3 * Copyright (c) 2012 Andrew D'Addesio
4 * Copyright (c) 2013-2014 Mozilla Corporation
6 * This file is part of Libav.
8 * Libav is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
13 * Libav is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with Libav; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26 * @author Andrew D'Addesio, Anton Khirnov
28 * Codec homepage: http://opus-codec.org/
29 * Specification: http://tools.ietf.org/html/rfc6716
30 * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
32 * Ogg-contained .opus files can be produced with opus-tools:
33 * http://git.xiph.org/?p=opus-tools.git
38 #include "libavutil/attributes.h"
39 #include "libavutil/audio_fifo.h"
40 #include "libavutil/channel_layout.h"
41 #include "libavutil/opt.h"
43 #include "libavresample/avresample.h"
46 #include "celp_filters.h"
53 static const uint16_t silk_frame_duration_ms[16] = {
61 /* number of samples of silence to feed to the resampler
63 static const int silk_resample_delay[] = {
67 static const uint8_t celt_band_end[] = { 13, 17, 17, 19, 21 };
69 static int get_silk_samplerate(int config)
81 static int opus_rc_init(OpusRangeCoder *rc, const uint8_t *data, int size)
83 int ret = init_get_bits8(&rc->gb, data, size);
88 rc->value = 127 - get_bits(&rc->gb, 7);
89 rc->total_read_bits = 9;
90 opus_rc_normalize(rc);
95 static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend,
98 rc->rb.position = rightend;
104 static void opus_fade(float *out,
105 const float *in1, const float *in2,
106 const float *window, int len)
109 for (i = 0; i < len; i++)
110 out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
113 static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
115 int celt_size = av_audio_fifo_size(s->celt_delay);
118 ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
122 else if (ret != nb_samples) {
123 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
129 if (celt_size != nb_samples) {
130 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
133 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
134 for (i = 0; i < s->output_channels; i++) {
135 s->fdsp->vector_fmac_scalar(s->out[i],
136 s->celt_output[i], 1.0,
141 if (s->redundancy_idx) {
142 for (i = 0; i < s->output_channels; i++)
143 opus_fade(s->out[i], s->out[i],
144 s->redundancy_output[i] + 120 + s->redundancy_idx,
145 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
146 s->redundancy_idx = 0;
149 s->out[0] += nb_samples;
150 s->out[1] += nb_samples;
151 s->out_size -= nb_samples * sizeof(float);
156 static int opus_init_resample(OpusStreamContext *s)
158 float delay[16] = { 0.0 };
159 uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
162 av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
163 ret = avresample_open(s->avr);
165 av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
169 ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
170 silk_resample_delay[s->packet.bandwidth]);
172 av_log(s->avctx, AV_LOG_ERROR,
173 "Error feeding initial silence to the resampler.\n");
180 static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
183 enum OpusBandwidth bw = s->packet.bandwidth;
185 if (s->packet.mode == OPUS_MODE_SILK &&
186 bw == OPUS_BANDWIDTH_MEDIUMBAND)
187 bw = OPUS_BANDWIDTH_WIDEBAND;
189 ret = opus_rc_init(&s->redundancy_rc, data, size);
192 opus_raw_init(&s->redundancy_rc, data + size, size);
194 ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
195 s->redundancy_output,
196 s->packet.stereo + 1, 240,
197 0, celt_band_end[s->packet.bandwidth]);
203 av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
207 static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
209 int samples = s->packet.frame_duration;
211 int redundancy_size, redundancy_pos;
212 int ret, i, consumed;
213 int delayed_samples = s->delayed_samples;
215 ret = opus_rc_init(&s->rc, data, size);
219 /* decode the silk frame */
220 if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
221 if (!avresample_is_open(s->avr)) {
222 ret = opus_init_resample(s);
227 samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
228 FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
229 s->packet.stereo + 1,
230 silk_frame_duration_ms[s->packet.config]);
232 av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
236 samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
237 s->packet.frame_duration,
238 (uint8_t**)s->silk_output,
239 sizeof(s->silk_buf[0]),
242 av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
245 s->delayed_samples += s->packet.frame_duration - samples;
247 ff_silk_flush(s->silk);
249 // decode redundancy information
250 consumed = opus_rc_tell(&s->rc);
251 if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
252 redundancy = opus_rc_p2model(&s->rc, 12);
253 else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
