Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity

Merge commit 'b70d7a4ac72d23f3448f3b08b770fdf5f57de222'

Browse Source 
* commit 'b70d7a4ac72d23f3448f3b08b770fdf5f57de222':
  lavc: add a native Opus decoder.

Conflicts:
	Changelog
	configure
	libavcodec/version.h

Fate tests pass with both avresample as well as swresample based opus decoder, but
are disabled (reference files are very large so i want to think a day or 2 about
if theres an alternative or if they could be avoided, they also dont match the
official samples)

Merged-by: Michael Niedermayer <michaelni@gmx.at>
pull/68/head
Michael Niedermayer 11 years ago
parent 8b96f31817 b70d7a4ac7
commit 2c7d3ecfc9
14 changed files with 5743 additions and 2 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 1
      Changelog
  2. 3
      configure
  3. 4
      libavcodec/Makefile
  4. 2
      libavcodec/allcodecs.c
  5. 428
      libavcodec/opus.c
  6. 429
      libavcodec/opus.h
  7. 2220
      libavcodec/opus_celt.c
  8. 268
      libavcodec/opus_imdct.c
  9. 75
      libavcodec/opus_parser.c
  10. 1597
      libavcodec/opus_silk.c
  11. 674
      libavcodec/opusdec.c
  12. 4
      libavcodec/version.h
  13. 1
      tests/Makefile
  14. 39
      tests/fate/opus.mak

1
Changelog
Unescape View File

@ -23,6 +23,7 @@ version <next>:
- support for decoding through DXVA2 in ffmpeg
- libbs2b-based stereo-to-binaural audio filter
- libx264 reference frames count limiting depending on level
- native Opus decoder
version 2.2:

3
configure vendored
Unescape View File

@ -2110,6 +2110,7 @@ nellymoser_decoder_select="mdct sinewin"
nellymoser_encoder_select="audio_frame_queue mdct sinewin"
nuv_decoder_select="dsputil lzo"
on2avc_decoder_select="mdct"
opus_decoder_deps="avresample"
png_decoder_select="zlib"
png_encoder_select="dsputil zlib"
prores_decoder_select="dsputil"
@ -5139,6 +5140,8 @@ enabled subtitles_filter && prepend avfilter_deps "avformat avcodec"
enabled lavfi_indev && prepend avdevice_deps "avfilter"
enabled opus_decoder && prepend avcodec_deps "avresample"
expand_deps(){
lib_deps=${1}_deps
eval "deps=\$$lib_deps"

4
libavcodec/Makefile
Unescape View File

@ -332,6 +332,9 @@ OBJS-$(CONFIG_NELLYMOSER_DECODER) += nellymoserdec.o nellymoser.o
OBJS-$(CONFIG_NELLYMOSER_ENCODER) += nellymoserenc.o nellymoser.o
OBJS-$(CONFIG_NUV_DECODER) += nuv.o rtjpeg.o
OBJS-$(CONFIG_ON2AVC_DECODER) += on2avc.o on2avcdata.o
OBJS-$(CONFIG_OPUS_DECODER) += opusdec.o opus.o opus_celt.o \
opus_imdct.o opus_silk.o \
vorbis_data.o
OBJS-$(CONFIG_PAF_VIDEO_DECODER) += paf.o
OBJS-$(CONFIG_PAF_AUDIO_DECODER) += paf.o
OBJS-$(CONFIG_PAM_DECODER) += pnmdec.o pnm.o
@ -779,6 +782,7 @@ OBJS-$(CONFIG_MPEGAUDIO_PARSER) += mpegaudio_parser.o \
mpegaudiodecheader.o mpegaudiodata.o
OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o \
mpeg12.o mpeg12data.o
OBJS-$(CONFIG_OPUS_PARSER) += opus_parser.o opus.o vorbis_data.o
OBJS-$(CONFIG_PNG_PARSER) += png_parser.o
OBJS-$(CONFIG_PNM_PARSER) += pnm_parser.o pnm.o
OBJS-$(CONFIG_RV30_PARSER) += rv34_parser.o

2
libavcodec/allcodecs.c
Unescape View File

@ -372,6 +372,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(MPC8, mpc8);
REGISTER_ENCDEC (NELLYMOSER, nellymoser);
REGISTER_DECODER(ON2AVC, on2avc);
REGISTER_DECODER(OPUS, opus);
REGISTER_DECODER(PAF_AUDIO, paf_audio);
REGISTER_DECODER(QCELP, qcelp);
REGISTER_DECODER(QDM2, qdm2);
@ -562,6 +563,7 @@ void avcodec_register_all(void)
REGISTER_PARSER(MPEG4VIDEO, mpeg4video);
REGISTER_PARSER(MPEGAUDIO, mpegaudio);
REGISTER_PARSER(MPEGVIDEO, mpegvideo);
REGISTER_PARSER(OPUS, opus);
REGISTER_PARSER(PNG, png);
REGISTER_PARSER(PNM, pnm);
REGISTER_PARSER(RV30, rv30);

