FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/opusdec.c
Date: 2021-09-24 20:55:06
Exec Total Coverage
Lines: 283 392 72.2%
Branches: 155 234 66.2%

Line Branch Exec Source
1 /*
2 * Opus decoder
3 * Copyright (c) 2012 Andrew D'Addesio
4 * Copyright (c) 2013-2014 Mozilla Corporation
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg 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.
12 *
13 * FFmpeg 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.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23 /**
24 * @file
25 * Opus decoder
26 * @author Andrew D'Addesio, Anton Khirnov
27 *
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
31 *
32 * Ogg-contained .opus files can be produced with opus-tools:
33 * http://git.xiph.org/?p=opus-tools.git
34 */
35
36 #include <stdint.h>
37
38 #include "libavutil/attributes.h"
39 #include "libavutil/audio_fifo.h"
40 #include "libavutil/channel_layout.h"
41 #include "libavutil/opt.h"
42
43 #include "libswresample/swresample.h"
44
45 #include "avcodec.h"
46 #include "get_bits.h"
47 #include "internal.h"
48 #include "mathops.h"
49 #include "opus.h"
50 #include "opustab.h"
51 #include "opus_celt.h"
52
53 static const uint16_t silk_frame_duration_ms[16] = {
54 10, 20, 40, 60,
55 10, 20, 40, 60,
56 10, 20, 40, 60,
57 10, 20,
58 10, 20,
59 };
60
61 /* number of samples of silence to feed to the resampler
62 * at the beginning */
63 static const int silk_resample_delay[] = {
64 4, 8, 11, 11, 11
65 };
66
67 26219 static int get_silk_samplerate(int config)
68 {
69
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26219 if (config < 4)
70 1551 return 8000;
71
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24668 else if (config < 8)
72 1222 return 12000;
73 23446 return 16000;
74 }
75
76 68 static void opus_fade(float *out,
77 const float *in1, const float *in2,
78 const float *window, int len)
79 {
80 int i;
81
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7268 for (i = 0; i < len; i++)
82 7200 out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
83 68 }
84
85 17 static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
86 {
87 17 int celt_size = av_audio_fifo_size(s->celt_delay);
88 int ret, i;
89 17 ret = swr_convert(s->swr,
90 17 (uint8_t**)s->cur_out, nb_samples,
91 NULL, 0);
92
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17 if (ret < 0)
93 return ret;
94
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17 else if (ret != nb_samples) {
95 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
96 ret);
97 return AVERROR_BUG;
98 }
99
100
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17 if (celt_size) {
101 if (celt_size != nb_samples) {
102 av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
103 return AVERROR_BUG;
104 }
105 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
106 for (i = 0; i < s->output_channels; i++) {
107 s->fdsp->vector_fmac_scalar(s->cur_out[i],
108 s->celt_output[i], 1.0,
109 nb_samples);
110 }
111 }
112
113
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17 if (s->redundancy_idx) {
114
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12 for (i = 0; i < s->output_channels; i++)
115 8 opus_fade(s->cur_out[i], s->cur_out[i],
116 8 s->redundancy_output[i] + 120 + s->redundancy_idx,
117 8 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
118 4 s->redundancy_idx = 0;
119 }
120
121 17 s->cur_out[0] += nb_samples;
122 17 s->cur_out[1] += nb_samples;
123 17 s->remaining_out_size -= nb_samples * sizeof(float);
124
125 17 return 0;
126 }
127
128 32 static int opus_init_resample(OpusStreamContext *s)
129 {
130 static const float delay[16] = { 0.0 };
131 32 const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
132 int ret;
133
134 32 av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
135 32 ret = swr_init(s->swr);
136
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32 if (ret < 0) {
137 av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
138 return ret;
139 }
140
141 32 ret = swr_convert(s->swr,
142 NULL, 0,
143 32 delayptr, silk_resample_delay[s->packet.bandwidth]);
144
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32 if (ret < 0) {
145 av_log(s->avctx, AV_LOG_ERROR,
146 "Error feeding initial silence to the resampler.\n");
147 return ret;
148 }
149
150 32 return 0;
151 }
152
153 30 static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
154 {
155 30 int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);
156
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30 if (ret < 0)
157 goto fail;
158 30 ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size);
159
160 30 ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
161 30 s->redundancy_output,
162 30 s->packet.stereo + 1, 240,
163 30 0, ff_celt_band_end[s->packet.bandwidth]);
164
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30 if (ret < 0)
165 goto fail;
166
167 30 return 0;
168 fail:
169 av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
170 return ret;
171 }
172