257 redundancy_pos = opus_rc_p2model(&s->rc, 1);
259 if (s->packet.mode == OPUS_MODE_HYBRID)
260 redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
262 redundancy_size = size - (consumed + 7) / 8;
263 size -= redundancy_size;
265 av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
266 return AVERROR_INVALIDDATA;
269 if (redundancy_pos) {
270 ret = opus_decode_redundancy(s, data + size, redundancy_size);
273 ff_celt_flush(s->celt);
277 /* decode the CELT frame */
278 if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
279 float *out_tmp[2] = { s->out[0], s->out[1] };
280 float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
281 out_tmp : s->celt_output;
282 int celt_output_samples = samples;
283 int delay_samples = av_audio_fifo_size(s->celt_delay);
286 if (s->packet.mode == OPUS_MODE_HYBRID) {
287 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
289 for (i = 0; i < s->output_channels; i++) {
290 s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
292 out_tmp[i] += delay_samples;
294 celt_output_samples -= delay_samples;
296 av_log(s->avctx, AV_LOG_WARNING,
297 "Spurious CELT delay samples present.\n");
298 av_audio_fifo_drain(s->celt_delay, delay_samples);
299 if (s->avctx->err_recognition & AV_EF_EXPLODE)
304 opus_raw_init(&s->rc, data + size, size);
306 ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
307 s->packet.stereo + 1,
308 s->packet.frame_duration,
309 (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
310 celt_band_end[s->packet.bandwidth]);
314 if (s->packet.mode == OPUS_MODE_HYBRID) {
315 int celt_delay = s->packet.frame_duration - celt_output_samples;
316 void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
317 s->celt_output[1] + celt_output_samples };
319 for (i = 0; i < s->output_channels; i++) {
320 s->fdsp->vector_fmac_scalar(out_tmp[i],
321 s->celt_output[i], 1.0,
322 celt_output_samples);
325 ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
330 ff_celt_flush(s->celt);
332 if (s->redundancy_idx) {
333 for (i = 0; i < s->output_channels; i++)
334 opus_fade(s->out[i], s->out[i],
335 s->redundancy_output[i] + 120 + s->redundancy_idx,
336 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
337 s->redundancy_idx = 0;
340 if (!redundancy_pos) {
341 ff_celt_flush(s->celt);
342 ret = opus_decode_redundancy(s, data + size, redundancy_size);
346 for (i = 0; i < s->output_channels; i++) {
347 opus_fade(s->out[i] + samples - 120 + delayed_samples,
348 s->out[i] + samples - 120 + delayed_samples,
349 s->redundancy_output[i] + 120,
350 ff_celt_window2, 120 - delayed_samples);
352 s->redundancy_idx = 120 - delayed_samples;
355 for (i = 0; i < s->output_channels; i++) {
356 memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
357 opus_fade(s->out[i] + 120 + delayed_samples,
358 s->redundancy_output[i] + 120,
359 s->out[i] + 120 + delayed_samples,
360 ff_celt_window2, 120);
368 static int opus_decode_subpacket(OpusStreamContext *s,
369 const uint8_t *buf, int buf_size,
372 int output_samples = 0;
373 int flush_needed = 0;
376 /* check if we need to flush the resampler */
377 if (avresample_is_open(s->avr)) {
379 int64_t cur_samplerate;
380 av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
381 flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
383 flush_needed = !!s->delayed_samples;
387 if (!buf && !flush_needed)
390 /* use dummy output buffers if the channel is not mapped to anything */
392 (s->output_channels == 2 && !s->out[1])) {
393 av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
395 return AVERROR(ENOMEM);
397 s->out[0] = s->out_dummy;
399 s->out[1] = s->out_dummy;
402 /* flush the resampler if necessary */
404 ret = opus_flush_resample(s, s->delayed_samples);
406 av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
409 avresample_close(s->avr);
410 output_samples += s->delayed_samples;
411 s->delayed_samples = 0;
417 /* decode all the frames in the packet */
418 for (i = 0; i < s->packet.frame_count; i++) {
419 int size = s->packet.frame_size[i];
420 int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
423 av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
424 if (s->avctx->err_recognition & AV_EF_EXPLODE)
427 for (j = 0; j < s->output_channels; j++)
428 memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
429 samples = s->packet.frame_duration;
431 output_samples += samples;
433 for (j = 0; j < s->output_channels; j++)
434 s->out[j] += samples;
435 s->out_size -= samples * sizeof(float);
439 s->out[0] = s->out[1] = NULL;
442 return output_samples;
445 static int opus_decode_packet(AVCodecContext *avctx, void *data,
446 int *got_frame_ptr, AVPacket *avpkt)
448 OpusContext *c = avctx->priv_data;
449 AVFrame *frame = data;
450 const uint8_t *buf = avpkt->data;
451 int buf_size = avpkt->size;
452 int coded_samples = 0;
453 int decoded_samples = 0;
