|
|
|
/*
|
|
|
|
* Opus decoder/demuxer common functions
|
|
|
|
* Copyright (c) 2012 Andrew D'Addesio
|
|
|
|
* Copyright (c) 2013-2014 Mozilla Corporation
|
|
|
|
*
|
|
|
|
* This file is part of Libav.
|
|
|
|
*
|
|
|
|
* Libav 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.
|
|
|
|
*
|
|
|
|
* Libav 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 Libav; 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 "libavresample/avresample.h"
|
|
|
|
|
|
|
|
#include "avcodec.h"
|
|
|
|
#include "bitstream.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 CELT_VECTORS 11
|
|
|
|
#define CELT_ALLOC_STEPS 6
|
|
|
|
#define CELT_FINE_OFFSET 21
|
|
|
|
#define CELT_MAX_FINE_BITS 8
|
|
|
|
#define CELT_NORM_SCALE 16384
|
|
|
|
#define CELT_QTHETA_OFFSET 4
|
|
|
|
#define CELT_QTHETA_OFFSET_TWOPHASE 16
|
|
|
|
#define CELT_DEEMPH_COEFF 0.85000610f
|
|
|
|
#define CELT_POSTFILTER_MINPERIOD 15
|
|
|
|
#define CELT_ENERGY_SILENCE (-28.0f)
|
|
|
|
|
|
|
|
#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_ilog(i) (av_log2(i) + !!(i))
|
|
|
|
|
|
|
|
#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
|
|
|
|
};
|
|
|
|
|
|
|
|
enum OpusBandwidth {
|
|
|
|
OPUS_BANDWIDTH_NARROWBAND,
|
|
|
|
OPUS_BANDWIDTH_MEDIUMBAND,
|
|
|
|
OPUS_BANDWIDTH_WIDEBAND,
|
|
|
|
OPUS_BANDWIDTH_SUPERWIDEBAND,
|
|
|
|
OPUS_BANDWIDTH_FULLBAND
|
|
|
|
};
|
|
|
|
|
|
|
|
typedef struct RawBitsContext {
|
|
|
|
const uint8_t *position;
|
|
|
|
unsigned int bytes;
|
|
|
|
unsigned int cachelen;
|
|
|
|
unsigned int cacheval;
|
|
|
|
} RawBitsContext;
|
|
|
|
|
|
|
|
typedef struct OpusRangeCoder {
|
|
|
|
BitstreamContext bc;
|
|
|
|
RawBitsContext rb;
|
|
|
|
unsigned int range;
|
|
|
|
unsigned int value;
|
|
|
|
unsigned int total_read_bits;
|
|
|
|
} OpusRangeCoder;
|
|
|
|
|
|
|
|
typedef struct SilkContext SilkContext;
|
|
|
|
|
|
|
|
typedef struct CeltContext CeltContext;
|
|
|
|
|
|
|
|
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;
|
|
|
|
CeltContext *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;
|
|
|
|
|
|
|
|
AVAudioResampleContext *avr;
|
|
|
|
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;
|
|
|
|
|
|
|
|
static av_always_inline void opus_rc_normalize(OpusRangeCoder *rc)
|
|
|
|
{
|
|
|
|
while (rc->range <= 1<<23) {
|
|
|
|
rc->value = ((rc->value << 8) | (bitstream_read(&rc->bc, 8) ^ 0xFF)) & ((1u << 31) - 1);
|
|
|
|
rc->range <<= 8;
|
|
|
|
rc->total_read_bits += 8;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline void opus_rc_update(OpusRangeCoder *rc, unsigned int scale,
|
|
|
|
unsigned int low, unsigned int high,
|
|
|
|
unsigned int total)
|
|
|
|
{
|
|
|
|
rc->value -= scale * (total - high);
|
|
|
|
rc->range = low ? scale * (high - low)
|
|
|
|
: rc->range - scale * (total - high);
|
|
|
|
opus_rc_normalize(rc);
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline unsigned int opus_rc_getsymbol(OpusRangeCoder *rc, const uint16_t *cdf)
|
|
|
|
{
|
|
|
|
unsigned int k, scale, total, symbol, low, high;
|
|
|
|
|
|
|
|
total = *cdf++;
|
|
|
|
|
|
|
|
scale = rc->range / total;
|
|
|