428
libavcodec/opus.c
Unescape View File

@ -0,0 +1,428 @@
/*
* 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
*/
/**
* @file
* Opus decoder/parser shared code
*/
#include <stdint.h>
#include "libavutil/error.h"
#include "opus.h"
#include "vorbis.h"
static const uint16_t opus_frame_duration[32] = {
480, 960, 1920, 2880,
480, 960, 1920, 2880,
480, 960, 1920, 2880,
480, 960,
480, 960,
120, 240, 480, 960,
120, 240, 480, 960,
120, 240, 480, 960,
120, 240, 480, 960,
};
/**
* Read a 1- or 2-byte frame length
*/
static inline int xiph_lacing_16bit(const uint8_t **ptr, const uint8_t *end)
{
int val;
if (*ptr >= end)
return AVERROR_INVALIDDATA;
val = *(*ptr)++;
if (val >= 252) {
if (*ptr >= end)
return AVERROR_INVALIDDATA;
val += 4 * *(*ptr)++;
}
return val;
}
/**
* Read a multi-byte length (used for code 3 packet padding size)
*/
static inline int xiph_lacing_full(const uint8_t **ptr, const uint8_t *end)
{
int val = 0;
int next;
while (1) {
if (*ptr >= end || val > INT_MAX - 254)
return AVERROR_INVALIDDATA;
next = *(*ptr)++;
val += next;
if (next < 255)
break;
else
val--;
}
return val;
}
/**
* Parse Opus packet info from raw packet data
*/
int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
int self_delimiting)
{
const uint8_t *ptr = buf;
const uint8_t *end = buf + buf_size;
int padding = 0;
int frame_bytes, i;
if (buf_size < 1)
goto fail;
/* TOC byte */
i = *ptr++;
pkt->code = (i ) & 0x3;
pkt->stereo = (i >> 2) & 0x1;
pkt->config = (i >> 3) & 0x1F;
/* code 2 and code 3 packets have at least 1 byte after the TOC */
if (pkt->code >= 2 && buf_size < 2)
goto fail;
switch (pkt->code) {
case 0:
/* 1 frame */
pkt->frame_count = 1;
pkt->vbr = 0;
if (self_delimiting) {
int len = xiph_lacing_16bit(&ptr, end);
if (len < 0 || len > end - ptr)
goto fail;
end = ptr + len;
buf_size = end - buf;
}
frame_bytes = end - ptr;
if (frame_bytes > MAX_FRAME_SIZE)
goto fail;
pkt->frame_offset[0] = ptr - buf;
pkt->frame_size[0] = frame_bytes;
break;
case 1:
/* 2 frames, equal size */
pkt->frame_count = 2;
pkt->vbr = 0;
if (self_delimiting) {
int len = xiph_lacing_16bit(&ptr, end);
if (len < 0 || 2 * len > end - ptr)
goto fail;
end = ptr + 2 * len;
buf_size = end - buf;
}
frame_bytes = end - ptr;
if (frame_bytes & 1 || frame_bytes >> 1 > MAX_FRAME_SIZE)
goto fail;
pkt->frame_offset[0] = ptr - buf;
pkt->frame_size[0] = frame_bytes >> 1;
pkt->frame_offset[1] = pkt->frame_offset[0] + pkt->frame_size[0];
pkt->frame_size[1] = frame_bytes >> 1;
break;
case 2:
/* 2 frames, different sizes */
pkt->frame_count = 2;
pkt->vbr = 1;
/* read 1st frame size */
frame_bytes = xiph_lacing_16bit(&ptr, end);
if (frame_bytes < 0)
goto fail;
if (self_delimiting) {
int len = xiph_lacing_16bit(&ptr, end);
if (len < 0 || len + frame_bytes > end - ptr)
goto fail;
end = ptr + frame_bytes + len;
buf_size = end - buf;
}
pkt->frame_offset[0] = ptr - buf;
pkt->frame_size[0] = frame_bytes;
/* calculate 2nd frame size */
frame_bytes = end - ptr - pkt->frame_size[0];
if (frame_bytes < 0 || frame_bytes > MAX_FRAME_SIZE)
goto fail;
pkt->frame_offset[1] = pkt->frame_offset[0] + pkt->frame_size[0];
pkt->frame_size[1] = frame_bytes;
break;
case 3:
/* 1 to 48 frames, can be different sizes */
i = *ptr++;
pkt->frame_count = (i ) & 0x3F;
padding = (i >> 6) & 0x01;
pkt->vbr = (i >> 7) & 0x01;
if (pkt->frame_count == 0 || pkt->frame_count > MAX_FRAMES)
goto fail;
/* read padding size */
if (padding) {
padding = xiph_lacing_full(&ptr, end);
if (padding < 0)
goto fail;
}
/* read frame sizes */
if (pkt->vbr) {
/* for VBR, all frames except the final one have their size coded
in the bitstream. the last frame size is implicit. */
int total_bytes = 0;
for (i = 0; i < pkt->frame_count - 1; i++) {
frame_bytes = xiph_lacing_16bit(&ptr, end);
if (frame_bytes < 0)
goto fail;
pkt->frame_size[i] = frame_bytes;
total_bytes += frame_bytes;
}
if (self_delimiting) {
int len = xiph_lacing_16bit(&ptr, end);
if (len < 0 || len + total_bytes + padding > end - ptr)
goto fail;
end = ptr + total_bytes + len + padding;
buf_size = end - buf;
}
frame_bytes = end - ptr - padding;
if (total_bytes > frame_bytes)
goto fail;
pkt->frame_offset[0] = ptr - buf;
for (i = 1; i < pkt->frame_count; i++)
pkt->frame_offset[i] = pkt->frame_offset[i-1] + pkt->frame_size[i-1];
pkt->frame_size[pkt->frame_count-1] = frame_bytes - total_bytes;
} else {
/* for CBR, the remaining packet bytes are divided evenly between
the frames */
if (self_delimiting) {
frame_bytes = xiph_lacing_16bit(&ptr, end);
if (frame_bytes < 0 || pkt->frame_count * frame_bytes + padding > end - ptr)
goto fail;
end = ptr + pkt->frame_count * frame_bytes + padding;
buf_size = end - buf;
} else {
frame_bytes = end - ptr - padding;
if (frame_bytes % pkt->frame_count ||
frame_bytes / pkt->frame_count > MAX_FRAME_SIZE)
