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742 lines
24 KiB
C
Executable File
742 lines
24 KiB
C
Executable File
/*
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* Opus decoder
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* Copyright (c) 2012 Andrew D'Addesio
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* Copyright (c) 2013-2014 Mozilla Corporation
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Opus decoder
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* @author Andrew D'Addesio, Anton Khirnov
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*
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* Codec homepage: http://opus-codec.org/
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* Specification: http://tools.ietf.org/html/rfc6716
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* Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
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*
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* Ogg-contained .opus files can be produced with opus-tools:
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* http://git.xiph.org/?p=opus-tools.git
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*/
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#include <stdint.h>
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#include "libavutil/attributes.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "libswresample/swresample.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "internal.h"
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#include "mathops.h"
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#include "opus.h"
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#include "opustab.h"
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#include "opus_celt.h"
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static const uint16_t silk_frame_duration_ms[16] = {
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10, 20, 40, 60,
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10, 20, 40, 60,
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10, 20, 40, 60,
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10, 20,
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10, 20,
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};
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/* number of samples of silence to feed to the resampler
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* at the beginning */
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static const int silk_resample_delay[] = {
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4, 8, 11, 11, 11
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};
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static int get_silk_samplerate(int config)
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{
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if (config < 4)
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return 8000;
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else if (config < 8)
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return 12000;
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return 16000;
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}
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static void opus_fade(float *out,
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const float *in1, const float *in2,
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const float *window, int len)
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{
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int i;
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for (i = 0; i < len; i++)
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out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
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}
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static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
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{
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int celt_size = av_audio_fifo_size(s->celt_delay);
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int ret, i;
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ret = swr_convert(s->swr,
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(uint8_t**)s->out, nb_samples,
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NULL, 0);
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if (ret < 0)
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return ret;
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else if (ret != nb_samples) {
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av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
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ret);
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return AVERROR_BUG;
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}
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if (celt_size) {
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if (celt_size != nb_samples) {
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av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
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return AVERROR_BUG;
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}
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av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
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for (i = 0; i < s->output_channels; i++) {
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s->fdsp->vector_fmac_scalar(s->out[i],
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s->celt_output[i], 1.0,
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nb_samples);
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}
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}
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if (s->redundancy_idx) {
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for (i = 0; i < s->output_channels; i++)
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opus_fade(s->out[i], s->out[i],
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s->redundancy_output[i] + 120 + s->redundancy_idx,
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ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
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s->redundancy_idx = 0;
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}
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s->out[0] += nb_samples;
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s->out[1] += nb_samples;
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s->out_size -= nb_samples * sizeof(float);
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return 0;
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}
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static int opus_init_resample(OpusStreamContext *s)
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{
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static const float delay[16] = { 0.0 };
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const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
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int ret;
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av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
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ret = swr_init(s->swr);
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if (ret < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
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return ret;
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}
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ret = swr_convert(s->swr,
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NULL, 0,
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delayptr, silk_resample_delay[s->packet.bandwidth]);
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if (ret < 0) {
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av_log(s->avctx, AV_LOG_ERROR,
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"Error feeding initial silence to the resampler.\n");
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return ret;
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}
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return 0;
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}
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static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
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{
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int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);
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if (ret < 0)
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goto fail;
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ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size);
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ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
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s->redundancy_output,
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s->packet.stereo + 1, 240,
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0, ff_celt_band_end[s->packet.bandwidth]);
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if (ret < 0)
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goto fail;
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return 0;
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fail:
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av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
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return ret;
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}
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static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
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{
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int samples = s->packet.frame_duration;