|
symbol = rc->value / scale + 1;
|
|
|
|
symbol = total - FFMIN(symbol, total);
|
|
|
|
|
|
|
|
for (k = 0; cdf[k] <= symbol; k++);
|
|
|
|
high = cdf[k];
|
|
|
|
low = k ? cdf[k-1] : 0;
|
|
|
|
|
|
|
|
opus_rc_update(rc, scale, low, high, total);
|
|
|
|
|
|
|
|
return k;
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline unsigned int opus_rc_p2model(OpusRangeCoder *rc, unsigned int bits)
|
|
|
|
{
|
|
|
|
unsigned int k, scale;
|
|
|
|
scale = rc->range >> bits; // in this case, scale = symbol
|
|
|
|
|
|
|
|
if (rc->value >= scale) {
|
|
|
|
rc->value -= scale;
|
|
|
|
rc->range -= scale;
|
|
|
|
k = 0;
|
|
|
|
} else {
|
|
|
|
rc->range = scale;
|
|
|
|
k = 1;
|
|
|
|
}
|
|
|
|
opus_rc_normalize(rc);
|
|
|
|
return k;
|
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* CELT: estimate bits of entropy that have thus far been consumed for the
|
|
|
|
* current CELT frame, to integer and fractional (1/8th bit) precision
|
|
|
|
*/
|
|
|
|
static av_always_inline unsigned int opus_rc_tell(const OpusRangeCoder *rc)
|
|
|
|
{
|
|
|
|
return rc->total_read_bits - av_log2(rc->range) - 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline unsigned int opus_rc_tell_frac(const OpusRangeCoder *rc)
|
|
|
|
{
|
|
|
|
unsigned int i, total_bits, rcbuffer, range;
|
|
|
|
|
|
|
|
total_bits = rc->total_read_bits << 3;
|
|
|
|
rcbuffer = av_log2(rc->range) + 1;
|
|
|
|
range = rc->range >> (rcbuffer-16);
|
|
|
|
|
|
|
|
for (i = 0; i < 3; i++) {
|
|
|
|
int bit;
|
|
|
|
range = range * range >> 15;
|
|
|
|
bit = range >> 16;
|
|
|
|
rcbuffer = rcbuffer << 1 | bit;
|
|
|
|
range >>= bit;
|
|
|
|
}
|
|
|
|
|
|
|
|
return total_bits - rcbuffer;
|
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* CELT: read 1-25 raw bits at the end of the frame, backwards byte-wise
|
|
|
|
*/
|
|
|
|
static av_always_inline unsigned int opus_getrawbits(OpusRangeCoder *rc, unsigned int count)
|
|
|
|
{
|
|
|
|
unsigned int value = 0;
|
|
|
|
|
|
|
|
while (rc->rb.bytes && rc->rb.cachelen < count) {
|
|
|
|
rc->rb.cacheval |= *--rc->rb.position << rc->rb.cachelen;
|
|
|
|
rc->rb.cachelen += 8;
|
|
|
|
rc->rb.bytes--;
|
|
|
|
}
|
|
|
|
|
|
|
|
value = rc->rb.cacheval & ((1<<count)-1);
|
|
|
|
rc->rb.cacheval >>= count;
|
|
|
|
rc->rb.cachelen -= count;
|
|
|
|
rc->total_read_bits += count;
|
|
|
|
|
|
|
|
return value;
|
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* CELT: read a uniform distribution
|
|
|
|
*/
|
|
|
|
static av_always_inline unsigned int opus_rc_unimodel(OpusRangeCoder *rc, unsigned int size)
|
|
|
|
{
|
|
|
|
unsigned int bits, k, scale, total;
|
|
|
|
|
|
|
|
bits = opus_ilog(size - 1);
|
|
|
|
total = (bits > 8) ? ((size - 1) >> (bits - 8)) + 1 : size;
|
|
|
|
|
|
|
|
scale = rc->range / total;
|
|
|
|
k = rc->value / scale + 1;
|
|
|
|
k = total - FFMIN(k, total);
|
|
|
|
opus_rc_update(rc, scale, k, k + 1, total);
|
|
|
|
|
|
|
|
if (bits > 8) {
|
|
|
|
k = k << (bits - 8) | opus_getrawbits(rc, bits - 8);
|
|
|
|
return FFMIN(k, size - 1);
|
|
|
|
} else
|
|
|
|
return k;
