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FFmpeg/libavcodec/opusdec.c
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

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/*
* 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 "libswresample/swresample.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#include "mathops.h"
#include "opus.h"
#include "opustab.h"
#include "opus_celt.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 int get_silk_samplerate(int config)
{
if (config < 4)
return 8000;
else if (config < 8)
return 12000;
return 16000;
}
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 = swr_convert(s->swr,
(uint8_t**)s->out, nb_samples,
NULL, 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)
{
static const float delay[16] = { 0.0 };
const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
int ret;
av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
ret = swr_init(s->swr);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
return ret;
}
ret = swr_convert(s->swr,
NULL, 0,
delayptr, 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 = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);
if (ret < 0)
goto fail;
ff_opus_rc_dec_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, ff_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 = ff_opus_rc_dec_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 (!swr_is_initialized(s->swr)) {
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 = swr_convert(s->swr,
(uint8_t**)s->out, s->packet.frame_duration,
(const uint8_t**)s->silk_output, samples);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
av_assert2((samples & 7) == 0);
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 = ff_opus_rc_dec_log(&s->rc, 12);
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
redundancy = 1;
if (redundancy) {
redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
if (s->packet.mode == OPUS_MODE_HYBRID)
redundancy_size = ff_opus_rc_dec_uint(&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;
}
}
ff_opus_rc_dec_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,
ff_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,
float **out, int out_size,
int nb_samples)
{
int output_samples = 0;
int flush_needed = 0;
int i, j, ret;
s->out[0] = out[0];
s->out[1] = out[1];
s->out_size = out_size;
/* check if we need to flush the resampler */
if (swr_is_initialized(s->swr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->swr, "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;
}
swr_close(s->swr);
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 = INT_MAX;
int delayed_samples = 0;
int i, ret;
/* calculate the number of delayed samples */
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
s->out[0] =
s->out[1] = NULL;
delayed_samples = FFMAX(delayed_samples,
s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
}
/* 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 + 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)
return ret;
frame->nb_samples = 0;
memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
if (!map->copy)
c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
}
/* read the data from the sync buffers */
for (i = 0; i < c->nb_streams; i++) {
float **out = c->out + 2 * i;
int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
float sync_dummy[32];
int out_dummy = (!out[0]) | ((!out[1]) << 1);
if (!out[0])
out[0] = sync_dummy;
if (!out[1])
out[1] = sync_dummy;
if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
return AVERROR_BUG;
ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
if (ret < 0)
return ret;
if (out_dummy & 1)
out[0] = NULL;
else
out[0] += ret;
if (out_dummy & 2)
out[1] = NULL;
else
out[1] += ret;
c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
}
/* 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;
}
if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
av_log(avctx, AV_LOG_ERROR,
"Mismatching coded sample count in substream %d.\n", i);
return AVERROR_INVALIDDATA;
}
s->silk_samplerate = get_silk_samplerate(s->packet.config);
}
ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
c->out + 2 * i, c->out_size[i], coded_samples);
if (ret < 0)
return ret;
c->decoded_samples[i] = ret;
decoded_samples = FFMIN(decoded_samples, ret);
buf += s->packet.packet_size;
buf_size -= s->packet.packet_size;
}
/* buffer the extra samples */
for (i = 0; i < c->nb_streams; i++) {
int buffer_samples = c->decoded_samples[i] - decoded_samples;
if (buffer_samples) {
float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
buf[0] += decoded_samples;
buf[1] += decoded_samples;
ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
if (ret < 0)
return ret;
}
}
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 && decoded_samples > 0) {
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));
swr_close(s->swr);
av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
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);
swr_free(&s->swr);
}
av_freep(&c->streams);
if (c->sync_buffers) {
for (i = 0; i < c->nb_streams; i++)
av_audio_fifo_free(c->sync_buffers[i]);
}
av_freep(&c->sync_buffers);
av_freep(&c->decoded_samples);
av_freep(&c->out);
av_freep(&c->out_size);
c->nb_streams = 0;
av_freep(&c->channel_maps);
av_freep(&c->fdsp);
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;
c->fdsp = avpriv_float_dsp_alloc(0);
if (!c->fdsp)
return AVERROR(ENOMEM);
/* find out the channel configuration */
ret = ff_opus_parse_extradata(avctx, c);
if (ret < 0) {
av_freep(&c->fdsp);
return ret;
}
/* allocate and init each independent decoder */
c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));
c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));
c->sync_buffers = av_mallocz_array(c->nb_streams, sizeof(*c->sync_buffers));
c->decoded_samples = av_mallocz_array(c->nb_streams, sizeof(*c->decoded_samples));
if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {
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->swr =swr_alloc();
if (!s->swr)
goto fail;
layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
av_opt_set_int(s->swr, "filter_size", 16, 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;
}
c->sync_buffers[i] = av_audio_fifo_alloc(avctx->sample_fmt,
s->output_channels, 32);
if (!c->sync_buffers[i]) {
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 = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
};
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