lavc: add a native Opus decoder.

Initial implementation by Andrew D'Addesio <modchipv12@gmail.com> during
GSoC 2012.

Completion by Anton Khirnov <anton@khirnov.net>, sponsored by the
Mozilla Corporation.

Further contributions by:
Christophe Gisquet <christophe.gisquet@gmail.com>
Janne Grunau <janne-libav@jannau.net>
Luca Barbato <lu_zero@gentoo.org>

1 files changed, 674 insertions, 0 deletions

diff --git a/libavcodec/opusdec.c b/libavcodec/opusdec.c
new file mode 100644
index 0000000000..bf3a54b16b
--- /dev/null
+++ b/libavcodec/opusdec.c
@@ -0,0 +1,674 @@
+/*
+ * Opus decoder
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Opus decoder
+ * @author Andrew D'Addesio, Anton Khirnov
+ *
+ * Codec homepage: http://opus-codec.org/
+ * Specification: http://tools.ietf.org/html/rfc6716
+ * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
+ *
+ * Ogg-contained .opus files can be produced with opus-tools:
+ * http://git.xiph.org/?p=opus-tools.git
+ */
+
+#include <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+
+#include "libavresample/avresample.h"
+
+#include "avcodec.h"
+#include "celp_filters.h"
+#include "fft.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "mathops.h"
+#include "opus.h"
+
+static const uint16_t silk_frame_duration_ms[16] = {
+    10, 20, 40, 60,
+    10, 20, 40, 60,
+    10, 20, 40, 60,
+    10, 20,
+    10, 20,
+};
+
+/* number of samples of silence to feed to the resampler
+ * at the beginning */
+static const int silk_resample_delay[] = {
+    4, 8, 11, 11, 11
+};
+
+static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 };
+
+static int get_silk_samplerate(int config)
+{
+    if (config < 4)
+        return 8000;
+    else if (config < 8)
+        return 12000;
+    return 16000;
+}
+
+/**
+ * Range decoder
+ */
+static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)
+{
+    int ret = init_get_bits8(&rc->gb, data, size);
+    if (ret < 0)
+        return ret;
+
+    rc->range = 128;
+    rc->value = 127 - get_bits(&rc->gb, 7);
+    rc->total_read_bits = 9;
+    opus_rc_normalize(rc);
+
+    return 0;
+}
+
+static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend,
+                          unsigned int bytes)
+{
+    rc->rb.position = rightend;
+    rc->rb.bytes    = bytes;
+    rc->rb.cachelen = 0;
+    rc->rb.cacheval = 0;
+}
+
+static void opus_fade(float *out,
+                      const float *in1, const float *in2,
+                      const float *window, int len)
+{
+    int i;
+    for (i = 0; i < len; i++)
+        out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
+}
+
+static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
+{
+    int celt_size = av_audio_fifo_size(s->celt_delay);
+    int ret, i;
+
+    ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
+                             NULL, 0, 0);
+    if (ret < 0)
+        return ret;
+    else if (ret != nb_samples) {
+        av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
+               ret);
+        return AVERROR_BUG;
+    }
+
+    if (celt_size) {
+        if (celt_size != nb_samples) {
+            av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
+            return AVERROR_BUG;
+        }
+        av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
+        for (i = 0; i < s->output_channels; i++) {
+            s->fdsp->vector_fmac_scalar(s->out[i],
+                                        s->celt_output[i], 1.0,
+                                        nb_samples);
+        }
+    }
+
+    if (s->redundancy_idx) {
+        for (i = 0; i < s->output_channels; i++)
+            opus_fade(s->out[i], s->out[i],
+                      s->redundancy_output[i] + 120 + s->redundancy_idx,
+                      ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
+        s->redundancy_idx = 0;
+    }
+
+    s->out[0]   += nb_samples;
+    s->out[1]   += nb_samples;
+    s->out_size -= nb_samples * sizeof(float);
+
+    return 0;
+}
+
+static int opus_init_resample(OpusStreamContext *s)
+{
+    float delay[16] = { 0.0 };
+    uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
+    int ret;
+
+    av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
+    ret = avresample_open(s->avr);
+    if (ret < 0) {
+        av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
+        return ret;
+    }
+
+    ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
+                             silk_resample_delay[s->packet.bandwidth]);
+    if (ret < 0) {
+        av_log(s->avctx, AV_LOG_ERROR,
+               "Error feeding initial silence to the resampler.\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
+{
+    int ret;
+    enum OpusBandwidth bw = s->packet.bandwidth;
+
+    if (s->packet.mode == OPUS_MODE_SILK &&
+        bw == OPUS_BANDWIDTH_MEDIUMBAND)
+        bw = OPUS_BANDWIDTH_WIDEBAND;
+
+    ret = opus_rc_init(&s->redundancy_rc, data, size);
+    if (ret < 0)
+        goto fail;
+    opus_raw_init(&s->redundancy_rc, data + size, size);
+
+    ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
+                               s->redundancy_output,
+                               s->packet.stereo + 1, 240,
+                               0, celt_band_end[s->packet.bandwidth]);