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline int opus_rc_laplace(OpusRangeCoder *rc, unsigned int symbol, int decay)
|
|
|
|
{
|
|
|
|
/* extends the range coder to model a Laplace distribution */
|
|
|
|
int value = 0;
|
|
|
|
unsigned int scale, low = 0, center;
|
|
|
|
|
|
|
|
scale = rc->range >> 15;
|
|
|
|
center = rc->value / scale + 1;
|
|
|
|
center = (1 << 15) - FFMIN(center, 1 << 15);
|
|
|
|
|
|
|
|
if (center >= symbol) {
|
|
|
|
value++;
|
|
|
|
low = symbol;
|
|
|
|
symbol = 1 + ((32768 - 32 - symbol) * (16384-decay) >> 15);
|
|
|
|
|
|
|
|
while (symbol > 1 && center >= low + 2 * symbol) {
|
|
|
|
value++;
|
|
|
|
symbol *= 2;
|
|
|
|
low += symbol;
|
|
|
|
symbol = (((symbol - 2) * decay) >> 15) + 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (symbol <= 1) {
|
|
|
|
int distance = (center - low) >> 1;
|
|
|
|
value += distance;
|
|
|
|
low += 2 * distance;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (center < low + symbol)
|
|
|
|
value *= -1;
|
|
|
|
else
|
|
|
|
low += symbol;
|
|
|
|
}
|
|
|
|
|
|
|
|
opus_rc_update(rc, scale, low, FFMIN(low + symbol, 32768), 32768);
|
|
|
|
|
|
|
|
return value;
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline unsigned int opus_rc_stepmodel(OpusRangeCoder *rc, int k0)
|
|
|
|
{
|
|
|
|
/* Use a probability of 3 up to itheta=8192 and then use 1 after */
|
|
|
|
unsigned int k, scale, symbol, total = (k0+1)*3 + k0;
|
|
|
|
scale = rc->range / total;
|
|
|
|
symbol = rc->value / scale + 1;
|
|
|
|
symbol = total - FFMIN(symbol, total);
|
|
|
|
|
|
|
|
k = (symbol < (k0+1)*3) ? symbol/3 : symbol - (k0+1)*2;
|
|
|
|
|
|
|
|
opus_rc_update(rc, scale, (k <= k0) ? 3*(k+0) : (k-1-k0) + 3*(k0+1),
|
|
|
|
(k <= k0) ? 3*(k+1) : (k-0-k0) + 3*(k0+1), total);
|
|
|
|
return k;
|
|
|
|
}
|
|
|
|
|
|
|
|
static av_always_inline unsigned int opus_rc_trimodel(OpusRangeCoder *rc, int qn)
|
|
|
|
{
|
|
|
|
unsigned int k, scale, symbol, total, low, center;
|
|
|
|
|
|
|
|
total = ((qn>>1) + 1) * ((qn>>1) + 1);
|
|
|
|
scale = rc->range / total;
|
|
|
|
center = rc->value / scale + 1;
|
|
|
|
center = total - FFMIN(center, total);
|
|
|
|
|
|
|
|
if (center < total >> 1) {
|
|
|
|
k = (ff_sqrt(8 * center + 1) - 1) >> 1;
|
|
|
|
low = k * (k + 1) >> 1;
|
|
|
|
symbol = k + 1;
|
|
|
|
} else {
|
|
|
|
k = (2*(qn + 1) - ff_sqrt(8*(total - center - 1) + 1)) >> 1;
|
|
|
|
low = total - ((qn + 1 - k) * (qn + 2 - k) >> 1);
|
|
|
|
symbol = qn + 1 - k;
|
|
|
|
}
|
|
|
|
|
|
|
|
opus_rc_update(rc, scale, low, low + symbol, total);
|
|
|
|
|
|
|
|
return k;
|
|
|
|
}
|
|
|
|
|
|
|
|
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);
|
|
|
|
|
|
|
|
int ff_celt_init(AVCodecContext *avctx, CeltContext **s, int output_channels);
|
|
|
|
|
|
|
|
void ff_celt_free(CeltContext **s);
|
|
|
|
|
|
|
|
void ff_celt_flush(CeltContext *s);
|
|
|
|
|
|
|
|
int ff_celt_decode_frame(CeltContext *s, OpusRangeCoder *rc,
|
|
|
|
float **output, int coded_channels, int frame_size,
|
|
|
|
int startband, int endband);
|
|
|
|
|
|
|
|
extern const float ff_celt_window2[120];
|
|
|
|
|
|
|
|
#endif /* AVCODEC_OPUS_H */
|