goto fail;
frame_bytes /= pkt->frame_count;
}
pkt->frame_offset[0] = ptr - buf;
pkt->frame_size[0] = frame_bytes;
for (i = 1; i < pkt->frame_count; i++) {
pkt->frame_offset[i] = pkt->frame_offset[i-1] + pkt->frame_size[i-1];
pkt->frame_size[i] = frame_bytes;
}
}
}
pkt->packet_size = buf_size;
pkt->data_size = pkt->packet_size - padding;
/* total packet duration cannot be larger than 120ms */
pkt->frame_duration = opus_frame_duration[pkt->config];
if (pkt->frame_duration * pkt->frame_count > MAX_PACKET_DUR)
goto fail;
/* set mode and bandwidth */
if (pkt->config < 12) {
pkt->mode = OPUS_MODE_SILK;
pkt->bandwidth = pkt->config >> 2;
} else if (pkt->config < 16) {
pkt->mode = OPUS_MODE_HYBRID;
pkt->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND + (pkt->config >= 14);
} else {
pkt->mode = OPUS_MODE_CELT;
pkt->bandwidth = (pkt->config - 16) >> 2;
/* skip mediumband */
if (pkt->bandwidth)
pkt->bandwidth++;
}
return 0;
fail:
memset(pkt, 0, sizeof(*pkt));
return AVERROR_INVALIDDATA;
}
static int channel_reorder_vorbis(int nb_channels, int channel_idx)
{
return ff_vorbis_channel_layout_offsets[nb_channels - 1][channel_idx];
}
static int channel_reorder_unknown(int nb_channels, int channel_idx)
{
return channel_idx;
}
av_cold int ff_opus_parse_extradata(AVCodecContext *avctx,
OpusContext *s)
{
static const uint8_t default_channel_map[2] = { 0, 1 };
uint8_t default_extradata[19] = {
'O', 'p', 'u', 's', 'H', 'e', 'a', 'd',
1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
};
int (*channel_reorder)(int, int) = channel_reorder_unknown;
const uint8_t *extradata, *channel_map;
int extradata_size;
int version, channels, map_type, streams, stereo_streams, i, j;
uint64_t layout;
if (!avctx->extradata) {
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR,
"Multichannel configuration without extradata.\n");
return AVERROR(EINVAL);
}
default_extradata[9] = (avctx->channels == 1) ? 1 : 2;
extradata = default_extradata;
extradata_size = sizeof(default_extradata);
} else {
extradata = avctx->extradata;
extradata_size = avctx->extradata_size;
}
if (extradata_size < 19) {
av_log(avctx, AV_LOG_ERROR, "Invalid extradata size: %d\n",
extradata_size);
return AVERROR_INVALIDDATA;
}
version = extradata[8];
if (version > 15) {
avpriv_request_sample(avctx, "Extradata version %d", version);
return AVERROR_PATCHWELCOME;
}
avctx->delay = AV_RL16(extradata + 10);
channels = extradata[9];
if (!channels) {
av_log(avctx, AV_LOG_ERROR, "Zero channel count specified in the extadata\n");
return AVERROR_INVALIDDATA;
}
s->gain_i = AV_RL16(extradata + 16);
if (s->gain_i)
s->gain = pow(10, s->gain_i / (20.0 * 256));
map_type = extradata[18];
if (!map_type) {
if (channels > 2) {
av_log(avctx, AV_LOG_ERROR,
"Channel mapping 0 is only specified for up to 2 channels\n");
return AVERROR_INVALIDDATA;
}
layout = (channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
streams = 1;
stereo_streams = channels - 1;
channel_map = default_channel_map;
} else if (map_type == 1 || map_type == 255) {
if (extradata_size < 21 + channels) {
av_log(avctx, AV_LOG_ERROR, "Invalid extradata size: %d\n",
extradata_size);
return AVERROR_INVALIDDATA;
}
streams = extradata[19];
stereo_streams = extradata[20];
if (!streams || stereo_streams > streams ||
streams + stereo_streams > 255) {
av_log(avctx, AV_LOG_ERROR,
"Invalid stream/stereo stream count: %d/%d\n", streams, stereo_streams);
return AVERROR_INVALIDDATA;
}
if (map_type == 1) {
if (channels > 8) {
av_log(avctx, AV_LOG_ERROR,
"Channel mapping 1 is only specified for up to 8 channels\n");
return AVERROR_INVALIDDATA;
}
layout = ff_vorbis_channel_layouts[channels - 1];
channel_reorder = channel_reorder_vorbis;
} else
layout = 0;
channel_map = extradata + 21;
} else {
avpriv_request_sample(avctx, "Mapping type %d", map_type);
return AVERROR_PATCHWELCOME;
}
s->channel_maps = av_mallocz_array(channels, sizeof(*s->channel_maps));
if (!s->channel_maps)
return AVERROR(ENOMEM);
for (i = 0; i < channels; i++) {
ChannelMap *map = &s->channel_maps[i];
uint8_t idx = channel_map[channel_reorder(channels, i)];
if (idx == 255) {
map->silence = 1;
continue;
} else if (idx >= streams + stereo_streams) {
av_log(avctx, AV_LOG_ERROR,
"Invalid channel map for output channel %d: %d\n", i, idx);
return AVERROR_INVALIDDATA;
}
/* check that we din't see this index yet */
map->copy = 0;
for (j = 0; j < i; j++)
if (channel_map[channel_reorder(channels, j)] == idx) {
map->copy = 1;
map->copy_idx = j;
break;
}
if (idx < 2 * stereo_streams) {
map->stream_idx = idx / 2;
map->channel_idx = idx & 1;
} else {
map->stream_idx = idx - stereo_streams;
map->channel_idx = 0;
}
}
avctx->channels = channels;
avctx->channel_layout = layout;
s->nb_streams = streams;
s->nb_stereo_streams = stereo_streams;
return 0;
}

