libavcodec/opusdec.c - third_party/ffmpeg - Git at Google

 /*
  * 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 = 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, 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,
 };
	/*
	* 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 = 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, 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,
	};