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FFmpeg/libavcodec/opus.h

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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 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 "get_bits.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 {
GetBitContext gb;
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) | (get_bits(&rc->gb, 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 */