429
libavcodec/opus.h
Unescape View File

@ -0,0 +1,429 @@
/*
* 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 "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))
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 CeltIMDCTContext CeltIMDCTContext;
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;
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);
/**
* Init an iMDCT of the length 2 * 15 * (2^N)
*/
int ff_celt_imdct_init(CeltIMDCTContext **s, int N);
/**
* Free an iMDCT.
*/
void ff_celt_imdct_uninit(CeltIMDCTContext **s);
/**
* Calculate the middle half of the iMDCT
*/
void ff_celt_imdct_half(CeltIMDCTContext *s, float *dst, const float *src,
int src_stride, float scale);
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 */

2220
libavcodec/opus_celt.c
View File

File diff suppressed because it is too large Load Diff

268
libavcodec/opus_imdct.c
Unescape View File

@ -0,0 +1,268 @@
/*
* 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
*/
/**
* @file
* Celt non-power of 2 iMDCT
*/
#include <float.h>
#include <math.h>
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "fft.h"
#include "opus.h"
// complex c = a * b
#define CMUL3(cre, cim, are, aim, bre, bim) \
do { \
cre = are * bre - aim * bim; \
cim = are * bim + aim * bre; \
} while (0)
#define CMUL(c, a, b) CMUL3((c).re, (c).im, (a).re, (a).im, (b).re, (b).im)
// complex c = a * b
// d = a * conjugate(b)
#define CMUL2(c, d, a, b) \
do { \
float are = (a).re; \
float aim = (a).im; \
float bre = (b).re; \
float bim = (b).im; \
float rr = are * bre; \
float ri = are * bim; \
float ir = aim * bre; \
float ii = aim * bim; \
(c).re = rr - ii; \
(c).im = ri + ir; \
(d).re = rr + ii; \
(d).im = -ri + ir; \
} while (0)
struct CeltIMDCTContext {
int fft_n;
int len2;
int len4;
FFTComplex *tmp;
FFTComplex *twiddle_exptab;
FFTComplex *exptab[6];
};
av_cold void ff_celt_imdct_uninit(CeltIMDCTContext **ps)
{
CeltIMDCTContext *s = *ps;
int i;
if (!s)
return;
for (i = 0; i < FF_ARRAY_ELEMS(s->exptab); i++)
av_freep(&s->exptab[i]);
av_freep(&s->twiddle_exptab);
av_freep(&s->tmp);
av_freep(ps);
}
av_cold int ff_celt_imdct_init(CeltIMDCTContext **ps, int N)
{
CeltIMDCTContext *s;
int len2 = 15 * (1 << N);
int len = 2 * len2;
int i, j;
if (len2 > CELT_MAX_FRAME_SIZE)
return AVERROR(EINVAL);
s = av_mallocz(sizeof(*s));
if (!s)
return AVERROR(ENOMEM);
s->fft_n = N - 1;
s->len4 = len2 / 2;
s->len2 = len2;
s->tmp = av_malloc(len * 2 * sizeof(*s->tmp));
if (!s->tmp)
goto fail;
s->twiddle_exptab = av_malloc(s->len4 * sizeof(*s->twiddle_exptab));
if (!s->twiddle_exptab)
goto fail;
for (i = 0; i < s->len4; i++) {
s->twiddle_exptab[i].re = cos(2 * M_PI * (i + 0.125 + s->len4) / len);
s->twiddle_exptab[i].im = sin(2 * M_PI * (i + 0.125 + s->len4) / len);
}
for (i = 0; i < FF_ARRAY_ELEMS(s->exptab); i++) {
int N = 15 * (1 << i);
s->exptab[i] = av_malloc(sizeof(*s->exptab[i]) * FFMAX(N, 19));
if (!s->exptab[i])
goto fail;
for (j = 0; j < N; j++) {
s->exptab[i][j].re = cos(2 * M_PI * j / N);
s->exptab[i][j].im = sin(2 * M_PI * j / N);
}
}
// wrap around to simplify fft15
for (j = 15; j < 19; j++)
s->exptab[0][j] = s->exptab[0][j - 15];
*ps = s;
return 0;
fail:
ff_celt_imdct_uninit(&s);
return AVERROR(ENOMEM);
}
static void fft5(FFTComplex *out, const FFTComplex *in, int stride)
{
// [0] = exp(2 * i * pi / 5), [1] = exp(2 * i * pi * 2 / 5)
static const FFTComplex fact[] = { { 0.30901699437494745, 0.95105651629515353 },
{ -0.80901699437494734, 0.58778525229247325 } };
FFTComplex z[4][4];
CMUL2(z[0][0], z[0][3], in[1 * stride], fact[0]);
CMUL2(z[0][1], z[0][2], in[1 * stride], fact[1]);
CMUL2(z[1][0], z[1][3], in[2 * stride], fact[0]);
CMUL2(z[1][1], z[1][2], in[2 * stride], fact[1]);
CMUL2(z[2][0], z[2][3], in[3 * stride], fact[0]);
CMUL2(z[2][1], z[2][2], in[3 * stride], fact[1]);
CMUL2(z[3][0], z[3][3], in[4 * stride], fact[0]);
CMUL2(z[3][1], z[3][2], in[4 * stride], fact[1]);
out[0].re = in[0].re + in[stride].re + in[2 * stride].re + in[3 * stride].re + in[4 * stride].re;
out[0].im = in[0].im + in[stride].im + in[2 * stride].im + in[3 * stride].im + in[4 * stride].im;