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int redundancy = 0;
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int redundancy_size, redundancy_pos;
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int ret, i, consumed;
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int delayed_samples = s->delayed_samples;
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ret = ff_opus_rc_dec_init(&s->rc, data, size);
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if (ret < 0)
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return ret;
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/* decode the silk frame */
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if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
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if (!swr_is_initialized(s->swr)) {
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ret = opus_init_resample(s);
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if (ret < 0)
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return ret;
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}
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samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
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FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
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s->packet.stereo + 1,
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silk_frame_duration_ms[s->packet.config]);
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if (samples < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
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return samples;
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}
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samples = swr_convert(s->swr,
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(uint8_t**)s->out, s->packet.frame_duration,
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(const uint8_t**)s->silk_output, samples);
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if (samples < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
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return samples;
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}
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av_assert2((samples & 7) == 0);
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s->delayed_samples += s->packet.frame_duration - samples;
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} else
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ff_silk_flush(s->silk);
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// decode redundancy information
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consumed = opus_rc_tell(&s->rc);
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if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
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redundancy = ff_opus_rc_dec_log(&s->rc, 12);
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else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
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redundancy = 1;
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if (redundancy) {
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redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
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if (s->packet.mode == OPUS_MODE_HYBRID)
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redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
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else
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redundancy_size = size - (consumed + 7) / 8;
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size -= redundancy_size;
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if (size < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
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return AVERROR_INVALIDDATA;
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}
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if (redundancy_pos) {
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ret = opus_decode_redundancy(s, data + size, redundancy_size);
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if (ret < 0)
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return ret;
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ff_celt_flush(s->celt);
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}
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}
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/* decode the CELT frame */
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if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
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float *out_tmp[2] = { s->out[0], s->out[1] };
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float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
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out_tmp : s->celt_output;
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int celt_output_samples = samples;
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int delay_samples = av_audio_fifo_size(s->celt_delay);
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if (delay_samples) {
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if (s->packet.mode == OPUS_MODE_HYBRID) {
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av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
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for (i = 0; i < s->output_channels; i++) {
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s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
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delay_samples);
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out_tmp[i] += delay_samples;
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}
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celt_output_samples -= delay_samples;
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} else {
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av_log(s->avctx, AV_LOG_WARNING,
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"Spurious CELT delay samples present.\n");
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av_audio_fifo_drain(s->celt_delay, delay_samples);
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if (s->avctx->err_recognition & AV_EF_EXPLODE)
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return AVERROR_BUG;
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}
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}
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ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
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ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
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s->packet.stereo + 1,
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s->packet.frame_duration,
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(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
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ff_celt_band_end[s->packet.bandwidth]);
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if (ret < 0)
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return ret;
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if (s->packet.mode == OPUS_MODE_HYBRID) {
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int celt_delay = s->packet.frame_duration - celt_output_samples;
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void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
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s->celt_output[1] + celt_output_samples };
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for (i = 0; i < s->output_channels; i++) {
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s->fdsp->vector_fmac_scalar(out_tmp[i],
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s->celt_output[i], 1.0,
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celt_output_samples);
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}
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ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
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if (ret < 0)
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return ret;
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}
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} else
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ff_celt_flush(s->celt);
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if (s->redundancy_idx) {
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for (i = 0; i < s->output_channels; i++)
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opus_fade(s->out[i], s->out[i],
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s->redundancy_output[i] + 120 + s->redundancy_idx,
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ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
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s->redundancy_idx = 0;
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}
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if (redundancy) {
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if (!redundancy_pos) {
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ff_celt_flush(s->celt);
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ret = opus_decode_redundancy(s, data + size, redundancy_size);
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if (ret < 0)
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return ret;
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for (i = 0; i < s->output_channels; i++) {
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opus_fade(s->out[i] + samples - 120 + delayed_samples,
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s->out[i] + samples - 120 + delayed_samples,
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s->redundancy_output[i] + 120,
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ff_celt_window2, 120 - delayed_samples);
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if (delayed_samples)
