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FFmpeg/libavcodec/opus/dec.c
Anton Khirnov 3f9ca51015 lavc/opus*: move to opus/ subdir
2024-09-02 11:56:53 +02:00

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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/ffmath.h"
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"
#include "libavutil/mem.h"
#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "decode.h"
#include "opus.h"
#include "tab.h"
#include "celt.h"
#include "parse.h"
#include "rc.h"
#include "silk.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
};
typedef struct OpusStreamContext {
AVCodecContext *avctx;
int output_channels;
/* number of decoded samples for this stream */
int decoded_samples;
/* current output buffers for this stream */
float *out[2];
int out_size;
/* Buffer with samples from this stream for synchronizing
* the streams when they have different resampling delays */
AVAudioFifo *sync_buffer;
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];
DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
float *redundancy_output[2];
/* buffers for the next samples to be decoded */
float *cur_out[2];
int remaining_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;
typedef struct OpusContext {
AVClass *av_class;
struct OpusStreamContext *streams;
int apply_phase_inv;
AVFloatDSPContext *fdsp;
float gain;
OpusParseContext p;
} OpusContext;
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->cur_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->cur_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->cur_out[i], s->cur_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->cur_out[0] += nb_samples;
s->cur_out[1] += nb_samples;
s->remaining_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 = 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->cur_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->cur_out[0], s->cur_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->cur_out[i], s->cur_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->cur_out[i] + samples - 120 + delayed_samples,
s->cur_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->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
opus_fade(s->cur_out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
s->cur_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;
s->cur_out[0] = s->out[0];
s->cur_out[1] = s->out[1];
s->remaining_out_size = s->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->cur_out[0] ||
(s->output_channels == 2 && !s->cur_out[1])) {
av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size,
s->remaining_out_size);
if (!s->out_dummy)
return AVERROR(ENOMEM);
if (!s->cur_out[0])
s->cur_out[0] = s->out_dummy;
if (!s->cur_out[1])
s->cur_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->cur_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->cur_out[j] += samples;
s->remaining_out_size -= samples * sizeof(float);
}
finish:
s->cur_out[0] = s->cur_out[1] = NULL;
s->remaining_out_size = 0;
return output_samples;
}
static int opus_decode_packet(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
OpusContext *c = avctx->priv_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 (int i = 0; i < c->p.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(s->sync_buffer));
}
/* 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->p.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;
for (i = 0; i < avctx->ch_layout.nb_channels; i++) {
ChannelMap *map = &c->p.channel_maps[i];
if (!map->copy)
c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
}
/* read the data from the sync buffers */
for (int i = 0; i < c->p.nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
float **out = s->out;
int sync_size = av_audio_fifo_size(s->sync_buffer);
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(s->sync_buffer, (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;
s->out_size = frame->linesize[0] - ret * sizeof(float);
}
/* decode each sub-packet */
for (int i = 0; i < c->p.nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
if (i && buf) {
ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->p.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,
coded_samples);
if (ret < 0)
return ret;
s->decoded_samples = ret;
decoded_samples = FFMIN(decoded_samples, ret);
buf += s->packet.packet_size;
buf_size -= s->packet.packet_size;
}
/* buffer the extra samples */
for (int i = 0; i < c->p.nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
int buffer_samples = s->decoded_samples - decoded_samples;
if (buffer_samples) {
float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0],
s->out[1] ? s->out[1] : (float*)frame->extended_data[0] };
buf[0] += decoded_samples;
buf[1] += decoded_samples;
ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples);
if (ret < 0)
return ret;
}
}
for (i = 0; i < avctx->ch_layout.nb_channels; i++) {
ChannelMap *map = &c->p.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->p.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;
for (int i = 0; i < c->p.nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
memset(&s->packet, 0, sizeof(s->packet));
s->delayed_samples = 0;
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
swr_close(s->swr);
av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer));
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
}
}
static av_cold int opus_decode_close(AVCodecContext *avctx)
{
OpusContext *c = avctx->priv_data;
for (int i = 0; i < c->p.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->sync_buffer);
av_audio_fifo_free(s->celt_delay);
swr_free(&s->swr);
}
av_freep(&c->streams);
c->p.nb_streams = 0;
av_freep(&c->p.channel_maps);
av_freep(&c->fdsp);
return 0;
}
static av_cold int opus_decode_init(AVCodecContext *avctx)
{
OpusContext *c = avctx->priv_data;
int ret;
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->p);
if (ret < 0)
return ret;
if (c->p.gain_i)
c->gain = ff_exp10(c->p.gain_i / (20.0 * 256));
/* allocate and init each independent decoder */
c->streams = av_calloc(c->p.nb_streams, sizeof(*c->streams));
if (!c->streams) {
c->p.nb_streams = 0;
return AVERROR(ENOMEM);
}
for (int i = 0; i < c->p.nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
AVChannelLayout layout;
s->output_channels = (i < c->p.nb_stereo_streams) ? 2 : 1;
s->avctx = avctx;
for (int 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)
return AVERROR(ENOMEM);
layout = (s->output_channels == 1) ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
(AVChannelLayout)AV_CHANNEL_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_chlayout(s->swr, "in_chlayout", &layout, 0);
av_opt_set_chlayout(s->swr, "out_chlayout", &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)
return ret;
ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);
if (ret < 0)
return ret;
s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
s->output_channels, 1024);
if (!s->celt_delay)
return AVERROR(ENOMEM);
s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt,
s->output_channels, 32);
if (!s->sync_buffer)
return AVERROR(ENOMEM);
}
return 0;
}
#define OFFSET(x) offsetof(OpusContext, x)
#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
static const AVOption opus_options[] = {
{ "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },
{ NULL },
};
static const AVClass opus_class = {
.class_name = "Opus Decoder",
.item_name = av_default_item_name,
.option = opus_options,
.version = LIBAVUTIL_VERSION_INT,
};
const FFCodec ff_opus_decoder = {
.p.name = "opus",
CODEC_LONG_NAME("Opus"),
.p.priv_class = &opus_class,
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_OPUS,
.priv_data_size = sizeof(OpusContext),
.init = opus_decode_init,
.close = opus_decode_close,
FF_CODEC_DECODE_CB(opus_decode_packet),
.flush = opus_decode_flush,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};
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