out[1].re = in[0].re + z[0][0].re + z[1][1].re + z[2][2].re + z[3][3].re;
out[1].im = in[0].im + z[0][0].im + z[1][1].im + z[2][2].im + z[3][3].im;
out[2].re = in[0].re + z[0][1].re + z[1][3].re + z[2][0].re + z[3][2].re;
out[2].im = in[0].im + z[0][1].im + z[1][3].im + z[2][0].im + z[3][2].im;
out[3].re = in[0].re + z[0][2].re + z[1][0].re + z[2][3].re + z[3][1].re;
out[3].im = in[0].im + z[0][2].im + z[1][0].im + z[2][3].im + z[3][1].im;
out[4].re = in[0].re + z[0][3].re + z[1][2].re + z[2][1].re + z[3][0].re;
out[4].im = in[0].im + z[0][3].im + z[1][2].im + z[2][1].im + z[3][0].im;
}
static void fft15(CeltIMDCTContext *s, FFTComplex *out, const FFTComplex *in, int stride)
{
const FFTComplex *exptab = s->exptab[0];
FFTComplex tmp[5];
FFTComplex tmp1[5];
FFTComplex tmp2[5];
int k;
fft5(tmp, in, stride * 3);
fft5(tmp1, in + stride, stride * 3);
fft5(tmp2, in + 2 * stride, stride * 3);
for (k = 0; k < 5; k++) {
FFTComplex t1, t2;
CMUL(t1, tmp1[k], exptab[k]);
CMUL(t2, tmp2[k], exptab[2 * k]);
out[k].re = tmp[k].re + t1.re + t2.re;
out[k].im = tmp[k].im + t1.im + t2.im;
CMUL(t1, tmp1[k], exptab[k + 5]);
CMUL(t2, tmp2[k], exptab[2 * (k + 5)]);
out[k + 5].re = tmp[k].re + t1.re + t2.re;
out[k + 5].im = tmp[k].im + t1.im + t2.im;
CMUL(t1, tmp1[k], exptab[k + 10]);
CMUL(t2, tmp2[k], exptab[2 * k + 5]);
out[k + 10].re = tmp[k].re + t1.re + t2.re;
out[k + 10].im = tmp[k].im + t1.im + t2.im;
}
}
/*
* FFT of the length 15 * (2^N)
*/
static void fft_calc(CeltIMDCTContext *s, FFTComplex *out, const FFTComplex *in, int N, int stride)
{
if (N) {
const FFTComplex *exptab = s->exptab[N];
const int len2 = 15 * (1 << (N - 1));
int k;
fft_calc(s, out, in, N - 1, stride * 2);
fft_calc(s, out + len2, in + stride, N - 1, stride * 2);
for (k = 0; k < len2; k++) {
FFTComplex t;
CMUL(t, out[len2 + k], exptab[k]);
out[len2 + k].re = out[k].re - t.re;
out[len2 + k].im = out[k].im - t.im;
out[k].re += t.re;
out[k].im += t.im;
}
} else
fft15(s, out, in, stride);
}
void ff_celt_imdct_half(CeltIMDCTContext *s, float *dst, const float *src,
int stride, float scale)
{
FFTComplex *z = (FFTComplex *)dst;
const int len8 = s->len4 / 2;
const float *in1 = src;
const float *in2 = src + (s->len2 - 1) * stride;
int i;
for (i = 0; i < s->len4; i++) {
FFTComplex tmp = { *in2, *in1 };
CMUL(s->tmp[i], tmp, s->twiddle_exptab[i]);
in1 += 2 * stride;
in2 -= 2 * stride;
}
fft_calc(s, z, s->tmp, s->fft_n, 1);
for (i = 0; i < len8; i++) {
float r0, i0, r1, i1;
CMUL3(r0, i1, z[len8 - i - 1].im, z[len8 - i - 1].re, s->twiddle_exptab[len8 - i - 1].im, s->twiddle_exptab[len8 - i - 1].re);
CMUL3(r1, i0, z[len8 + i].im, z[len8 + i].re, s->twiddle_exptab[len8 + i].im, s->twiddle_exptab[len8 + i].re);
z[len8 - i - 1].re = scale * r0;
z[len8 - i - 1].im = scale * i0;
z[len8 + i].re = scale * r1;
z[len8 + i].im = scale * i1;
}
}

75
libavcodec/opus_parser.c
Unescape View File

@ -0,0 +1,75 @@
/*
* 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
*/
/**
* @file
* Opus parser
*
* Determines the duration for each packet.
*/
#include "avcodec.h"
#include "opus.h"
typedef struct OpusParseContext {
OpusContext ctx;
OpusPacket pkt;
int extradata_parsed;
} OpusParseContext;
static int opus_parse(AVCodecParserContext *ctx, AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size)
{
OpusParseContext *s = ctx->priv_data;
int ret;
if (!buf_size)
return 0;
if (avctx->extradata && !s->extradata_parsed) {
ret = ff_opus_parse_extradata(avctx, &s->ctx);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing Ogg extradata.\n");
goto fail;
}
av_freep(&s->ctx.channel_maps);
s->extradata_parsed = 1;
}
ret = ff_opus_parse_packet(&s->pkt, buf, buf_size, s->ctx.nb_streams > 1);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing Opus packet header.\n");
goto fail;
}
ctx->duration = s->pkt.frame_count * s->pkt.frame_duration;
fail:
*poutbuf = buf;
*poutbuf_size = buf_size;
return buf_size;
}
AVCodecParser ff_opus_parser = {
.codec_ids = { AV_CODEC_ID_OPUS },
.priv_data_size = sizeof(OpusParseContext),
.parser_parse = opus_parse,
};

