTAMP */ - -#ifdef __cplusplus -extern \"C\" { -#endif - -/* Prevent the only kind of declaration conflicts we can make. */ -#define lt_preloaded_symbols some_other_symbol - -/* External symbol declarations for the compiler. */\ -" - - if test "$dlself" = yes; 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RM-OPTIONS are options (such as \`-f') to be passed -to RM. - -If FILE is a libtool library, all the files associated with it are deleted. -Otherwise, only FILE itself is deleted using RM." - ;; - -*) - $echo "$modename: invalid operation mode \`$mode'" 1>&2 - $echo "$help" 1>&2 - exit 1 - ;; -esac - -echo -$echo "Try \`$modename --help' for more information about other modes." - -exit 0 - -# Local Variables: -# mode:shell-script -# sh-indentation:2 -# End: diff --git a/engine/voice_codecs/speex/source/missing b/engine/voice_codecs/speex/source/missing deleted file mode 100644 index 22e101ab..00000000 --- a/engine/voice_codecs/speex/source/missing +++ /dev/null @@ -1,198 +0,0 @@ -#! /bin/sh -# Common stub for a few missing GNU programs while installing. -# Copyright (C) 1996, 1997, 2001, 2002 Free Software Foundation, Inc. -# Franc,ois Pinard , 1996. - -# This program is free software; you can redistribute it and/or modify -# it under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 2, or (at your option) -# any later version. - -# This program is distributed in the hope that it will be useful, -# but WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -# GNU General Public License for more details. - -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA -# 02111-1307, USA. - -if test $# -eq 0; then - echo 1>&2 "Try \`$0 --help' for more information" - exit 1 -fi - -# In the cases where this matters, `missing' is being run in the -# srcdir already. -if test -f configure.in; then - configure_ac=configure.ac -else - configure_ac=configure.in -fi - -case "$1" in - - -h|--h|--he|--hel|--help) - echo "\ -$0 [OPTION]... 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Grab either from any GNU archive site." - file=`echo "$*" | sed -n 's/.*-o \([^ ]*\).*/\1/p'` - if test -z "$file"; then - file=`echo "$*" | sed 's/.* \([^ ]*\) *$/\1/'` - file=`sed -n '/^@setfilename/ { s/.* \([^ ]*\) *$/\1/; p; q; }' $file` - fi - touch $file - ;; - - *) - echo 1>&2 "\ -WARNING: \`$1' is needed, and you do not seem to have it handy on your - system. You might have modified some files without having the - proper tools for further handling them. Check the \`README' file, - it often tells you about the needed prerequirements for installing - this package. You may also peek at any GNU archive site, in case - some other package would contain this missing \`$1' program." - exit 1 - ;; -esac - -exit 0 diff --git a/engine/voice_codecs/speex/source/mkinstalldirs b/engine/voice_codecs/speex/source/mkinstalldirs deleted file mode 100644 index 4f58503e..00000000 --- a/engine/voice_codecs/speex/source/mkinstalldirs +++ /dev/null @@ -1,40 +0,0 @@ -#! /bin/sh -# mkinstalldirs --- make directory hierarchy -# Author: Noah Friedman -# Created: 1993-05-16 -# Public domain - -# $Id: mkinstalldirs,v 1.13 1999/01/05 03:18:55 bje Exp $ - -errstatus=0 - -for file -do - set fnord `echo ":$file" | sed -ne 's/^:\//#/;s/^://;s/\// /g;s/^#/\//;p'` - shift - - pathcomp= - for d - do - pathcomp="$pathcomp$d" - case "$pathcomp" in - -* ) pathcomp=./$pathcomp ;; - esac - - if test ! -d "$pathcomp"; then - echo "mkdir $pathcomp" - - mkdir "$pathcomp" || lasterr=$? - - if test ! -d "$pathcomp"; then - errstatus=$lasterr - fi - fi - - pathcomp="$pathcomp/" - done -done - -exit $errstatus - -# mkinstalldirs ends here diff --git a/engine/voice_codecs/speex/source/osx/libspeex.a b/engine/voice_codecs/speex/source/osx/libspeex.a deleted file mode 100644 index af9a8c94..00000000 Binary files a/engine/voice_codecs/speex/source/osx/libspeex.a and /dev/null differ diff --git a/engine/voice_codecs/speex/source/speex.sln b/engine/voice_codecs/speex/source/speex.sln deleted file mode 100644 index cede7d33..00000000 --- a/engine/voice_codecs/speex/source/speex.sln +++ /dev/null @@ -1,22 +0,0 @@ - -Microsoft Visual Studio Solution File, Format Version 12.00 -# Visual Studio 14 -VisualStudioVersion = 14.0.25420.1 -MinimumVisualStudioVersion = 10.0.40219.1 -Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "LibSpeex", "speex.vcxproj", "{60902B83-8B26-4DBD-B635-4CECF5F4744C}" -EndProject -Global - GlobalSection(SolutionConfigurationPlatforms) = preSolution - Debug|x86 = Debug|x86 - Release|x86 = Release|x86 - EndGlobalSection - GlobalSection(ProjectConfigurationPlatforms) = postSolution - {60902B83-8B26-4DBD-B635-4CECF5F4744C}.Debug|x86.ActiveCfg = Debug|Win32 - {60902B83-8B26-4DBD-B635-4CECF5F4744C}.Debug|x86.Build.0 = Debug|Win32 - {60902B83-8B26-4DBD-B635-4CECF5F4744C}.Release|x86.ActiveCfg = Release|Win32 - {60902B83-8B26-4DBD-B635-4CECF5F4744C}.Release|x86.Build.0 = Release|Win32 - EndGlobalSection - GlobalSection(SolutionProperties) = preSolution - HideSolutionNode = FALSE - EndGlobalSection -EndGlobal diff --git a/engine/voice_codecs/speex/source/speex.vcxproj b/engine/voice_codecs/speex/source/speex.vcxproj deleted file mode 100644 index b6da34be..00000000 --- a/engine/voice_codecs/speex/source/speex.vcxproj +++ /dev/null @@ -1,167 +0,0 @@ - - - - - Debug - Win32 - - - Release - Win32 - - - - LibSpeex - {60902B83-8B26-4DBD-B635-4CECF5F4744C} - LibSpeex - Win32Proj - 8.1 - - - - StaticLibrary - v140_xp - Unicode - true - - - StaticLibrary - v140_xp - Unicode - - - - - - - - - - - - - <_ProjectFileVersion>14.0.25431.1 - - - $(SolutionDir)$(Configuration)_2015\lib\ - $(SolutionDir)$(Configuration)_2015\$(ProjectName)\ - - - $(SolutionDir)$(Configuration)_2015\lib\ - $(SolutionDir)$(Configuration)_2015\$(ProjectName)\ - - - - Disabled - ../../speex/include;../../speex/win32;%(AdditionalIncludeDirectories) - WIN32;_DEBUG;_LIB;HAVE_CONFIG_H;M_PI=3.14159265358979323846;%(PreprocessorDefinitions);_CRT_SECURE_NO_WARNINGS - true - true - - EnableFastChecks - MultiThreadedDebug - true - Fast - false - - Level3 - true - ProgramDatabase - 4018;4244;%(DisableSpecificWarnings) - - - win32\libspeex\2015\debug\libspeex.lib - - - - - MaxSpeed - true - ../../speex/include;../../speex/win32;%(AdditionalIncludeDirectories) - WIN32;NDEBUG;_LIB;HAVE_CONFIG_H;M_PI=3.14159265358979323846;%(PreprocessorDefinitions);_CRT_SECURE_NO_WARNINGS - true - - MultiThreaded - true - Fast - false - - Level3 - true - ProgramDatabase - 4018;4244;%(DisableSpecificWarnings) - - - win32\libspeex\2015\release\libspeex.lib - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - \ No newline at end of file diff --git a/engine/voice_codecs/speex/source/win32/libspeex/2015/debug/libspeex.lib b/engine/voice_codecs/speex/source/win32/libspeex/2015/debug/libspeex.lib deleted file mode 100644 index aec22379..00000000 Binary files a/engine/voice_codecs/speex/source/win32/libspeex/2015/debug/libspeex.lib and /dev/null differ diff --git a/engine/voice_codecs/speex/source/win32/libspeex/2015/release/libspeex.lib b/engine/voice_codecs/speex/source/win32/libspeex/2015/release/libspeex.lib deleted file mode 100644 index 78ee73c9..00000000 Binary files a/engine/voice_codecs/speex/source/win32/libspeex/2015/release/libspeex.lib and /dev/null differ diff --git a/engine/voice_codecs/speex/source/win32/libspeex/release/libspeex.lib b/engine/voice_codecs/speex/source/win32/libspeex/release/libspeex.lib deleted file mode 100644 index 8be14a88..00000000 Binary files a/engine/voice_codecs/speex/source/win32/libspeex/release/libspeex.lib and /dev/null differ diff --git a/engine/voice_codecs/speex/source/libspeex/speex.h b/engine/voice_codecs/speex/speex/speex.h similarity index 62% rename from engine/voice_codecs/speex/source/libspeex/speex.h rename to engine/voice_codecs/speex/speex/speex.h index cafd0cda..28c4b44d 100644 --- a/engine/voice_codecs/speex/source/libspeex/speex.h +++ b/engine/voice_codecs/speex/speex/speex.h @@ -1,4 +1,4 @@ -/* Copyright (C) 2002 Jean-Marc Valin*/ +/* Copyright (C) 2002-2006 Jean-Marc Valin*/ /** @file speex.h @brief Describes the different modes of the codec @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -35,7 +35,12 @@ #ifndef SPEEX_H #define SPEEX_H +/** @defgroup Codec Speex encoder and decoder + * This is the Speex codec itself. + * @{ + */ +#include "speex_types.h" #include "speex_bits.h" #ifdef __cplusplus @@ -56,7 +61,7 @@ extern "C" { /** Set quality value */ #define SPEEX_SET_QUALITY 4 /** Get current quality setting */ -#define SPEEX_GET_QUALITY 5 +/* #define SPEEX_GET_QUALITY 5 -- Doesn't make much sense, does it? */ /** Set sub-mode to use */ #define SPEEX_SET_MODE 6 @@ -93,10 +98,10 @@ extern "C" { /** Get current bit-rate used by the encoder or decoder */ #define SPEEX_GET_BITRATE 19 -/**Define a handler function for in-band Speex request*/ +/** Define a handler function for in-band Speex request*/ #define SPEEX_SET_HANDLER 20 -/**Define a handler function for in-band user-defined request*/ +/** Define a handler function for in-band user-defined request*/ #define SPEEX_SET_USER_HANDLER 22 /** Set sampling rate used in bit-rate computation */ @@ -126,16 +131,34 @@ extern "C" { /** Get DTX status (1 for on, 0 for off) */ #define SPEEX_GET_DTX 35 +/** Set submode encoding in each frame (1 for yes, 0 for no, setting to no breaks the standard) */ +#define SPEEX_SET_SUBMODE_ENCODING 36 +/** Get submode encoding in each frame */ +#define SPEEX_GET_SUBMODE_ENCODING 37 -/* Used internally, not to be used in applications */ -/** Used internally*/ -#define SPEEX_GET_PI_GAIN 100 -/** Used internally*/ -#define SPEEX_GET_EXC 101 -/** Used internally*/ -#define SPEEX_GET_INNOV 102 -/** Used internally*/ -#define SPEEX_GET_DTX_STATUS 103 +/*#define SPEEX_SET_LOOKAHEAD 38*/ +/** Returns the lookahead used by Speex separately for an encoder and a decoder. + * Sum encoder and decoder lookahead values to get the total codec lookahead. */ +#define SPEEX_GET_LOOKAHEAD 39 + +/** Sets tuning for packet-loss concealment (expected loss rate) */ +#define SPEEX_SET_PLC_TUNING 40 +/** Gets tuning for PLC */ +#define SPEEX_GET_PLC_TUNING 41 + +/** Sets the max bit-rate allowed in VBR mode */ +#define SPEEX_SET_VBR_MAX_BITRATE 42 +/** Gets the max bit-rate allowed in VBR mode */ +#define SPEEX_GET_VBR_MAX_BITRATE 43 + +/** Turn on/off input/output high-pass filtering */ +#define SPEEX_SET_HIGHPASS 44 +/** Get status of input/output high-pass filtering */ +#define SPEEX_GET_HIGHPASS 45 + +/** Get "activity level" of the last decoded frame, i.e. + how much damage we cause if we remove the frame */ +#define SPEEX_GET_ACTIVITY 47 /* Preserving compatibility:*/ @@ -145,6 +168,8 @@ extern "C" { #define SPEEX_GET_PF 1 + + /* Values allowed for mode queries */ /** Query the frame size of a mode */ #define SPEEX_MODE_FRAME_SIZE 0 @@ -153,52 +178,84 @@ extern "C" { #define SPEEX_SUBMODE_BITS_PER_FRAME 1 + +/** Get major Speex version */ +#define SPEEX_LIB_GET_MAJOR_VERSION 1 +/** Get minor Speex version */ +#define SPEEX_LIB_GET_MINOR_VERSION 3 +/** Get micro Speex version */ +#define SPEEX_LIB_GET_MICRO_VERSION 5 +/** Get extra Speex version */ +#define SPEEX_LIB_GET_EXTRA_VERSION 7 +/** Get Speex version string */ +#define SPEEX_LIB_GET_VERSION_STRING 9 + +/*#define SPEEX_LIB_SET_ALLOC_FUNC 10 +#define SPEEX_LIB_GET_ALLOC_FUNC 11 +#define SPEEX_LIB_SET_FREE_FUNC 12 +#define SPEEX_LIB_GET_FREE_FUNC 13 + +#define SPEEX_LIB_SET_WARNING_FUNC 14 +#define SPEEX_LIB_GET_WARNING_FUNC 15 +#define SPEEX_LIB_SET_ERROR_FUNC 16 +#define SPEEX_LIB_GET_ERROR_FUNC 17 +*/ + /** Number of defined modes in Speex */ #define SPEEX_NB_MODES 3 +/** modeID for the defined narrowband mode */ +#define SPEEX_MODEID_NB 0 + +/** modeID for the defined wideband mode */ +#define SPEEX_MODEID_WB 1 + +/** modeID for the defined ultra-wideband mode */ +#define SPEEX_MODEID_UWB 2 + struct SpeexMode; /* Prototypes for mode function pointers */ /** Encoder state initialization function */ -typedef void *(*encoder_init_func)(struct SpeexMode *mode); +typedef void *(*encoder_init_func)(const struct SpeexMode *mode); /** Encoder state destruction function */ typedef void (*encoder_destroy_func)(void *st); /** Main encoding function */ -typedef int (*encode_func)(void *state, float *in, SpeexBits *bits); +typedef int (*encode_func)(void *state, void *in, SpeexBits *bits); /** Function for controlling the encoder options */ typedef int (*encoder_ctl_func)(void *state, int request, void *ptr); /** Decoder state initialization function */ -typedef void *(*decoder_init_func)(struct SpeexMode *mode); +typedef void *(*decoder_init_func)(const struct SpeexMode *mode); /** Decoder state destruction function */ typedef void (*decoder_destroy_func)(void *st); /** Main decoding function */ -typedef int (*decode_func)(void *state, SpeexBits *bits, float *out); +typedef int (*decode_func)(void *state, SpeexBits *bits, void *out); /** Function for controlling the decoder options */ typedef int (*decoder_ctl_func)(void *state, int request, void *ptr); /** Query function for a mode */ -typedef int (*mode_query_func)(void *mode, int request, void *ptr); +typedef int (*mode_query_func)(const void *mode, int request, void *ptr); -/** Struct defining a Speex mode */ +/** Struct defining a Speex mode */ typedef struct SpeexMode { /** Pointer to the low-level mode data */ - void *mode; + const void *mode; /** Pointer to the mode query function */ mode_query_func query; - + /** The name of the mode (you should not rely on this to identify the mode)*/ - char *modeName; + const char *modeName; /**ID of the mode*/ int modeID; @@ -234,47 +291,59 @@ typedef struct SpeexMode { } SpeexMode; /** - * Returns a handle to a newly created Speex encoder state structure. For now, - * the "mode" argument can be &nb_mode or &wb_mode . In the future, more modes - * may be added. Note that for now if you have more than one channels to + * Returns a handle to a newly created Speex encoder state structure. For now, + * the "mode" argument can be &nb_mode or &wb_mode . In the future, more modes + * may be added. Note that for now if you have more than one channels to * encode, you need one state per channel. * - * @param mode The mode to use (either speex_nb_mode or speex_wb.mode) - * @return A newly created encoder + * @param mode The mode to use (either speex_nb_mode or speex_wb.mode) + * @return A newly created encoder state or NULL if state allocation fails */ -void *speex_encoder_init(SpeexMode *mode); +void *speex_encoder_init(const SpeexMode *mode); -/** Frees all resources associated to an existing Speex encoder state. +/** Frees all resources associated to an existing Speex encoder state. * @param state Encoder state to be destroyed */ void speex_encoder_destroy(void *state); /** Uses an existing encoder state to encode one frame of speech pointed to by "in". The encoded bit-stream is saved in "bits". @param state Encoder state - @param in Frame that will be encoded with a +-2^16 range + @param in Frame that will be encoded with a +-2^15 range. This data MAY be + overwritten by the encoder and should be considered uninitialised + after the call. @param bits Bit-stream where the data will be written + @return 0 if frame needs not be transmitted (DTX only), 1 otherwise */ int speex_encode(void *state, float *in, SpeexBits *bits); +/** Uses an existing encoder state to encode one frame of speech pointed to by + "in". The encoded bit-stream is saved in "bits". + @param state Encoder state + @param in Frame that will be encoded with a +-2^15 range + @param bits Bit-stream where the data will be written + @return 0 if frame needs not be transmitted (DTX only), 1 otherwise + */ +int speex_encode_int(void *state, spx_int16_t *in, SpeexBits *bits); + /** Used like the ioctl function to control the encoder parameters * * @param state Encoder state * @param request ioctl-type request (one of the SPEEX_* macros) * @param ptr Data exchanged to-from function - * @return 0 if frame needs not be transmitted (DTX only), 1 otherwise + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter */ int speex_encoder_ctl(void *state, int request, void *ptr); -/** Returns a handle to a newly created decoder state structure. For now, +/** Returns a handle to a newly created decoder state structure. For now, * the mode argument can be &nb_mode or &wb_mode . In the future, more modes * may be added. Note that for now if you have more than one channels to * decode, you need one state per channel. * * @param mode Speex mode (one of speex_nb_mode or speex_wb_mode) - * @return A newly created decoder state - */ -void *speex_decoder_init(SpeexMode *mode); + * @return A newly created decoder state or NULL if state allocation fails + */ +void *speex_decoder_init(const SpeexMode *mode); /** Frees all resources associated to an existing decoder state. * @@ -288,16 +357,26 @@ void speex_decoder_destroy(void *state); * @param state Decoder state * @param bits Bit-stream from which to decode the frame (NULL if the packet was lost) * @param out Where to write the decoded frame - * @return return status (0 for no error, -1 for end of stream, -2 other) + * @return return status (0 for no error, -1 for end of stream, -2 corrupt stream) */ int speex_decode(void *state, SpeexBits *bits, float *out); +/** Uses an existing decoder state to decode one frame of speech from + * bit-stream bits. The output speech is saved written to out. + * + * @param state Decoder state + * @param bits Bit-stream from which to decode the frame (NULL if the packet was lost) + * @param out Where to write the decoded frame + * @return return status (0 for no error, -1 for end of stream, -2 