456 /* decode the header of the first sub-packet to find out the sample count */
458 OpusPacket *pkt = &c->streams[0].packet;
459 ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
461 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
464 coded_samples += pkt->frame_count * pkt->frame_duration;
465 c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
468 frame->nb_samples = coded_samples + c->streams[0].delayed_samples;
470 /* no input or buffered data => nothing to do */
471 if (!frame->nb_samples) {
476 /* setup the data buffers */
477 ret = ff_get_buffer(avctx, frame, 0);
479 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
482 frame->nb_samples = 0;
484 for (i = 0; i < avctx->channels; i++) {
485 ChannelMap *map = &c->channel_maps[i];
487 c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
490 for (i = 0; i < c->nb_streams; i++)
491 c->streams[i].out_size = frame->linesize[0];
493 /* decode each sub-packet */
494 for (i = 0; i < c->nb_streams; i++) {
495 OpusStreamContext *s = &c->streams[i];
498 ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
500 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
503 s->silk_samplerate = get_silk_samplerate(s->packet.config);
506 ret = opus_decode_subpacket(&c->streams[i], buf,
507 s->packet.data_size, coded_samples);
510 if (decoded_samples && ret != decoded_samples) {
511 av_log(avctx, AV_LOG_ERROR, "Different numbers of decoded samples "
512 "in a multi-channel stream\n");
513 return AVERROR_INVALIDDATA;
515 decoded_samples = ret;
516 buf += s->packet.packet_size;
517 buf_size -= s->packet.packet_size;
520 for (i = 0; i < avctx->channels; i++) {
521 ChannelMap *map = &c->channel_maps[i];
523 /* handle copied channels */
525 memcpy(frame->extended_data[i],
526 frame->extended_data[map->copy_idx],
528 } else if (map->silence) {
529 memset(frame->extended_data[i], 0, frame->linesize[0]);
533 c->fdsp.vector_fmul_scalar((float*)frame->extended_data[i],
534 (float*)frame->extended_data[i],
535 c->gain, FFALIGN(decoded_samples, 8));
539 frame->nb_samples = decoded_samples;
540 *got_frame_ptr = !!decoded_samples;
545 static av_cold void opus_decode_flush(AVCodecContext *ctx)
547 OpusContext *c = ctx->priv_data;
550 for (i = 0; i < c->nb_streams; i++) {
551 OpusStreamContext *s = &c->streams[i];
553 memset(&s->packet, 0, sizeof(s->packet));
554 s->delayed_samples = 0;
557 av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
558 avresample_close(s->avr);
560 ff_silk_flush(s->silk);
561 ff_celt_flush(s->celt);
565 static av_cold int opus_decode_close(AVCodecContext *avctx)
567 OpusContext *c = avctx->priv_data;
570 for (i = 0; i < c->nb_streams; i++) {
571 OpusStreamContext *s = &c->streams[i];
573 ff_silk_free(&s->silk);
574 ff_celt_free(&s->celt);
576 av_freep(&s->out_dummy);
577 s->out_dummy_allocated_size = 0;
579 av_audio_fifo_free(s->celt_delay);
580 avresample_free(&s->avr);
583 av_freep(&c->streams);
586 av_freep(&c->channel_maps);
591 static av_cold int opus_decode_init(AVCodecContext *avctx)
593 OpusContext *c = avctx->priv_data;
596 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
597 avctx->sample_rate = 48000;
599 avpriv_float_dsp_init(&c->fdsp, 0);
601 /* find out the channel configuration */
602 ret = ff_opus_parse_extradata(avctx, c);
606 /* allocate and init each independent decoder */
607 c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
610 ret = AVERROR(ENOMEM);
614 for (i = 0; i < c->nb_streams; i++) {
615 OpusStreamContext *s = &c->streams[i];
618 s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
622 for (j = 0; j < s->output_channels; j++) {
623 s->silk_output[j] = s->silk_buf[j];
624 s->celt_output[j] = s->celt_buf[j];
625 s->redundancy_output[j] = s->redundancy_buf[j];
630 s->avr = avresample_alloc_context();
634 layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
635 av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0);
636 av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0);
637 av_opt_set_int(s->avr, "in_channel_layout", layout, 0);
638 av_opt_set_int(s->avr, "out_channel_layout", layout, 0);
639 av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0);
641 ret = ff_silk_init(avctx, &s->silk, s->output_channels);
645 ret = ff_celt_init(avctx, &s->celt, s->output_channels);
649 s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
650 s->output_channels, 1024);
651 if (!s->celt_delay) {
652 ret = AVERROR(ENOMEM);
659 opus_decode_close(avctx);
663 AVCodec ff_opus_decoder = {
665 .long_name = NULL_IF_CONFIG_SMALL("Opus"),
666 .type = AVMEDIA_TYPE_AUDIO,
667 .id = AV_CODEC_ID_OPUS,
668 .priv_data_size = sizeof(OpusContext),
669 .init = opus_decode_init,
670 .close = opus_decode_close,
671 .decode = opus_decode_packet,
672 .flush = opus_decode_flush,
673 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_DELAY,