+    if (ret < 0)
+        goto fail;
+
+    return 0;
+fail:
+    av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
+    return ret;
+}
+
+static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
+{
+    int samples    = s->packet.frame_duration;
+    int redundancy = 0;
+    int redundancy_size, redundancy_pos;
+    int ret, i, consumed;
+    int delayed_samples = s->delayed_samples;
+
+    ret = opus_rc_init(&s->rc, data, size);
+    if (ret < 0)
+        return ret;
+
+    /* decode the silk frame */
+    if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
+        if (!avresample_is_open(s->avr)) {
+            ret = opus_init_resample(s);
+            if (ret < 0)
+                return ret;
+        }
+
+        samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
+                                            FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
+                                            s->packet.stereo + 1,
+                                            silk_frame_duration_ms[s->packet.config]);
+        if (samples < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
+            return samples;
+        }
+
+        samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
+                                     s->packet.frame_duration,
+                                     (uint8_t**)s->silk_output,
+                                     sizeof(s->silk_buf[0]),
+                                     samples);
+        if (samples < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
+            return samples;
+        }
+        s->delayed_samples += s->packet.frame_duration - samples;
+    } else
+        ff_silk_flush(s->silk);
+
+    // decode redundancy information
+    consumed = opus_rc_tell(&s->rc);
+    if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
+        redundancy = opus_rc_p2model(&s->rc, 12);
+    else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
+        redundancy = 1;
+
+    if (redundancy) {
+        redundancy_pos = opus_rc_p2model(&s->rc, 1);
+
+        if (s->packet.mode == OPUS_MODE_HYBRID)
+            redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
+        else
+            redundancy_size = size - (consumed + 7) / 8;
+        size -= redundancy_size;
+        if (size < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        if (redundancy_pos) {
+            ret = opus_decode_redundancy(s, data + size, redundancy_size);
+            if (ret < 0)
+                return ret;
+            ff_celt_flush(s->celt);
+        }
+    }
+
+    /* decode the CELT frame */
+    if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
+        float *out_tmp[2] = { s->out[0], s->out[1] };
+        float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
+                      out_tmp : s->celt_output;
+        int celt_output_samples = samples;
+        int delay_samples = av_audio_fifo_size(s->celt_delay);
+
+        if (delay_samples) {
+            if (s->packet.mode == OPUS_MODE_HYBRID) {
+                av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
+
+                for (i = 0; i < s->output_channels; i++) {
+                    s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
+                                                delay_samples);
+                    out_tmp[i] += delay_samples;
+                }
+                celt_output_samples -= delay_samples;
+            } else {
+                av_log(s->avctx, AV_LOG_WARNING,
+                       "Spurious CELT delay samples present.\n");
+                av_audio_fifo_drain(s->celt_delay, delay_samples);
+                if (s->avctx->err_recognition & AV_EF_EXPLODE)
+                    return AVERROR_BUG;
+            }
+        }
+
+        opus_raw_init(&s->rc, data + size, size);
+
+        ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
+                                   s->packet.stereo + 1,
+                                   s->packet.frame_duration,
+                                   (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
+                                   celt_band_end[s->packet.bandwidth]);
+        if (ret < 0)
+            return ret;
+
+        if (s->packet.mode == OPUS_MODE_HYBRID) {
+            int celt_delay = s->packet.frame_duration - celt_output_samples;
+            void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
+                                  s->celt_output[1] + celt_output_samples };
+
+            for (i = 0; i < s->output_channels; i++) {
+                s->fdsp->vector_fmac_scalar(out_tmp[i],
+                                            s->celt_output[i], 1.0,
+                                            celt_output_samples);
+            }
+
+            ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
+            if (ret < 0)
+                return ret;
+        }
+    } else
+        ff_celt_flush(s->celt);
+
+    if (s->redundancy_idx) {
+        for (i = 0; i < s->output_channels; i++)
+            opus_fade(s->out[i], s->out[i],
+                      s->redundancy_output[i] + 120 + s->redundancy_idx,
+                      ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