1597
libavcodec/opus_silk.c
View File

File diff suppressed because it is too large Load Diff

674
libavcodec/opusdec.c
Unescape View File

@ -0,0 +1,674 @@
/*
* Opus decoder
* 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
*/
/**
* @file
* Opus decoder
* @author Andrew D'Addesio, Anton Khirnov
*
* Codec homepage: http://opus-codec.org/
* Specification: http://tools.ietf.org/html/rfc6716
* Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
*
* Ogg-contained .opus files can be produced with opus-tools:
* http://git.xiph.org/?p=opus-tools.git
*/
#include <stdint.h>
#include "libavutil/attributes.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavresample/avresample.h"
#include "avcodec.h"
#include "celp_filters.h"
#include "fft.h"
#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
#include "opus.h"
static const uint16_t silk_frame_duration_ms[16] = {
10, 20, 40, 60,
10, 20, 40, 60,
10, 20, 40, 60,
10, 20,
10, 20,
};
/* number of samples of silence to feed to the resampler
* at the beginning */
static const int silk_resample_delay[] = {
4, 8, 11, 11, 11
};
static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 };
static int get_silk_samplerate(int config)
{
if (config < 4)
return 8000;
else if (config < 8)
return 12000;
return 16000;
}
/**
* Range decoder
*/
static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)
{
int ret = init_get_bits8(&rc->gb, data, size);
if (ret < 0)
return ret;
rc->range = 128;
rc->value = 127 - get_bits(&rc->gb, 7);
rc->total_read_bits = 9;
opus_rc_normalize(rc);
return 0;
}
static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend,
unsigned int bytes)
{
rc->rb.position = rightend;
rc->rb.bytes = bytes;
rc->rb.cachelen = 0;
rc->rb.cacheval = 0;
}
static void opus_fade(float *out,
const float *in1, const float *in2,
const float *window, int len)
{
int i;
for (i = 0; i < len; i++)
out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
}
static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
{
int celt_size = av_audio_fifo_size(s->celt_delay);
int ret, i;
ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
NULL, 0, 0);
if (ret < 0)
return ret;
else if (ret != nb_samples) {
av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
ret);
return AVERROR_BUG;
}
if (celt_size) {
if (celt_size != nb_samples) {
av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
return AVERROR_BUG;
}
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(s->out[i],
s->celt_output[i], 1.0,
nb_samples);
}
}
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
opus_fade(s->out[i], s->out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
s->out[0] += nb_samples;
s->out[1] += nb_samples;
s->out_size -= nb_samples * sizeof(float);
return 0;
}
static int opus_init_resample(OpusStreamContext *s)
{
float delay[16] = { 0.0 };
uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
int ret;
av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
ret = avresample_open(s->avr);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
return ret;
}
ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
silk_resample_delay[s->packet.bandwidth]);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR,
"Error feeding initial silence to the resampler.\n");
return ret;
}
return 0;
}
static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
{
int ret;
enum OpusBandwidth bw = s->packet.bandwidth;
if (s->packet.mode == OPUS_MODE_SILK &&
bw == OPUS_BANDWIDTH_MEDIUMBAND)
bw = OPUS_BANDWIDTH_WIDEBAND;
ret = opus_rc_init(&s->redundancy_rc, data, size);
if (ret < 0)
goto fail;
opus_raw_init(&s->redundancy_rc, data + size, size);
ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
s->redundancy_output,
s->packet.stereo + 1, 240,
0, celt_band_end[s->packet.bandwidth]);
if (ret < 0)
goto fail;
return 0;
fail:
av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
return ret;
}
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
{
int samples = s->packet.frame_duration;
int redundancy = 0;
int redundancy_size, redundancy_pos;
int ret, i, consumed;
int delayed_samples = s->delayed_samples;
ret = opus_rc_init(&s->rc, data, size);
if (ret < 0)
return ret;
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
if (!avresample_is_open(s->avr)) {
ret = opus_init_resample(s);
if (ret < 0)
return ret;
}
samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
s->packet.stereo + 1,
silk_frame_duration_ms[s->packet.config]);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
s->packet.frame_duration,
(uint8_t**)s->silk_output,
sizeof(s->silk_buf[0]),
samples);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
s->delayed_samples += s->packet.frame_duration - samples;
} else
ff_silk_flush(s->silk);
// decode redundancy information
consumed = opus_rc_tell(&s->rc);
if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
redundancy = opus_rc_p2model(&s->rc, 12);
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
redundancy = 1;
if (redundancy) {
redundancy_pos = opus_rc_p2model(&s->rc, 1);
if (s->packet.mode == OPUS_MODE_HYBRID)
redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
else
redundancy_size = size - (consumed + 7) / 8;
size -= redundancy_size;
if (size < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
return AVERROR_INVALIDDATA;
}
if (redundancy_pos) {
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
ff_celt_flush(s->celt);
}
}
/* decode the CELT frame */
if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
float *out_tmp[2] = { s->out[0], s->out[1] };
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
out_tmp : s->celt_output;
int celt_output_samples = samples;
int delay_samples = av_audio_fifo_size(s->celt_delay);
if (delay_samples) {
if (s->packet.mode == OPUS_MODE_HYBRID) {
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
delay_samples);
out_tmp[i] += delay_samples;
}
celt_output_samples -= delay_samples;
} else {
av_log(s->avctx, AV_LOG_WARNING,
"Spurious CELT delay samples present.\n");
av_audio_fifo_drain(s->celt_delay, delay_samples);
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_BUG;
}
}
opus_raw_init(&s->rc, data + size, size);
ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
s->packet.stereo + 1,
s->packet.frame_duration,
(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
celt_band_end[s->packet.bandwidth]);
if (ret < 0)
return ret;
if (s->packet.mode == OPUS_MODE_HYBRID) {
int celt_delay = s->packet.frame_duration - celt_output_samples;
void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
s->celt_output[1] + celt_output_samples };
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i],
s->celt_output[i], 1.0,
celt_output_samples);
}
ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
if (ret < 0)
return ret;
}
} else
ff_celt_flush(s->celt);
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
opus_fade(s->out[i], s->out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
if (redundancy) {
