third_party_ffmpeg/libavcodec/opus.h

/*
 * Opus decoder/demuxer common functions
 * Copyright (c) 2012 Andrew D'Addesio
 * Copyright (c) 2013-2014 Mozilla Corporation
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#ifndef AVCODEC_OPUS_H
#define AVCODEC_OPUS_H

#include <stdint.h>

#include "libavutil/audio_fifo.h"
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"

#include "libswresample/swresample.h"

#include "avcodec.h"
#include "opus_rc.h"

#define MAX_FRAME_SIZE               1275
#define MAX_FRAMES                   48
#define MAX_PACKET_DUR               5760

#define CELT_SHORT_BLOCKSIZE         120
#define CELT_OVERLAP                 CELT_SHORT_BLOCKSIZE
#define CELT_MAX_LOG_BLOCKS          3
#define CELT_MAX_FRAME_SIZE          (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS))
#define CELT_MAX_BANDS               21

#define SILK_HISTORY                 322
#define SILK_MAX_LPC                 16

#define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1)
#define ROUND_MUL16(a,b)  ((MUL16(a, b) + 16384) >> 15)

#define OPUS_TS_HEADER     0x7FE0        // 0x3ff (11 bits)
#define OPUS_TS_MASK       0xFFE0        // top 11 bits

static const uint8_t opus_default_extradata[30] = {
    'O', 'p', 'u', 's', 'H', 'e', 'a', 'd',
    1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
    0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
};

enum OpusMode {
    OPUS_MODE_SILK,
    OPUS_MODE_HYBRID,
    OPUS_MODE_CELT,

    OPUS_MODE_NB
};

enum OpusBandwidth {
    OPUS_BANDWIDTH_NARROWBAND,
    OPUS_BANDWIDTH_MEDIUMBAND,
    OPUS_BANDWIDTH_WIDEBAND,
    OPUS_BANDWIDTH_SUPERWIDEBAND,
    OPUS_BANDWIDTH_FULLBAND,

    OPUS_BANDWITH_NB
};

typedef struct SilkContext SilkContext;

typedef struct CeltFrame CeltFrame;

typedef struct OpusPacket {
    int packet_size;                /**< packet size */
    int data_size;                  /**< size of the useful data -- packet size - padding */
    int code;                       /**< packet code: specifies the frame layout */
    int stereo;                     /**< whether this packet is mono or stereo */
    int vbr;                        /**< vbr flag */
    int config;                     /**< configuration: tells the audio mode,
                                     **                bandwidth, and frame duration */
    int frame_count;                /**< frame count */
    int frame_offset[MAX_FRAMES];   /**< frame offsets */
    int frame_size[MAX_FRAMES];     /**< frame sizes */
    int frame_duration;             /**< frame duration, in samples @ 48kHz */
    enum OpusMode mode;             /**< mode */
    enum OpusBandwidth bandwidth;   /**< bandwidth */
} OpusPacket;

typedef struct OpusStreamContext {
    AVCodecContext *avctx;
    int output_channels;

    OpusRangeCoder rc;
    OpusRangeCoder redundancy_rc;
    SilkContext *silk;
    CeltFrame *celt;
    AVFloatDSPContext *fdsp;

    float silk_buf[2][960];
    float *silk_output[2];
    DECLARE_ALIGNED(32, float, celt_buf)[2][960];
    float *celt_output[2];

    float redundancy_buf[2][960];
    float *redundancy_output[2];

    /* data buffers for the final output data */
    float *out[2];
    int out_size;

    float *out_dummy;
    int    out_dummy_allocated_size;

    SwrContext *swr;
    AVAudioFifo *celt_delay;
    int silk_samplerate;
    /* number of samples we still want to get from the resampler */
    int delayed_samples;

    OpusPacket packet;

    int redundancy_idx;
} OpusStreamContext;

// a mapping between an opus stream and an output channel
typedef struct ChannelMap {
    int stream_idx;
    int channel_idx;

    // when a single decoded channel is mapped to multiple output channels, we
    // write to the first output directly and copy from it to the others
    // this field is set to 1 for those copied output channels
    int copy;
    // this is the index of the output channel to copy from
    int copy_idx;

    // this channel is silent
    int silence;
} ChannelMap;

typedef struct OpusContext {
    OpusStreamContext *streams;

    /* current output buffers for each streams */
    float **out;
    int   *out_size;
    /* Buffers for synchronizing the streams when they have different
     * resampling delays */
    AVAudioFifo **sync_buffers;
    /* number of decoded samples for each stream */
    int         *decoded_samples;

    int             nb_streams;
    int      nb_stereo_streams;

    AVFloatDSPContext *fdsp;
    int16_t gain_i;
    float   gain;

    ChannelMap *channel_maps;
} OpusContext;

int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
                         int self_delimited);

int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s);

int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels);
void ff_silk_free(SilkContext **ps);
void ff_silk_flush(SilkContext *s);

/**
 * Decode the LP layer of one Opus frame (which may correspond to several SILK
 * frames).
 */
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
                              float *output[2],
                              enum OpusBandwidth bandwidth, int coded_channels,
                              int duration_ms);

#endif /* AVCODEC_OPUS_H */