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s->redundancy_idx = 120 - delayed_samples;
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}
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} else {
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for (i = 0; i < s->output_channels; i++) {
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memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
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opus_fade(s->out[i] + 120 + delayed_samples,
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s->redundancy_output[i] + 120,
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s->out[i] + 120 + delayed_samples,
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ff_celt_window2, 120);
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}
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}
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}
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return samples;
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}
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static int opus_decode_subpacket(OpusStreamContext *s,
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const uint8_t *buf, int buf_size,
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float **out, int out_size,
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int nb_samples)
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{
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int output_samples = 0;
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int flush_needed = 0;
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int i, j, ret;
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s->out[0] = out[0];
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s->out[1] = out[1];
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s->out_size = out_size;
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/* check if we need to flush the resampler */
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if (swr_is_initialized(s->swr)) {
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if (buf) {
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int64_t cur_samplerate;
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av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
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flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
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} else {
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flush_needed = !!s->delayed_samples;
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}
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}
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if (!buf && !flush_needed)
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return 0;
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/* use dummy output buffers if the channel is not mapped to anything */
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if (!s->out[0] ||
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(s->output_channels == 2 && !s->out[1])) {
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av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
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if (!s->out_dummy)
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return AVERROR(ENOMEM);
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if (!s->out[0])
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s->out[0] = s->out_dummy;
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if (!s->out[1])
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s->out[1] = s->out_dummy;
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}
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/* flush the resampler if necessary */
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if (flush_needed) {
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ret = opus_flush_resample(s, s->delayed_samples);
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if (ret < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
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return ret;
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}
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swr_close(s->swr);
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output_samples += s->delayed_samples;
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s->delayed_samples = 0;
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if (!buf)
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goto finish;
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}
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|
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/* decode all the frames in the packet */
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for (i = 0; i < s->packet.frame_count; i++) {
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int size = s->packet.frame_size[i];
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int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
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|
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if (samples < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
|
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if (s->avctx->err_recognition & AV_EF_EXPLODE)
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return samples;
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for (j = 0; j < s->output_channels; j++)
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memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
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samples = s->packet.frame_duration;
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}
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output_samples += samples;
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|
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for (j = 0; j < s->output_channels; j++)
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s->out[j] += samples;
|
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s->out_size -= samples * sizeof(float);
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}
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|
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finish:
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s->out[0] = s->out[1] = NULL;
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s->out_size = 0;
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|
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return output_samples;
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}
|
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|
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static int opus_decode_packet(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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OpusContext *c = avctx->priv_data;
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
|
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int coded_samples = 0;
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int decoded_samples = INT_MAX;
|
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int delayed_samples = 0;
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int i, ret;
|
|
|
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/* calculate the number of delayed samples */
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for (i = 0; i < c->nb_streams; i++) {
|
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OpusStreamContext *s = &c->streams[i];
|
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s->out[0] =
|
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s->out[1] = NULL;
|
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delayed_samples = FFMAX(delayed_samples,
|
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s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
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}
|
|
|
|
/* decode the header of the first sub-packet to find out the sample count */
|
|
if (buf) {
|
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OpusPacket *pkt = &c->streams[0].packet;
|
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ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
|
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if (ret < 0) {
|
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av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
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return ret;
|
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}
|
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coded_samples += pkt->frame_count * pkt->frame_duration;
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c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
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}
|
|
|
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frame->nb_samples = coded_samples + delayed_samples;
|
|
|
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/* no input or buffered data => nothing to do */
|
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if (!frame->nb_samples) {
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*got_frame_ptr = 0;
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return 0;
|
|
}
|
|
|
|
/* setup the data buffers */
|
|
ret = ff_get_buffer(avctx, frame, 0);
|
|
if (ret < 0)
|
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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];
|
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}
|
|
|
|
/* 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, c->apply_phase_inv);
|
|
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;
|
|
}
|
|
|
|
#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,
|
|
};
|
|
|
|
AVCodec ff_opus_decoder = {
|
|
.name = "opus",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Opus"),
|
|
.priv_class = &opus_class,
|
|
.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,
|
|
};
|