FFmpeg/libavcodec/opus.h
Rostislav Pehlivanov 07b78340dd opus_celt: rename structures to better names and reorganize them
This is meant to be applied on top of my previous patch which
split PVQ into celt_pvq.c and made opus_celt.h

Essentially nothing has been changed other than renaming CeltFrame
to CeltBlock (CeltFrame had absolutely nothing at all to do with
a frame) and CeltContext to CeltFrame.
3 variables have been put in CeltFrame as they make more sense
there rather than being passed around as arguments.
The coefficients have been moved to the CeltBlock structure
(why the hell were they in CeltContext and not in CeltFrame??).

Now the encoder would be able to use the exact context the decoder
uses (plus a couple of extra fields in there).

FATE passes, no slowdowns, etc.

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2017-02-14 06:15:36 +00:00

193 lines
5.8 KiB
C

/*
* 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 */