+        s->redundancy_idx = 0;
+    }
+    if (redundancy) {
+        if (!redundancy_pos) {
+            ff_celt_flush(s->celt);
+            ret = opus_decode_redundancy(s, data + size, redundancy_size);
+            if (ret < 0)
+                return ret;
+
+            for (i = 0; i < s->output_channels; i++) {
+                opus_fade(s->out[i] + samples - 120 + delayed_samples,
+                          s->out[i] + samples - 120 + delayed_samples,
+                          s->redundancy_output[i] + 120,
+                          ff_celt_window2, 120 - delayed_samples);
+                if (delayed_samples)
+                    s->redundancy_idx = 120 - delayed_samples;
+            }
+        } else {
+            for (i = 0; i < s->output_channels; i++) {
+                memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
+                opus_fade(s->out[i] + 120 + delayed_samples,
+                          s->redundancy_output[i] + 120,
+                          s->out[i] + 120 + delayed_samples,
+                          ff_celt_window2, 120);
+            }
+        }
+    }
+
+    return samples;
+}
+
+static int opus_decode_subpacket(OpusStreamContext *s,
+                                 const uint8_t *buf, int buf_size,
+                                 int nb_samples)
+{
+    int output_samples = 0;
+    int flush_needed   = 0;
+    int i, j, ret;
+
+    /* check if we need to flush the resampler */
+    if (avresample_is_open(s->avr)) {
+        if (buf) {
+            int64_t cur_samplerate;
+            av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
+            flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
+        } else {
+            flush_needed = !!s->delayed_samples;
+        }
+    }
+
+    if (!buf && !flush_needed)
+        return 0;
+
+    /* use dummy output buffers if the channel is not mapped to anything */
+    if (!s->out[0] ||
+        (s->output_channels == 2 && !s->out[1])) {
+        av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
+        if (!s->out_dummy)
+            return AVERROR(ENOMEM);
+        if (!s->out[0])
+            s->out[0] = s->out_dummy;
+        if (!s->out[1])
+            s->out[1] = s->out_dummy;
+    }
+
+    /* flush the resampler if necessary */
+    if (flush_needed) {
+        ret = opus_flush_resample(s, s->delayed_samples);
+        if (ret < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
+            return ret;
+        }
+        avresample_close(s->avr);
+        output_samples += s->delayed_samples;
+        s->delayed_samples = 0;
+
+        if (!buf)
+            goto finish;
+    }
+
+    /* decode all the frames in the packet */
+    for (i = 0; i < s->packet.frame_count; i++) {
+        int size = s->packet.frame_size[i];
+        int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
+
+        if (samples < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
+            if (s->avctx->err_recognition & AV_EF_EXPLODE)
+                return samples;
+
+            for (j = 0; j < s->output_channels; j++)
+                memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
+            samples = s->packet.frame_duration;
+        }
+        output_samples += samples;
+
+        for (j = 0; j < s->output_channels; j++)
+            s->out[j] += samples;
+        s->out_size -= samples * sizeof(float);
+    }
+
+finish:
+    s->out[0] = s->out[1] = NULL;
+    s->out_size = 0;
+
+    return output_samples;
+}
+
+static int opus_decode_packet(AVCodecContext *avctx, void *data,
+                              int *got_frame_ptr, AVPacket *avpkt)
+{
+    OpusContext *c      = avctx->priv_data;
+    AVFrame *frame      = data;
+    const uint8_t *buf  = avpkt->data;
+    int buf_size        = avpkt->size;
+    int coded_samples   = 0;
+    int decoded_samples = 0;
+    int i, ret;
+
+    /* decode the header of the first sub-packet to find out the sample count */
+    if (buf) {
+        OpusPacket *pkt = &c->streams[0].packet;
+        ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
+            return ret;
+        }
+        coded_samples += pkt->frame_count * pkt->frame_duration;
+        c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
+    }
+
+    frame->nb_samples = coded_samples + c->streams[0].delayed_samples;
+
+    /* no input or buffered data => nothing to do */
+    if (!frame->nb_samples) {
+        *got_frame_ptr = 0;
+        return 0;
+    }
+
+    /* setup the data buffers */
+    ret = ff_get_buffer(avctx, frame, 0);
+    if (ret < 0) {
+        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
+    }
+    frame->nb_samples = 0;
+
+    for (i = 0; i < avctx->channels; i++) {
+        ChannelMap *map = &c->channel_maps[i];
+        if (!map->copy)
+            c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
+    }
+
+    for (i = 0; i < c->nb_streams; i++)
+        c->streams[i].out_size = frame->linesize[0];
+
+    /* decode each sub-packet */
+    for (i = 0; i < c->nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+
+        if (i && buf) {
+            ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
