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/*
* 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 */
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