if (!redundancy_pos) {
ff_celt_flush(s->celt);
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
for (i = 0; i < s->output_channels; i++) {
opus_fade(s->out[i] + samples - 120 + delayed_samples,
s->out[i] + samples - 120 + delayed_samples,
s->redundancy_output[i] + 120,
ff_celt_window2, 120 - delayed_samples);
if (delayed_samples)
s->redundancy_idx = 120 - delayed_samples;
}
} else {
for (i = 0; i < s->output_channels; i++) {
memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
opus_fade(s->out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
s->out[i] + 120 + delayed_samples,
ff_celt_window2, 120);
}
}
}
return samples;
}
static int opus_decode_subpacket(OpusStreamContext *s,
const uint8_t *buf, int buf_size,
int nb_samples)
{
int output_samples = 0;
int flush_needed = 0;
int i, j, ret;
/* check if we need to flush the resampler */
if (avresample_is_open(s->avr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
}
}
if (!buf && !flush_needed)
return 0;
/* use dummy output buffers if the channel is not mapped to anything */
if (!s->out[0] ||
(s->output_channels == 2 && !s->out[1])) {
av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
if (!s->out_dummy)
return AVERROR(ENOMEM);
if (!s->out[0])
s->out[0] = s->out_dummy;
if (!s->out[1])
s->out[1] = s->out_dummy;
}
/* flush the resampler if necessary */
if (flush_needed) {
ret = opus_flush_resample(s, s->delayed_samples);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
return ret;
}
avresample_close(s->avr);
output_samples += s->delayed_samples;
s->delayed_samples = 0;
if (!buf)
goto finish;
}
/* decode all the frames in the packet */
for (i = 0; i < s->packet.frame_count; i++) {
int size = s->packet.frame_size[i];
int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return samples;
for (j = 0; j < s->output_channels; j++)
memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
samples = s->packet.frame_duration;
}
output_samples += samples;
for (j = 0; j < s->output_channels; j++)
s->out[j] += samples;
s->out_size -= samples * sizeof(float);
}
finish:
s->out[0] = s->out[1] = NULL;
s->out_size = 0;
return output_samples;
}
static int opus_decode_packet(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
OpusContext *c = avctx->priv_data;
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int coded_samples = 0;
int decoded_samples = 0;
int i, ret;
/* decode the header of the first sub-packet to find out the sample count */
if (buf) {
OpusPacket *pkt = &c->streams[0].packet;
ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
return ret;
}
coded_samples += pkt->frame_count * pkt->frame_duration;
c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
}
frame->nb_samples = coded_samples + c->streams[0].delayed_samples;
/* no input or buffered data => nothing to do */
if (!frame->nb_samples) {
*got_frame_ptr = 0;
return 0;
}
/* setup the data buffers */
ret = ff_get_buffer(avctx, frame, 0);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
frame->nb_samples = 0;
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
if (!map->copy)
c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
}
for (i = 0; i < c->nb_streams; i++)
c->streams[i].out_size = frame->linesize[0];
/* decode each sub-packet */
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
if (i && buf) {
ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
return ret;
}
s->silk_samplerate = get_silk_samplerate(s->packet.config);
}
ret = opus_decode_subpacket(&c->streams[i], buf,
s->packet.data_size, coded_samples);
if (ret < 0)
return ret;
if (decoded_samples && ret != decoded_samples) {
av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "
"in a multi-channel stream\n");
return AVERROR_INVALIDDATA;
}
decoded_samples = ret;
buf += s->packet.packet_size;
buf_size -= s->packet.packet_size;
}
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
/* handle copied channels */
if (map->copy) {
memcpy(frame->extended_data[i],
frame->extended_data[map->copy_idx],
frame->linesize[0]);
} else if (map->silence) {
memset(frame->extended_data[i], 0, frame->linesize[0]);
}
if (c->gain_i) {
c->fdsp.vector_fmul_scalar((float*)frame->extended_data[i],
(float*)frame->extended_data[i],
c->gain, FFALIGN(decoded_samples, 8));
}
}
frame->nb_samples = decoded_samples;
*got_frame_ptr = !!decoded_samples;
return avpkt->size;
}
static av_cold void opus_decode_flush(AVCodecContext *ctx)
{
OpusContext *c = ctx->priv_data;
int i;
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
memset(&s->packet, 0, sizeof(s->packet));
s->delayed_samples = 0;
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
avresample_close(s->avr);
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
static av_cold int opus_decode_close(AVCodecContext *avctx)
{
OpusContext *c = avctx->priv_data;
int i;
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
ff_silk_free(&s->silk);
ff_celt_free(&s->celt);
av_freep(&s->out_dummy);
s->out_dummy_allocated_size = 0;
av_audio_fifo_free(s->celt_delay);
avresample_free(&s->avr);
}
av_freep(&c->streams);
c->nb_streams = 0;
av_freep(&c->channel_maps);
return 0;
}
static av_cold int opus_decode_init(AVCodecContext *avctx)
{
OpusContext *c = avctx->priv_data;
int ret, i, j;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avctx->sample_rate = 48000;
avpriv_float_dsp_init(&c->fdsp, 0);
/* find out the channel configuration */
ret = ff_opus_parse_extradata(avctx, c);
if (ret < 0)
return ret;
/* allocate and init each independent decoder */
c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
if (!c->streams) {
c->nb_streams = 0;
ret = AVERROR(ENOMEM);
goto fail;
}
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
uint64_t layout;
s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
s->avctx = avctx;
for (j = 0; j < s->output_channels; j++) {
s->silk_output[j] = s->silk_buf[j];
s->celt_output[j] = s->celt_buf[j];
s->redundancy_output[j] = s->redundancy_buf[j];
}
s->fdsp = &c->fdsp;
s->avr = avresample_alloc_context();
if (!s->avr)
goto fail;
layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->avr, "in_channel_layout", layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", layout, 0);
av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0);
ret = ff_silk_init(avctx, &s->silk, s->output_channels);
if (ret < 0)
goto fail;
ret = ff_celt_init(avctx, &s->celt, s->output_channels);
if (ret < 0)
goto fail;
s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
s->output_channels, 1024);
if (!s->celt_delay) {
ret = AVERROR(ENOMEM);
goto fail;
}
}
return 0;
fail:
opus_decode_close(avctx);
return ret;
}
AVCodec ff_opus_decoder = {
.name = "opus",
.long_name = NULL_IF_CONFIG_SMALL("Opus"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_OPUS,
.priv_data_size = sizeof(OpusContext),
.init = opus_decode_init,
.close = opus_decode_close,
.decode = opus_decode_packet,
.flush = opus_decode_flush,
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY,
};