+            if (ret < 0) {
+                av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
+                return ret;
+            }
+            s->silk_samplerate = get_silk_samplerate(s->packet.config);
+        }
+
+        ret = opus_decode_subpacket(&c->streams[i], buf,
+                                    s->packet.data_size, coded_samples);
+        if (ret < 0)
+            return ret;
+        if (decoded_samples && ret != decoded_samples) {
+            av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "
+                   "in a multi-channel stream\n");
+            return AVERROR_INVALIDDATA;
+        }
+        decoded_samples = ret;
+        buf      += s->packet.packet_size;
+        buf_size -= s->packet.packet_size;
+    }
+
+    for (i = 0; i < avctx->channels; i++) {
+        ChannelMap *map = &c->channel_maps[i];
+
+        /* handle copied channels */
+        if (map->copy) {
+            memcpy(frame->extended_data[i],
+                   frame->extended_data[map->copy_idx],
+                   frame->linesize[0]);
+        } else if (map->silence) {
+            memset(frame->extended_data[i], 0, frame->linesize[0]);
+        }
+
+        if (c->gain_i) {
+            c->fdsp.vector_fmul_scalar((float*)frame->extended_data[i],
+                                       (float*)frame->extended_data[i],
+                                       c->gain, FFALIGN(decoded_samples, 8));
+        }
+    }
+
+    frame->nb_samples = decoded_samples;
+    *got_frame_ptr    = !!decoded_samples;
+
+    return avpkt->size;
+}
+
+static av_cold void opus_decode_flush(AVCodecContext *ctx)
+{
+    OpusContext *c = ctx->priv_data;
+    int i;
+
+    for (i = 0; i < c->nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+
+        memset(&s->packet, 0, sizeof(s->packet));
+        s->delayed_samples = 0;
+
+        if (s->celt_delay)
+            av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
+        avresample_close(s->avr);
+
+        ff_silk_flush(s->silk);
+        ff_celt_flush(s->celt);
+    }
+}
+
+static av_cold int opus_decode_close(AVCodecContext *avctx)
+{
+    OpusContext *c = avctx->priv_data;
+    int i;
+
+    for (i = 0; i < c->nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+
+        ff_silk_free(&s->silk);
+        ff_celt_free(&s->celt);
+
+        av_freep(&s->out_dummy);
+        s->out_dummy_allocated_size = 0;
+
+        av_audio_fifo_free(s->celt_delay);
+        avresample_free(&s->avr);
+    }
+
+    av_freep(&c->streams);
+    c->nb_streams = 0;
+
+    av_freep(&c->channel_maps);
+
+    return 0;
+}
+
+static av_cold int opus_decode_init(AVCodecContext *avctx)
+{
+    OpusContext *c = avctx->priv_data;
+    int ret, i, j;
+
+    avctx->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+    avctx->sample_rate = 48000;
+
+    avpriv_float_dsp_init(&c->fdsp, 0);
+
+    /* find out the channel configuration */
+    ret = ff_opus_parse_extradata(avctx, c);
+    if (ret < 0)
+        return ret;
+
+    /* allocate and init each independent decoder */
+    c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
+    if (!c->streams) {
+        c->nb_streams = 0;
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    for (i = 0; i < c->nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+        uint64_t layout;
+
+        s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
+
+        s->avctx = avctx;
+
+        for (j = 0; j < s->output_channels; j++) {
+            s->silk_output[j]       = s->silk_buf[j];
+            s->celt_output[j]       = s->celt_buf[j];
+            s->redundancy_output[j] = s->redundancy_buf[j];
+        }
+
+        s->fdsp = &c->fdsp;
+
+        s->avr = avresample_alloc_context();
+        if (!s->avr)
+            goto fail;
+
+        layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
+        av_opt_set_int(s->avr, "in_sample_fmt",      avctx->sample_fmt,  0);
+        av_opt_set_int(s->avr, "out_sample_fmt",     avctx->sample_fmt,  0);
+        av_opt_set_int(s->avr, "in_channel_layout",  layout,             0);
+        av_opt_set_int(s->avr, "out_channel_layout", layout,             0);
+        av_opt_set_int(s->avr, "out_sample_rate",    avctx->sample_rate, 0);
+
+        ret = ff_silk_init(avctx, &s->silk, s->output_channels);
+        if (ret < 0)
+            goto fail;
+
+        ret = ff_celt_init(avctx, &s->celt, s->output_channels);
+        if (ret < 0)
+            goto fail;
+
+        s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
+                                            s->output_channels, 1024);
+        if (!s->celt_delay) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+    }
+
+    return 0;
+fail:
+    opus_decode_close(avctx);
+    return ret;
+}
+
+AVCodec ff_opus_decoder = {
+    .name            = "opus",
+    .long_name       = NULL_IF_CONFIG_SMALL("Opus"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_OPUS,
+    .priv_data_size  = sizeof(OpusContext),
+    .init            = opus_decode_init,
+    .close           = opus_decode_close,
+    .decode          = opus_decode_packet,
+    .flush           = opus_decode_flush,
+    .capabilities    = CODEC_CAP_DR1 | CODEC_CAP_DELAY,
+};