corrupt stream) + */ +int speex_decode_int(void *state, SpeexBits *bits, spx_int16_t *out); + /** Used like the ioctl function to control the encoder parameters * * @param state Decoder state * @param request ioctl-type request (one of the SPEEX_* macros) * @param ptr Data exchanged to-from function - * @return 0 for no error, 1 if a terminator is reached, 2 for another error + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter */ int speex_decoder_ctl(void *state, int request, void *ptr); @@ -307,25 +386,40 @@ int speex_decoder_ctl(void *state, int request, void *ptr); * @param mode Speex mode * @param request ioctl-type request (one of the SPEEX_* macros) * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter */ -int speex_mode_query(SpeexMode *mode, int request, void *ptr); +int speex_mode_query(const SpeexMode *mode, int request, void *ptr); +/** Functions for controlling the behavior of libspeex + * @param request ioctl-type request (one of the SPEEX_LIB_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter + */ +int speex_lib_ctl(int request, void *ptr); /** Default narrowband mode */ -extern SpeexMode speex_nb_mode; +extern const SpeexMode speex_nb_mode; /** Default wideband mode */ -extern SpeexMode speex_wb_mode; +extern const SpeexMode speex_wb_mode; /** Default "ultra-wideband" mode */ -extern SpeexMode speex_uwb_mode; +extern const SpeexMode speex_uwb_mode; /** List of all modes available */ -extern SpeexMode *speex_mode_list[SPEEX_NB_MODES]; +extern const SpeexMode * const speex_mode_list[SPEEX_NB_MODES]; + +/** Obtain one of the modes available */ +const SpeexMode * speex_lib_get_mode (int mode); + +#ifndef WIN32 +/* We actually override the function in the narrowband case so that we can avoid linking in the wideband stuff */ +#define speex_lib_get_mode(mode) ((mode)==SPEEX_MODEID_NB ? &speex_nb_mode : speex_lib_get_mode (mode)) +#endif #ifdef __cplusplus } #endif - +/** @}*/ #endif diff --git a/engine/voice_codecs/speex/source/libspeex/speex_bits.h b/engine/voice_codecs/speex/speex/speex_bits.h similarity index 78% rename from engine/voice_codecs/speex/source/libspeex/speex_bits.h rename to engine/voice_codecs/speex/speex/speex_bits.h index b41fdad7..f98b8268 100644 --- a/engine/voice_codecs/speex/source/libspeex/speex_bits.h +++ b/engine/voice_codecs/speex/speex/speex_bits.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -35,21 +35,23 @@ #ifndef BITS_H #define BITS_H +/** @defgroup SpeexBits SpeexBits: Bit-stream manipulations + * This is the structure that holds the bit-stream when encoding or decoding + * with Speex. It allows some manipulations as well. + * @{ + */ #ifdef __cplusplus extern "C" { #endif -/** Maximum size of the bit-stream (for fixed-size allocation) */ -#define MAX_BYTES_PER_FRAME 2000 - /** Bit-packing data structure representing (part of) a bit-stream. */ typedef struct SpeexBits { - char *bytes; /**< "raw" data */ + char *chars; /**< "raw" data */ int nbBits; /**< Total number of bits stored in the stream*/ - int bytePtr; /**< Position of the byte "cursor" */ - int bitPtr; /**< Position of the bit "cursor" within the current byte */ - int owner; /**< Does the struct "own" the "raw" buffer (member "bytes") */ + int charPtr; /**< Position of the byte "cursor" */ + int bitPtr; /**< Position of the bit "cursor" within the current char */ + int owner; /**< Does the struct "own" the "raw" buffer (member "chars") */ int overflow;/**< Set to one if we try to read past the valid data */ int buf_size;/**< Allocated size for buffer */ int reserved1; /**< Reserved for future use */ @@ -62,6 +64,9 @@ void speex_bits_init(SpeexBits *bits); /** Initializes SpeexBits struct using a pre-allocated buffer*/ void speex_bits_init_buffer(SpeexBits *bits, void *buff, int buf_size); +/** Sets the bits in a SpeexBits struct to use data from an existing buffer (for decoding without copying data) */ +void speex_bits_set_bit_buffer(SpeexBits *bits, void *buff, int buf_size); + /** Frees all resources associated to a SpeexBits struct. Right now this does nothing since no resources are allocated, but this could change in the future.*/ void speex_bits_destroy(SpeexBits *bits); @@ -72,16 +77,23 @@ void speex_bits_reset(SpeexBits *bits); void speex_bits_rewind(SpeexBits *bits); /** Initializes the bit-stream from the data in an area of memory */ -void speex_bits_read_from(SpeexBits *bits, char *bytes, int len); +void speex_bits_read_from(SpeexBits *bits, const char *bytes, int len); /** Append bytes to the bit-stream + * * @param bits Bit-stream to operate on * @param bytes pointer to the bytes what will be appended * @param len Number of bytes of append */ -void speex_bits_read_whole_bytes(SpeexBits *bits, char *bytes, int len); +void speex_bits_read_whole_bytes(SpeexBits *bits, const char *bytes, int len); -/** Write the content of a bit-stream to an area of memory */ +/** Write the content of a bit-stream to an area of memory + * + * @param bits Bit-stream to operate on + * @param bytes Memory location where to write the bits + * @param max_len Maximum number of bytes to write (i.e. size of the "bytes" buffer) + * @return Number of bytes written to the "bytes" buffer +*/ int speex_bits_write(SpeexBits *bits, char *bytes, int max_len); /** Like speex_bits_write, but writes only the complete bytes in the stream. Also removes the written bytes from the stream */ @@ -117,17 +129,23 @@ unsigned int speex_bits_unpack_unsigned(SpeexBits *bits, int nbBits); */ int speex_bits_nbytes(SpeexBits *bits); -/** Same as speex_bits_unpack_unsigned, but without modifying the cursor position */ +/** Same as speex_bits_unpack_unsigned, but without modifying the cursor position + * + * @param bits Bit-stream to operate on + * @param nbBits Number of bits to look for + * @return Value of the bits peeked, interpreted as unsigned + */ unsigned int speex_bits_peek_unsigned(SpeexBits *bits, int nbBits); /** Get the value of the next bit in the stream, without modifying the - * "cursor" position - * + * "cursor" position + * * @param bits Bit-stream to operate on + * @return Value of the bit peeked (one bit only) */ int speex_bits_peek(SpeexBits *bits); -/** Advances the position of the "bit cursor" in the stream +/** Advances the position of the "bit cursor" in the stream * * @param bits Bit-stream to operate on * @param n Number of bits to advance @@ -137,11 +155,12 @@ void speex_bits_advance(SpeexBits *bits, int n); /** Returns the number of bits remaining to be read in a stream * * @param bits Bit-stream to operate on + * @return Number of bits that can still be read from the stream */ int speex_bits_remaining(SpeexBits *bits); -/** Insert a terminator so that the data can be sent as a packet while auto-detecting - * the number of frames in each packet +/** Insert a terminator so that the data can be sent as a packet while auto-detecting + * the number of frames in each packet * * @param bits Bit-stream to operate on */ @@ -151,4 +170,5 @@ void speex_bits_insert_terminator(SpeexBits *bits); } #endif +/* @} */ #endif diff --git a/engine/voice_codecs/speex/source/libspeex/speex_callbacks.h b/engine/voice_codecs/speex/speex/speex_callbacks.h similarity index 91% rename from engine/voice_codecs/speex/source/libspeex/speex_callbacks.h rename to engine/voice_codecs/speex/speex/speex_callbacks.h index f6334f22..3928c34f 100644 --- a/engine/voice_codecs/speex/source/libspeex/speex_callbacks.h +++ b/engine/voice_codecs/speex/speex/speex_callbacks.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -35,6 +35,9 @@ #ifndef SPEEX_CALLBACKS_H #define SPEEX_CALLBACKS_H +/** @defgroup SpeexCallbacks Various definitions for Speex callbacks supported by the decoder. + * @{ + */ #include "speex.h" @@ -110,13 +113,16 @@ int speex_default_user_handler(SpeexBits *bits, void *state, void *data); - +/** Standard handler for low mode request (change low mode, no questions asked) */ int speex_std_low_mode_request_handler(SpeexBits *bits, void *state, void *data); +/** Standard handler for VBR request (Set VBR, no questions asked) */ int speex_std_vbr_request_handler(SpeexBits *bits, void *state, void *data); +/** Standard handler for enhancer request (Turn enhancer on/off, no questions asked) */ int speex_std_enh_request_handler(SpeexBits *bits, void *state, void *data); +/** Standard handler for VBR quality request (Set VBR quality, no questions asked) */ int speex_std_vbr_quality_request_handler(SpeexBits *bits, void *state, void *data); @@ -124,5 +130,5 @@ int speex_std_vbr_quality_request_handler(SpeexBits *bits, void *state, void *da } #endif - +/** @} */ #endif diff --git a/engine/voice_codecs/speex/speex/speex_config_types.h b/engine/voice_codecs/speex/speex/speex_config_types.h new file mode 100644 index 00000000..ca1f5a3c --- /dev/null +++ b/engine/voice_codecs/speex/speex/speex_config_types.h @@ -0,0 +1,12 @@ +#ifndef __SPEEX_TYPES_H__ +#define __SPEEX_TYPES_H__ + +#include + +typedef int16_t spx_int16_t; +typedef uint16_t spx_uint16_t; +typedef int32_t spx_int32_t; +typedef uint32_t spx_uint32_t; + +#endif + diff --git a/engine/voice_codecs/speex/speex/speex_echo.h b/engine/voice_codecs/speex/speex/speex_echo.h new file mode 100644 index 00000000..4c1aa5a5 --- /dev/null +++ b/engine/voice_codecs/speex/speex/speex_echo.h @@ -0,0 +1,170 @@ +/* Copyright (C) Jean-Marc Valin */ +/** + @file speex_echo.h + @brief Echo cancellation +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SPEEX_ECHO_H +#define SPEEX_ECHO_H +/** @defgroup SpeexEchoState SpeexEchoState: Acoustic echo canceller + * This is the acoustic echo canceller module. + * @{ + */ +#include "speexdsp_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** Obtain frame size used by the AEC */ +#define SPEEX_ECHO_GET_FRAME_SIZE 3 + +/** Set sampling rate */ +#define SPEEX_ECHO_SET_SAMPLING_RATE 24 +/** Get sampling rate */ +#define SPEEX_ECHO_GET_SAMPLING_RATE 25 + +/* Can't set window sizes */ +/** Get size of impulse response (int32) */ +#define SPEEX_ECHO_GET_IMPULSE_RESPONSE_SIZE 27 + +/* Can't set window content */ +/** Get impulse response (int32[]) */ +#define SPEEX_ECHO_GET_IMPULSE_RESPONSE 29 + +/** Internal echo canceller state. Should never be accessed directly. */ +struct SpeexEchoState_; + +/** @class SpeexEchoState + * This holds the state of the echo canceller. You need one per channel. +*/ + +/** Internal echo canceller state. Should never be accessed directly. */ +typedef struct SpeexEchoState_ SpeexEchoState; + +/** Creates a new echo canceller state + * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms) + * @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms) + * @return Newly-created echo canceller state + */ +SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); + +/** Creates a new multi-channel echo canceller state + * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms) + * @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms) + * @param nb_mic Number of microphone channels + * @param nb_speakers Number of speaker channels + * @return Newly-created echo canceller state + */ +SpeexEchoState *speex_echo_state_init_mc(int frame_size, int filter_length, int nb_mic, int nb_speakers); + +/** Destroys an echo canceller state + * @param st Echo canceller state +*/ +void speex_echo_state_destroy(SpeexEchoState *st); + +/** Performs echo cancellation a frame, based on the audio sent to the speaker (no delay is added + * to playback in this form) + * + * @param st Echo canceller state + * @param rec Signal from the microphone (near end + far end echo) + * @param play Signal played to the speaker (received from far end) + * @param out Returns near-end signal with echo removed + */ +void speex_echo_cancellation(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out); + +/** Performs echo cancellation a frame (deprecated) */ +void speex_echo_cancel(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out, spx_int32_t *Yout); + +/** Perform echo cancellation using internal playback buffer, which is delayed by two frames + * to account for the delay introduced by most soundcards (but it could be off!) + * @param st Echo canceller state + * @param rec Signal from the microphone (near end + far end echo) + * @param out Returns near-end signal with echo removed +*/ +void speex_echo_capture(SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out); + +/** Let the echo canceller know that a frame was just queued to the soundcard + * @param st Echo canceller state + * @param play Signal played to the speaker (received from far end) +*/ +void speex_echo_playback(SpeexEchoState *st, const spx_int16_t *play); + +/** Reset the echo canceller to its original state + * @param st Echo canceller state + */ +void speex_echo_state_reset(SpeexEchoState *st); + +/** Used like the ioctl function to control the echo canceller parameters + * + * @param st Echo canceller state + * @param request ioctl-type request (one of the SPEEX_ECHO_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown + */ +int speex_echo_ctl(SpeexEchoState *st, int request, void *ptr); + + + +struct SpeexDecorrState_; + +typedef struct SpeexDecorrState_ SpeexDecorrState; + + +/** Create a state for the channel decorrelation algorithm + This is useful for multi-channel echo cancellation only + * @param rate Sampling rate + * @param channels Number of channels (it's a bit pointless if you don't have at least 2) + * @param frame_size Size of the frame to process at ones (counting samples *per* channel) +*/ +SpeexDecorrState *speex_decorrelate_new(int rate, int channels, int frame_size); + +/** Remove correlation between the channels by modifying the phase and possibly + adding noise in a way that is not (or little) perceptible. + * @param st Decorrelator state + * @param in Input audio in interleaved format + * @param out Result of the decorrelation (out *may* alias in) + * @param strength How much alteration of the audio to apply from 0 to 100. +*/ +void speex_decorrelate(SpeexDecorrState *st, const spx_int16_t *in, spx_int16_t *out, int strength); + +/** Destroy a Decorrelation state + * @param st State to destroy +*/ +void speex_decorrelate_destroy(SpeexDecorrState *st); + + +#ifdef __cplusplus +} +#endif + + +/** @}*/ +#endif diff --git a/engine/voice_codecs/speex/source/libspeex/speex_header.h b/engine/voice_codecs/speex/speex/speex_header.h similarity index 59% rename from engine/voice_codecs/speex/source/libspeex/speex_header.h rename to engine/voice_codecs/speex/speex/speex_header.h index 8ded9114..ffe8c471 100644 --- a/engine/voice_codecs/speex/source/libspeex/speex_header.h +++ b/engine/voice_codecs/speex/speex/speex_header.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -36,6 +36,12 @@ #ifndef SPEEX_HEADER_H #define SPEEX_HEADER_H +/** @defgroup SpeexHeader SpeexHeader: Makes it easy to write/parse an Ogg/Speex header + * This is the Speex header for the Ogg encapsulation. You don't need that if you just use RTP. + * @{ + */ + +#include "speex_types.h" #ifdef __cplusplus extern "C" { @@ -43,30 +49,33 @@ extern "C" { struct SpeexMode; +/** Length of the Speex header identifier */ +#define SPEEX_HEADER_STRING_LENGTH 8 + /** Maximum number of characters for encoding the Speex version number in the header */ #define SPEEX_HEADER_VERSION_LENGTH 20 /** Speex header info for file-based formats */ typedef struct SpeexHeader { - char speex_string[8]; /**< Identifies a Speex bit-stream, always set to "Speex " */ + char speex_string[SPEEX_HEADER_STRING_LENGTH]; /**< Identifies a Speex bit-stream, always set to "Speex " */ char speex_version[SPEEX_HEADER_VERSION_LENGTH]; /**< Speex version */ - int speex_version_id; /**< Version for Speex (for checking compatibility) */ - int header_size; /**< Total size of the header ( sizeof(SpeexHeader) ) */ - int rate; /**< Sampling rate used */ - int mode; /**< Mode used (0 for narrowband, 1 for wideband) */ - int mode_bitstream_version; /**< Version ID of the bit-stream */ - int nb_channels; /**< Number of channels encoded */ - int bitrate; /**< Bit-rate used */ - int frame_size; /**< Size of frames */ - int vbr; /**< 1 for a VBR encoding, 0 otherwise */ - int frames_per_packet; /**< Number of frames stored per Ogg packet */ - int extra_headers; /**< Number of additional headers after the comments */ - int reserved1; /**< Reserved for future use, must be zero */ - int reserved2; /**< Reserved for future use, must be zero */ + spx_int32_t speex_version_id; /**< Version for Speex (for checking compatibility) */ + spx_int32_t header_size; /**< Total size of the header ( sizeof(SpeexHeader) ) */ + spx_int32_t rate; /**< Sampling rate used */ + spx_int32_t mode; /**< Mode used (0 for narrowband, 1 for wideband) */ + spx_int32_t mode_bitstream_version; /**< Version ID of the bit-stream */ + spx_int32_t nb_channels; /**< Number of channels encoded */ + spx_int32_t bitrate; /**< Bit-rate used */ + spx_int32_t frame_size; /**< Size of frames */ + spx_int32_t vbr; /**< 1 for a VBR encoding, 0 otherwise */ + spx_int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */ + spx_int32_t extra_headers; /**< Number of additional headers after the comments */ + spx_int32_t reserved1; /**< Reserved for future use, must be zero */ + spx_int32_t reserved2; /**< Reserved for future use, must be zero */ } SpeexHeader; /** Initializes a SpeexHeader using basic information */ -void speex_init_header(SpeexHeader *header, int rate, int nb_channels, struct SpeexMode *m); +void speex_init_header(SpeexHeader *header, int rate, int nb_channels, const struct SpeexMode *m); /** Creates the header packet from the header itself (mostly involves endianness conversion) */ char *speex_header_to_packet(SpeexHeader *header, int *size); @@ -74,9 +83,12 @@ char *speex_header_to_packet(SpeexHeader *header, int *size); /** Creates a SpeexHeader from a packet */ SpeexHeader *speex_packet_to_header(char *packet, int size); +/** Frees the memory allocated by either speex_header_to_packet() or speex_packet_to_header() */ +void speex_header_free(void *ptr); + #ifdef __cplusplus } #endif - +/** @} */ #endif diff --git a/engine/voice_codecs/speex/speex/speex_jitter.h b/engine/voice_codecs/speex/speex/speex_jitter.h new file mode 100644 index 00000000..8fc8d7ec --- /dev/null +++ b/engine/voice_codecs/speex/speex/speex_jitter.h @@ -0,0 +1,197 @@ +/* Copyright (C) 2002 Jean-Marc Valin */ +/** + @file speex_jitter.h + @brief Adaptive jitter buffer for Speex +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +*/ + +#ifndef SPEEX_JITTER_H +#define SPEEX_JITTER_H +/** @defgroup JitterBuffer JitterBuffer: Adaptive jitter buffer + * This is the jitter buffer that reorders UDP/RTP packets and adjusts the buffer size + * to maintain good quality and low latency. + * @{ + */ + +#include "speexdsp_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** Generic adaptive jitter buffer state */ +struct JitterBuffer_; + +/** Generic adaptive jitter buffer state */ +typedef struct JitterBuffer_ JitterBuffer; + +/** Definition of an incoming packet */ +typedef struct _JitterBufferPacket JitterBufferPacket; + +/** Definition of an incoming packet */ +struct _JitterBufferPacket { + char *data; /**< Data bytes contained in the packet */ + spx_uint32_t len; /**< Length of the packet in bytes */ + spx_uint32_t timestamp; /**< Timestamp for the packet */ + spx_uint32_t span; /**< Time covered by the packet (same units as timestamp) */ + spx_uint16_t sequence; /**< RTP Sequence number if available (0 otherwise) */ + spx_uint32_t user_data; /**< Put whatever data you like here (it's ignored by the jitter buffer) */ +}; + +/** Packet has been retrieved */ +#define JITTER_BUFFER_OK 0 +/** Packet is lost or is late */ +#define JITTER_BUFFER_MISSING 1 +/** A "fake" packet is meant to be inserted here to increase buffering */ +#define JITTER_BUFFER_INSERTION 2 +/** There was an error in the jitter buffer */ +#define JITTER_BUFFER_INTERNAL_ERROR -1 +/** Invalid argument */ +#define JITTER_BUFFER_BAD_ARGUMENT -2 + + +/** Set minimum amount of extra buffering required (margin) */ +#define JITTER_BUFFER_SET_MARGIN 0 +/** Get minimum amount of extra buffering required (margin) */ +#define JITTER_BUFFER_GET_MARGIN 1 +/* JITTER_BUFFER_SET_AVAILABLE_COUNT wouldn't make sense */ + +/** Get the amount of available packets currently buffered */ +#define JITTER_BUFFER_GET_AVAILABLE_COUNT 3 +/** Included because of an early misspelling (will remove in next release) */ +#define JITTER_BUFFER_GET_AVALIABLE_COUNT 3 + +/** Assign a function to destroy unused packet. When setting that, the jitter + buffer no longer copies packet data. */ +#define JITTER_BUFFER_SET_DESTROY_CALLBACK 4 +/** */ +#define JITTER_BUFFER_GET_DESTROY_CALLBACK 5 + +/** Tell the jitter buffer to only adjust the delay in multiples of the step parameter provided */ +#define JITTER_BUFFER_SET_DELAY_STEP 6 +/** */ +#define JITTER_BUFFER_GET_DELAY_STEP 7 + +/** Tell the jitter buffer to only do concealment in multiples of the size parameter provided */ +#define JITTER_BUFFER_SET_CONCEALMENT_SIZE 8 +#define JITTER_BUFFER_GET_CONCEALMENT_SIZE 9 + +/** Absolute max amount of loss that can be tolerated regardless of the delay. Typical loss + should be half of that or less. */ +#define JITTER_BUFFER_SET_MAX_LATE_RATE 10 +#define JITTER_BUFFER_GET_MAX_LATE_RATE 11 + +/** Equivalent cost of one percent late packet in timestamp units */ +#define JITTER_BUFFER_SET_LATE_COST 12 +#define JITTER_BUFFER_GET_LATE_COST 13 + + +/** Initialises jitter buffer + * + * @param step_size Starting value for the size of concleanment packets and delay + adjustment steps. Can be changed at any time using JITTER_BUFFER_SET_DELAY_STEP + and JITTER_BUFFER_GET_CONCEALMENT_SIZE. + * @return Newly created jitter buffer state + */ +JitterBuffer *jitter_buffer_init(int step_size); + +/** Restores jitter buffer to its original state + * + * @param jitter Jitter buffer state + */ +void jitter_buffer_reset(JitterBuffer *jitter); + +/** Destroys jitter buffer + * + * @param jitter Jitter buffer state + */ +void jitter_buffer_destroy(JitterBuffer *jitter); + +/** Put one packet into the jitter buffer + * + * @param jitter Jitter buffer state + * @param packet Incoming packet +*/ +void jitter_buffer_put(JitterBuffer *jitter, const JitterBufferPacket *packet); + +/** Get one packet from the jitter buffer + * + * @param jitter Jitter buffer state + * @param packet Returned packet + * @param desired_span Number of samples (or units) we wish to get from the buffer (no guarantee) + * @param current_timestamp Timestamp for the returned packet +*/ +int jitter_buffer_get(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t desired_span, spx_int32_t *start_offset); + +/** Used right after jitter_buffer_get() to obtain another packet that would have the same timestamp. + * This is mainly useful for media where a single "frame" can be split into several packets. + * + * @param jitter Jitter buffer state + * @param packet Returned packet + */ +int jitter_buffer_get_another(JitterBuffer *jitter, JitterBufferPacket *packet); + +/** Get pointer timestamp of jitter buffer + * + * @param jitter Jitter buffer state +*/ +int jitter_buffer_get_pointer_timestamp(JitterBuffer *jitter); + +/** Advance by one tick + * + * @param jitter Jitter buffer state +*/ +void jitter_buffer_tick(JitterBuffer *jitter); + +/** Telling the jitter buffer about the remaining data in the application buffer + * @param jitter Jitter buffer state + * @param rem Amount of data buffered by the application (timestamp units) + */ +void jitter_buffer_remaining_span(JitterBuffer *jitter, spx_uint32_t rem); + +/** Used like the ioctl function to control the jitter buffer parameters + * + * @param jitter Jitter buffer state + * @param request ioctl-type request (one of the JITTER_BUFFER_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown +*/ +int jitter_buffer_ctl(JitterBuffer *jitter, int request, void *ptr); + +int jitter_buffer_update_delay(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t *start_offset); + +/* @} */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/engine/voice_codecs/speex/speex/speex_preprocess.h b/engine/voice_codecs/speex/speex/speex_preprocess.h new file mode 100644 index 00000000..a2e1210d --- /dev/null +++ b/engine/voice_codecs/speex/speex/speex_preprocess.h @@ -0,0 +1,219 @@ +/* Copyright (C) 2003 Epic Games + Written by Jean-Marc Valin */ +/** + * @file speex_preprocess.h + * @brief Speex preprocessor. The preprocess can do noise suppression, + * residual echo suppression (after using the echo canceller), automatic + * gain control (AGC) and voice activity detection (VAD). +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SPEEX_PREPROCESS_H +#define SPEEX_PREPROCESS_H +/** @defgroup SpeexPreprocessState SpeexPreprocessState: The Speex preprocessor + * This is the Speex preprocessor. The preprocess can do noise suppression, + * residual echo suppression (after using the echo canceller), automatic + * gain control (AGC) and voice activity detection (VAD). + * @{ + */ + +#include "speexdsp_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** State of the preprocessor (one per channel). Should never be accessed directly. */ +struct SpeexPreprocessState_; + +/** State of the preprocessor (one per channel). Should never be accessed directly. */ +typedef struct SpeexPreprocessState_ SpeexPreprocessState; + + +/** Creates a new preprocessing state. You MUST create one state per channel processed. + * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms). Must be + * the same value as that used for the echo canceller for residual echo cancellation to work. + * @param sampling_rate Sampling rate used for the input. + * @return Newly created preprocessor state +*/ +SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate); + +/** Destroys a preprocessor state + * @param st Preprocessor state to destroy +*/ +void speex_preprocess_state_destroy(SpeexPreprocessState *st); + +/** Preprocess a frame + * @param st Preprocessor state + * @param x Audio sample vector (in and out). Must be same size as specified in speex_preprocess_state_init(). + * @return Bool value for voice activity (1 for speech, 0 for noise/silence), ONLY if VAD turned on. +*/ +int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x); + +/** Preprocess a frame (deprecated, use speex_preprocess_run() instead)*/ +int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo); + +/** Update preprocessor state, but do not compute the output + * @param st Preprocessor