4
libavcodec/version.h
Unescape View File

@ -29,8 +29,8 @@
#include "libavutil/version.h"
#define LIBAVCODEC_VERSION_MAJOR 55
#define LIBAVCODEC_VERSION_MINOR 61
#define LIBAVCODEC_VERSION_MICRO 101
#define LIBAVCODEC_VERSION_MINOR 62
#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \

1
tests/Makefile
Unescape View File

@ -119,6 +119,7 @@ include $(SRC_PATH)/tests/fate/microsoft.mak
include $(SRC_PATH)/tests/fate/monkeysaudio.mak
include $(SRC_PATH)/tests/fate/mp3.mak
include $(SRC_PATH)/tests/fate/mpc.mak
include $(SRC_PATH)/tests/fate/opus.mak
include $(SRC_PATH)/tests/fate/pcm.mak
include $(SRC_PATH)/tests/fate/probe.mak
include $(SRC_PATH)/tests/fate/prores.mak

39
tests/fate/opus.mak
Unescape View File

@ -0,0 +1,39 @@
# The samples were produced by simply rewrapping the official test vectors from
# their custom format into Matroska.
# The reference files were created with our decoder and tested against the
# libopus output with the official opus_compare tool. We cannot use libopus
# output as reference directly, because the use of different resamplers would
# require too high fuzz values, which can hide bugs.
# Before adding new tests here, always make sure they pass opus_compare.
OPUS_CELT_SAMPLES = $(addprefix testvector, 01 07 11) tron.6ch.tinypkts
OPUS_HYBRID_SAMPLES = $(addprefix testvector, 05 06)
OPUS_SILK_SAMPLES = $(addprefix testvector, 02 03 04)
OPUS_SAMPLES = $(addprefix testvector, 08 09 10 12)
define FATE_OPUS_TEST
FATE_OPUS += fate-opus-$(1)
FATE_OPUS$(2) += fate-opus-$(1)
fate-opus-$(1): CMD = ffmpeg -i $(TARGET_SAMPLES)/opus/$(1).mka -f f32le -
fate-opus-$(1): REF = $(TARGET_SAMPLES)/opus/$(1).f32
endef
$(foreach N,$(OPUS_CELT_SAMPLES), $(eval $(call FATE_OPUS_TEST,$(N),_CELT)))
$(foreach N,$(OPUS_HYBRID_SAMPLES),$(eval $(call FATE_OPUS_TEST,$(N),_HYBRID)))
$(foreach N,$(OPUS_SILK_SAMPLES), $(eval $(call FATE_OPUS_TEST,$(N),_SILK)))
$(foreach N,$(OPUS_SAMPLES), $(eval $(call FATE_OPUS_TEST,$(N),)))
FATE_OPUS := $(sort $(FATE_OPUS))
$(FATE_OPUS): CMP = stddev
$(FATE_OPUS): CMP_UNIT = f32
$(FATE_OPUS): FUZZ = 3
$(FATE_OPUS_CELT): CMP = oneoff
$(FATE_OPUS_CELT): FUZZ = 5
#FATE_SAMPLES_AVCONV-$(call DEMDEC, MATROSKA, OPUS) += $(FATE_OPUS)
fate-opus-celt: $(FATE_OPUS_CELT)
fate-opus-hybrid: $(FATE_OPUS_HYBRID)
fate-opus-silk: $(FATE_OPUS_SILK)
fate-opus: $(FATE_OPUS)
Write Preview
Loading…
Cancel
Save