state + * @param x Audio sample vector (in only). Must be same size as specified in speex_preprocess_state_init(). +*/ +void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x); + +/** Used like the ioctl function to control the preprocessor parameters + * @param st Preprocessor state + * @param request ioctl-type request (one of the SPEEX_PREPROCESS_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown +*/ +int speex_preprocess_ctl(SpeexPreprocessState *st, int request, void *ptr); + + + +/** Set preprocessor denoiser state */ +#define SPEEX_PREPROCESS_SET_DENOISE 0 +/** Get preprocessor denoiser state */ +#define SPEEX_PREPROCESS_GET_DENOISE 1 + +/** Set preprocessor Automatic Gain Control state */ +#define SPEEX_PREPROCESS_SET_AGC 2 +/** Get preprocessor Automatic Gain Control state */ +#define SPEEX_PREPROCESS_GET_AGC 3 + +/** Set preprocessor Voice Activity Detection state */ +#define SPEEX_PREPROCESS_SET_VAD 4 +/** Get preprocessor Voice Activity Detection state */ +#define SPEEX_PREPROCESS_GET_VAD 5 + +/** Set preprocessor Automatic Gain Control level (float) */ +#define SPEEX_PREPROCESS_SET_AGC_LEVEL 6 +/** Get preprocessor Automatic Gain Control level (float) */ +#define SPEEX_PREPROCESS_GET_AGC_LEVEL 7 + +/** Set preprocessor dereverb state */ +#define SPEEX_PREPROCESS_SET_DEREVERB 8 +/** Get preprocessor dereverb state */ +#define SPEEX_PREPROCESS_GET_DEREVERB 9 + +/** Set preprocessor dereverb level */ +#define SPEEX_PREPROCESS_SET_DEREVERB_LEVEL 10 +/** Get preprocessor dereverb level */ +#define SPEEX_PREPROCESS_GET_DEREVERB_LEVEL 11 + +/** Set preprocessor dereverb decay */ +#define SPEEX_PREPROCESS_SET_DEREVERB_DECAY 12 +/** Get preprocessor dereverb decay */ +#define SPEEX_PREPROCESS_GET_DEREVERB_DECAY 13 + +/** Set probability required for the VAD to go from silence to voice */ +#define SPEEX_PREPROCESS_SET_PROB_START 14 +/** Get probability required for the VAD to go from silence to voice */ +#define SPEEX_PREPROCESS_GET_PROB_START 15 + +/** Set probability required for the VAD to stay in the voice state (integer percent) */ +#define SPEEX_PREPROCESS_SET_PROB_CONTINUE 16 +/** Get probability required for the VAD to stay in the voice state (integer percent) */ +#define SPEEX_PREPROCESS_GET_PROB_CONTINUE 17 + +/** Set maximum attenuation of the noise in dB (negative number) */ +#define SPEEX_PREPROCESS_SET_NOISE_SUPPRESS 18 +/** Get maximum attenuation of the noise in dB (negative number) */ +#define SPEEX_PREPROCESS_GET_NOISE_SUPPRESS 19 + +/** Set maximum attenuation of the residual echo in dB (negative number) */ +#define SPEEX_PREPROCESS_SET_ECHO_SUPPRESS 20 +/** Get maximum attenuation of the residual echo in dB (negative number) */ +#define SPEEX_PREPROCESS_GET_ECHO_SUPPRESS 21 + +/** Set maximum attenuation of the residual echo in dB when near end is active (negative number) */ +#define SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE 22 +/** Get maximum attenuation of the residual echo in dB when near end is active (negative number) */ +#define SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE 23 + +/** Set the corresponding echo canceller state so that residual echo suppression can be performed (NULL for no residual echo suppression) */ +#define SPEEX_PREPROCESS_SET_ECHO_STATE 24 +/** Get the corresponding echo canceller state */ +#define SPEEX_PREPROCESS_GET_ECHO_STATE 25 + +/** Set maximal gain increase in dB/second (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_INCREMENT 26 + +/** Get maximal gain increase in dB/second (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_INCREMENT 27 + +/** Set maximal gain decrease in dB/second (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_DECREMENT 28 + +/** Get maximal gain decrease in dB/second (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_DECREMENT 29 + +/** Set maximal gain in dB (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_MAX_GAIN 30 + +/** Get maximal gain in dB (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_MAX_GAIN 31 + +/* Can't set loudness */ +/** Get loudness */ +#define SPEEX_PREPROCESS_GET_AGC_LOUDNESS 33 + +/* Can't set gain */ +/** Get current gain (int32 percent) */ +#define SPEEX_PREPROCESS_GET_AGC_GAIN 35 + +/* Can't set spectrum size */ +/** Get spectrum size for power spectrum (int32) */ +#define SPEEX_PREPROCESS_GET_PSD_SIZE 37 + +/* Can't set power spectrum */ +/** Get power spectrum (int32[] of squared values) */ +#define SPEEX_PREPROCESS_GET_PSD 39 + +/* Can't set noise size */ +/** Get spectrum size for noise estimate (int32) */ +#define SPEEX_PREPROCESS_GET_NOISE_PSD_SIZE 41 + +/* Can't set noise estimate */ +/** Get noise estimate (int32[] of squared values) */ +#define SPEEX_PREPROCESS_GET_NOISE_PSD 43 + +/* Can't set speech probability */ +/** Get speech probability in last frame (int32). */ +#define SPEEX_PREPROCESS_GET_PROB 45 + +/** Set preprocessor Automatic Gain Control level (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_TARGET 46 +/** Get preprocessor Automatic Gain Control level (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_TARGET 47 + +#ifdef __cplusplus +} +#endif + +/** @}*/ +#endif diff --git a/engine/voice_codecs/speex/speex/speex_resampler.h b/engine/voice_codecs/speex/speex/speex_resampler.h new file mode 100644 index 00000000..901de37b --- /dev/null +++ b/engine/voice_codecs/speex/speex/speex_resampler.h @@ -0,0 +1,343 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: speex_resampler.h + Resampling code + + The design goals of this code are: + - Very fast algorithm + - Low memory requirement + - Good *perceptual* quality (and not best SNR) + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + + +#ifndef SPEEX_RESAMPLER_H +#define SPEEX_RESAMPLER_H + +#ifdef OUTSIDE_SPEEX + +/********* WARNING: MENTAL SANITY ENDS HERE *************/ + +/* If the resampler is defined outside of Speex, we change the symbol names so that + there won't be any clash if linking with Speex later on. */ + +/* #define RANDOM_PREFIX your software name here */ +#ifndef RANDOM_PREFIX +#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes" +#endif + +#define CAT_PREFIX2(a,b) a ## b +#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b) + +#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init) +#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac) +#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy) +#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float) +#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int) +#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float) +#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int) +#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate) +#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate) +#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac) +#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio) +#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality) +#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality) +#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride) +#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride) +#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride) +#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride) +#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency) +#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency) +#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros) +#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem) +#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror) + +#define spx_int16_t short +#define spx_int32_t int +#define spx_uint16_t unsigned short +#define spx_uint32_t unsigned int + +#define speex_assert(cond) + +#else /* OUTSIDE_SPEEX */ + +#include "speexdsp_types.h" + +#endif /* OUTSIDE_SPEEX */ + +#ifdef __cplusplus +extern "C" { +#endif + +#define SPEEX_RESAMPLER_QUALITY_MAX 10 +#define SPEEX_RESAMPLER_QUALITY_MIN 0 +#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4 +#define SPEEX_RESAMPLER_QUALITY_VOIP 3 +#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5 + +enum { + RESAMPLER_ERR_SUCCESS = 0, + RESAMPLER_ERR_ALLOC_FAILED = 1, + RESAMPLER_ERR_BAD_STATE = 2, + RESAMPLER_ERR_INVALID_ARG = 3, + RESAMPLER_ERR_PTR_OVERLAP = 4, + RESAMPLER_ERR_OVERFLOW = 5, + + RESAMPLER_ERR_MAX_ERROR +}; + +struct SpeexResamplerState_; +typedef struct SpeexResamplerState_ SpeexResamplerState; + +/** Create a new resampler with integer input and output rates. + * @param nb_channels Number of channels to be processed + * @param in_rate Input sampling rate (integer number of Hz). + * @param out_rate Output sampling rate (integer number of Hz). + * @param quality Resampling quality between 0 and 10, where 0 has poor quality + * and 10 has very high quality. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, + spx_uint32_t in_rate, + spx_uint32_t out_rate, + int quality, + int *err); + +/** Create a new resampler with fractional input/output rates. The sampling + * rate ratio is an arbitrary rational number with both the numerator and + * denominator being 32-bit integers. + * @param nb_channels Number of channels to be processed + * @param ratio_num Numerator of the sampling rate ratio + * @param ratio_den Denominator of the sampling rate ratio + * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). + * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). + * @param quality Resampling quality between 0 and 10, where 0 has poor quality + * and 10 has very high quality. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate, + int quality, + int *err); + +/** Destroy a resampler state. + * @param st Resampler state + */ +void speex_resampler_destroy(SpeexResamplerState *st); + +/** Resample a float array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param channel_index Index of the channel to process for the multi-channel + * base (0 otherwise) + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the + * number of samples processed + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written + */ +int speex_resampler_process_float(SpeexResamplerState *st, + spx_uint32_t channel_index, + const float *in, + spx_uint32_t *in_len, + float *out, + spx_uint32_t *out_len); + +/** Resample an int array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param channel_index Index of the channel to process for the multi-channel + * base (0 otherwise) + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written + */ +int speex_resampler_process_int(SpeexResamplerState *st, + spx_uint32_t channel_index, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, + spx_uint32_t *out_len); + +/** Resample an interleaved float array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed. This is all per-channel. + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written. + * This is all per-channel. + */ +int speex_resampler_process_interleaved_float(SpeexResamplerState *st, + const float *in, + spx_uint32_t *in_len, + float *out, + spx_uint32_t *out_len); + +/** Resample an interleaved int array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed. This is all per-channel. + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written. + * This is all per-channel. + */ +int speex_resampler_process_interleaved_int(SpeexResamplerState *st, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, + spx_uint32_t *out_len); + +/** Set (change) the input/output sampling rates (integer value). + * @param st Resampler state + * @param in_rate Input sampling rate (integer number of Hz). + * @param out_rate Output sampling rate (integer number of Hz). + */ +int speex_resampler_set_rate(SpeexResamplerState *st, + spx_uint32_t in_rate, + spx_uint32_t out_rate); + +/** Get the current input/output sampling rates (integer value). + * @param st Resampler state + * @param in_rate Input sampling rate (integer number of Hz) copied. + * @param out_rate Output sampling rate (integer number of Hz) copied. + */ +void speex_resampler_get_rate(SpeexResamplerState *st, + spx_uint32_t *in_rate, + spx_uint32_t *out_rate); + +/** Set (change) the input/output sampling rates and resampling ratio + * (fractional values in Hz supported). + * @param st Resampler state + * @param ratio_num Numerator of the sampling rate ratio + * @param ratio_den Denominator of the sampling rate ratio + * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). + * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). + */ +int speex_resampler_set_rate_frac(SpeexResamplerState *st, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate); + +/** Get the current resampling ratio. This will be reduced to the least + * common denominator. + * @param st Resampler state + * @param ratio_num Numerator of the sampling rate ratio copied + * @param ratio_den Denominator of the sampling rate ratio copied + */ +void speex_resampler_get_ratio(SpeexResamplerState *st, + spx_uint32_t *ratio_num, + spx_uint32_t *ratio_den); + +/** Set (change) the conversion quality. + * @param st Resampler state + * @param quality Resampling quality between 0 and 10, where 0 has poor + * quality and 10 has very high quality. + */ +int speex_resampler_set_quality(SpeexResamplerState *st, + int quality); + +/** Get the conversion quality. + * @param st Resampler state + * @param quality Resampling quality between 0 and 10, where 0 has poor + * quality and 10 has very high quality. + */ +void speex_resampler_get_quality(SpeexResamplerState *st, + int *quality); + +/** Set (change) the input stride. + * @param st Resampler state + * @param stride Input stride + */ +void speex_resampler_set_input_stride(SpeexResamplerState *st, + spx_uint32_t stride); + +/** Get the input stride. + * @param st Resampler state + * @param stride Input stride copied + */ +void speex_resampler_get_input_stride(SpeexResamplerState *st, + spx_uint32_t *stride); + +/** Set (change) the output stride. + * @param st Resampler state + * @param stride Output stride + */ +void speex_resampler_set_output_stride(SpeexResamplerState *st, + spx_uint32_t stride); + +/** Get the output stride. + * @param st Resampler state copied + * @param stride Output stride + */ +void speex_resampler_get_output_stride(SpeexResamplerState *st, + spx_uint32_t *stride); + +/** Get the latency introduced by the resampler measured in input samples. + * @param st Resampler state + */ +int speex_resampler_get_input_latency(SpeexResamplerState *st); + +/** Get the latency introduced by the resampler measured in output samples. + * @param st Resampler state + */ +int speex_resampler_get_output_latency(SpeexResamplerState *st); + +/** Make sure that the first samples to go out of the resamplers don't have + * leading zeros. This is only useful before starting to use a newly created + * resampler. It is recommended to use that when resampling an audio file, as + * it will generate a file with the same length. For real-time processing, + * it is probably easier not to use this call (so that the output duration + * is the same for the first frame). + * @param st Resampler state + */ +int speex_resampler_skip_zeros(SpeexResamplerState *st); + +/** Reset a resampler so a new (unrelated) stream can be processed. + * @param st Resampler state + */ +int speex_resampler_reset_mem(SpeexResamplerState *st); + +/** Returns the English meaning for an error code + * @param err Error code + * @return English string + */ +const char *speex_resampler_strerror(int err); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/engine/voice_codecs/speex/source/libspeex/speex_stereo.h b/engine/voice_codecs/speex/speex/speex_stereo.h similarity index 69% rename from engine/voice_codecs/speex/source/libspeex/speex_stereo.h rename to engine/voice_codecs/speex/speex/speex_stereo.h index 3fbeb91b..5844f5a1 100644 --- a/engine/voice_codecs/speex/source/libspeex/speex_stereo.h +++ b/engine/voice_codecs/speex/speex/speex_stereo.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -34,10 +34,20 @@ #ifndef STEREO_H #define STEREO_H +/** @defgroup SpeexStereoState SpeexStereoState: Handling Speex stereo files + * This describes the Speex intensity stereo encoding/decoding + * @{ + */ +#include "speex_types.h" #include "speex_bits.h" -/** State used for decoding (intensity) stereo information */ + +#ifdef __cplusplus +extern "C" { +#endif + +/** If you access any of these fields directly, I'll personally come and bite you */ typedef struct SpeexStereoState { float balance; /**< Left/right balance info */ float e_ratio; /**< Ratio of energies: E(left+right)/[E(left)+E(right)] */ @@ -47,16 +57,36 @@ typedef struct SpeexStereoState { float reserved2; /**< Reserved for future use */ } SpeexStereoState; -/** Initialization value for a stereo state */ -#define SPEEX_STEREO_STATE_INIT {1,.5,1,1} +/** Deprecated. Use speex_stereo_state_init() instead. */ +#define SPEEX_STEREO_STATE_INIT {1,.5,1,1,0,0} + +/** Initialise/create a stereo stereo state */ +SpeexStereoState *speex_stereo_state_init(); + +/** Reset/re-initialise an already allocated stereo state */ +void speex_stereo_state_reset(SpeexStereoState *stereo); + +/** Destroy a stereo stereo state */ +void speex_stereo_state_destroy(SpeexStereoState *stereo); /** Transforms a stereo frame into a mono frame and stores intensity stereo info in 'bits' */ void speex_encode_stereo(float *data, int frame_size, SpeexBits *bits); +/** Transforms a stereo frame into a mono frame and stores intensity stereo info in 'bits' */ +void speex_encode_stereo_int(spx_int16_t *data, int frame_size, SpeexBits *bits); + /** Transforms a mono frame into a stereo frame using intensity stereo info */ void speex_decode_stereo(float *data, int frame_size, SpeexStereoState *stereo); +/** Transforms a mono frame into a stereo frame using intensity stereo info */ +void speex_decode_stereo_int(spx_int16_t *data, int frame_size, SpeexStereoState *stereo); + /** Callback handler for intensity stereo info */ int speex_std_stereo_request_handler(SpeexBits *bits, void *state, void *data); +#ifdef __cplusplus +} +#endif + +/** @} */ #endif diff --git a/engine/voice_codecs/speex/speex/speex_types.h b/engine/voice_codecs/speex/speex/speex_types.h new file mode 100644 index 00000000..bdc63295 --- /dev/null +++ b/engine/voice_codecs/speex/speex/speex_types.h @@ -0,0 +1,126 @@ +/* speex_types.h taken from libogg */ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: #ifdef jail to whip a few platforms into the UNIX ideal. + last mod: $Id: os_types.h 7524 2004-08-11 04:20:36Z conrad $ + + ********************************************************************/ +/** + @file speex_types.h + @brief Speex types +*/ +#ifndef _SPEEX_TYPES_H +#define _SPEEX_TYPES_H + +#if defined(_WIN32) + +# if defined(__CYGWIN__) +# include <_G_config.h> + typedef _G_int32_t spx_int32_t; + typedef _G_uint32_t spx_uint32_t; + typedef _G_int16_t spx_int16_t; + typedef _G_uint16_t spx_uint16_t; +# elif defined(__MINGW32__) + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; +# elif defined(__MWERKS__) + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; +# else + /* MSVC/Borland */ + typedef __int32 spx_int32_t; + typedef unsigned __int32 spx_uint32_t; + typedef __int16 spx_int16_t; + typedef unsigned __int16 spx_uint16_t; +# endif + +#elif defined(__MACOS__) + +# include + typedef SInt16 spx_int16_t; + typedef UInt16 spx_uint16_t; + typedef SInt32 spx_int32_t; + typedef UInt32 spx_uint32_t; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined(__BEOS__) + + /* Be */ +# include + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short spx_int16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(R5900) + + /* PS2 EE */ + typedef int spx_int32_t; + typedef unsigned spx_uint32_t; + typedef short spx_int16_t; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef signed int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef long spx_int32_t; + typedef unsigned long spx_uint32_t; + +#elif defined(CONFIG_TI_C6X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#else + +#include "speex_config_types.h" + +#endif + +#endif /* _SPEEX_TYPES_H */ diff --git a/engine/voice_codecs/speex/speex/speexdsp_config_types.h b/engine/voice_codecs/speex/speex/speexdsp_config_types.h new file mode 100644 index 00000000..ca1f5a3c --- /dev/null +++ b/engine/voice_codecs/speex/speex/speexdsp_config_types.h @@ -0,0 +1,12 @@ +#ifndef __SPEEX_TYPES_H__ +#define __SPEEX_TYPES_H__ + +#include + +typedef int16_t spx_int16_t; +typedef uint16_t spx_uint16_t; +typedef int32_t spx_int32_t; +typedef uint32_t spx_uint32_t; + +#endif + diff --git a/engine/voice_codecs/speex/speex/speexdsp_types.h b/engine/voice_codecs/speex/speex/speexdsp_types.h new file mode 100644 index 00000000..4b4a76a4 --- /dev/null +++ b/engine/voice_codecs/speex/speex/speexdsp_types.h @@ -0,0 +1,126 @@ +/* speexdsp_types.h taken from libogg */ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: #ifdef jail to whip a few platforms into the UNIX ideal. + last mod: $Id: os_types.h 7524 2004-08-11 04:20:36Z conrad $ + + ********************************************************************/ +/** + @file speexdsp_types.h + @brief Speex types +*/ +#ifndef _SPEEX_TYPES_H +#define _SPEEX_TYPES_H + +#if defined(_WIN32) + +# if defined(__CYGWIN__) +# include <_G_config.h> + typedef _G_int32_t spx_int32_t; + typedef _G_uint32_t spx_uint32_t; + typedef _G_int16_t spx_int16_t; + typedef _G_uint16_t spx_uint16_t; +# elif defined(__MINGW32__) + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; +# elif defined(__MWERKS__) + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; +# else + /* MSVC/Borland */ + typedef __int32 spx_int32_t; + typedef unsigned __int32 spx_uint32_t; + typedef __int16 spx_int16_t; + typedef unsigned __int16 spx_uint16_t; +# endif + +#elif defined(__MACOS__) + +# include + typedef SInt16 spx_int16_t; + typedef UInt16 spx_uint16_t; + typedef SInt32 spx_int32_t; + typedef UInt32 spx_uint32_t; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined(__BEOS__) + + /* Be */ +# include + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short spx_int16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(R5900) + + /* PS2 EE */ + typedef int spx_int32_t; + typedef unsigned spx_uint32_t; + typedef short spx_int16_t; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef signed int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef long spx_int32_t; + typedef unsigned long spx_uint32_t; + +#elif defined(CONFIG_TI_C6X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#else + +#include "speexdsp_config_types.h" + +#endif + +#endif /* _SPEEX_TYPES_H */ diff --git a/engine/voice_codecs/speex/wscript b/engine/voice_codecs/speex/wscript new file mode 100755 index 00000000..d1b1f581 --- /dev/null +++ b/engine/voice_codecs/speex/wscript @@ -0,0 +1,54 @@ +#! /usr/bin/env python +# encoding: utf-8 + +from waflib import Utils +import os + +top = '.' +PROJECT_NAME = 'vaudio_speex' + +def options(opt): + # stub + return + +def configure(conf): + conf.define('SPEEX_EXPORTS',1) +# conf.define('NO_HOOK_MALLOC',1) + +def build(bld): + source = [ + 'VoiceEncoder_Speex.cpp', + '../frame_encoder/voice_codec_frame.cpp', + '../../../tier1/interface.cpp' + ] + + includes = [ + '.', + '../frame_encoder', + '../../../public', + '../../../public/tier1', + '../../../public/tier0', + '../../', + '../../../common', + '../../audio/public' + ] + + defines = [] + + libs = ['tier0','tier1','vstdlib','SPEEX'] + + install_path = bld.env.LIBDIR + + bld.shlib( + source = source, + target = PROJECT_NAME, + name = PROJECT_NAME, + features = 'c cxx', + includes = includes, + defines = defines, + use = libs, + install_path = install_path, + subsystem = bld.env.MSVC_SUBSYSTEM, + idx = bld.get_taskgen_count() + ) + diff --git a/engine/wscript b/engine/wscript index 29a03ad8..161d1f4d 100755 --- a/engine/wscript +++ b/engine/wscript @@ -22,7 +22,8 @@ def configure(conf): 'ENGINE_DLL', 'VERSION_SAFE_STEAM_API_INTERFACES', 'USE_BREAKPAD_HANDLER', - 'USE_CONVARS' + 'USE_CONVARS', + 'VOICE_OVER_IP' ]) def build(bld): @@ -311,7 +312,7 @@ def build(bld): 'vgui_helpers.cpp', 'vgui_texturebudgetpanel.cpp', 'vgui_vprofgraphpanel.cpp', - 'vgui_vprofpanel.cpp', + 'vgui_vprofpanel.cpp', 'toolframework.cpp' ] diff --git a/scripts/waifulib/compiler_optimizations.py b/scripts/waifulib/compiler_optimizations.py index 56c35c1e..f5fb4744 100644 --- a/scripts/waifulib/compiler_optimizations.py +++ b/scripts/waifulib/compiler_optimizations.py @@ -48,7 +48,7 @@ CFLAGS = { 'common': { # disable thread-safe local static initialization for C++11 code, as it cause crashes on Windows XP 'msvc': ['/D_USING_V110_SDK71_', '/Zi', '/FS', '/Zc:threadSafeInit-', '/MT'], - 'clang': ['-g0', '-gdwarf-2', '-fvisibility=hidden'], + 'clang': ['-g', '-gdwarf-2', '-fvisibility=hidden'], 'gcc': ['-g0', '-fvisibility=hidden'], 'owcc': ['-fno-short-enum', '-ffloat-store', '-g3'] }, @@ -75,7 +75,7 @@ CFLAGS = { 'debug': { 'msvc': ['/Od'], 'owcc': ['-O0', '-fno-omit-frame-pointer', '-funwind-tables', '-fno-omit-leaf-frame-pointer'], - 'default': ['-O2', '-ftree-vectorize'] + 'default': ['-O3'] #, '-ftree-vectorize'] }, 'sanitize': { 'msvc': ['/Od', '/RTC1'], diff --git a/wscript b/wscript index bbe4e113..6624260e 100644 --- a/wscript +++ b/wscript @@ -38,6 +38,7 @@ projects={ 'dmxloader', 'engine', 'engine/voice_codecs/minimp3', + 'engine/voice_codecs/opus', 'filesystem', 'game/client', 'game/server', @@ -269,7 +270,7 @@ def configure(conf): cflags, linkflags = conf.get_optimization_flags() - flags = ['-fPIC'] + flags = ['-fPIC'] #, '-fsanitize=undefined'] if conf.env.DEST_OS == 'android': flags += [ @@ -340,6 +341,8 @@ def configure(conf): conf.check_cfg(package='libpng', uselib_store='PNG', args=['--cflags', '--libs']) conf.check_cfg(package='libcurl', uselib_store='CURL', args=['--cflags', '--libs']) conf.check_cfg(package='zlib', uselib_store='ZLIB', args=['--cflags', '--libs']) + conf.check_cfg(package='opus', uselib_store='OPUS', args=['--cflags', '--libs']) +# conf.check_cfg(package='speex', uselib_store='SPEEX', args=['--cflags', '--libs']) else: conf.check(lib='SDL2', uselib_store='SDL2') conf.check(lib='freetype2', uselib_store='FT2') @@ -352,6 +355,8 @@ def configure(conf): conf.check(lib='ssl', uselib_store='SSL') conf.check(lib='expat', uselib_store='EXPAT') conf.check(lib='android_support', uselib_store='ANDROID_SUPPORT') + conf.check(lib='opus', uselib_store='OPUS') +# conf.check(lib='speex', uselib_store='SPEEX') if conf.env.DEST_OS != 'win32': conf.check_cc(lib='dl', mandatory=False)