173 34393 static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
174 {
175 34393 int samples = s->packet.frame_duration;
176 34393 int redundancy = 0;
177 int redundancy_size, redundancy_pos;
178 int ret, i, consumed;
179 34393 int delayed_samples = s->delayed_samples;
180
181 34393 ret = ff_opus_rc_dec_init(&s->rc, data, size);
182
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34393 if (ret < 0)
183 return ret;
184
185 /* decode the silk frame */
186
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34393 if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
187
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9471 if (!swr_is_initialized(s->swr)) {
188 32 ret = opus_init_resample(s);
189
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32 if (ret < 0)
190 return ret;
191 }
192
193 9471 samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
194 9471 FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
195 9471 s->packet.stereo + 1,
196 9471 silk_frame_duration_ms[s->packet.config]);
197
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9471 if (samples < 0) {
198 av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
199 return samples;
200 }
201 9471 samples = swr_convert(s->swr,
202 9471 (uint8_t**)s->cur_out, s->packet.frame_duration,
203 9471 (const uint8_t**)s->silk_output, samples);
204
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9471 if (samples < 0) {
205 av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
206 return samples;
207 }
208 av_assert2((samples & 7) == 0);
209 9471 s->delayed_samples += s->packet.frame_duration - samples;
210 } else
211 24922 ff_silk_flush(s->silk);
212
213 // decode redundancy information
214 34393 consumed = opus_rc_tell(&s->rc);
215
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34393 if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
216 4904 redundancy = ff_opus_rc_dec_log(&s->rc, 12);
217
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29489 else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
218 14 redundancy = 1;
219
220
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34393 if (redundancy) {
221 30 redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
222
223
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30 if (s->packet.mode == OPUS_MODE_HYBRID)
224 16 redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
225 else
226 14 redundancy_size = size - (consumed + 7) / 8;
227 30 size -= redundancy_size;
228
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30 if (size < 0) {
229 av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
230 return AVERROR_INVALIDDATA;
231 }
232
233
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30 if (redundancy_pos) {
234 14 ret = opus_decode_redundancy(s, data + size, redundancy_size);
235
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14 if (ret < 0)
236 return ret;
237 14 ff_celt_flush(s->celt);
238 }
239 }
240
241 /* decode the CELT frame */
242
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64253 if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
243 29860 float *out_tmp[2] = { s->cur_out[0], s->cur_out[1] };
244 59720 float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
245
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29860 out_tmp : s->celt_output;
246 29860 int celt_output_samples = samples;
247 29860 int delay_samples = av_audio_fifo_size(s->celt_delay);
248
249
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29860 if (delay_samples) {
250 if (s->packet.mode == OPUS_MODE_HYBRID) {
251 av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
252
253 for (i = 0; i < s->output_channels; i++) {
254 s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
255 delay_samples);
256 out_tmp[i] += delay_samples;
257 }
258 celt_output_samples -= delay_samples;
259 } else {
260 av_log(s->avctx, AV_LOG_WARNING,
261 "Spurious CELT delay samples present.\n");
262 av_audio_fifo_drain(s->celt_delay, delay_samples);
263 if (s->avctx->err_recognition & AV_EF_EXPLODE)
264 return AVERROR_BUG;
265 }
266 }
267
268 29860 ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
269
270 29860 ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
271 29860 s->packet.stereo + 1,
272 s->packet.frame_duration,
273 29860 (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
274
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29860 ff_celt_band_end[s->packet.bandwidth]);
275
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29860 if (ret < 0)
276 return ret;
277
278
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29860 if (s->packet.mode == OPUS_MODE_HYBRID) {
279 4938 int celt_delay = s->packet.frame_duration - celt_output_samples;
280 4938 void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
281 4938 s->celt_output[1] + celt_output_samples };
282
283
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14814 for (i = 0; i < s->output_channels; i++) {
284 9876 s->fdsp->vector_fmac_scalar(out_tmp[i],
285 9876 s->celt_output[i], 1.0,
286 celt_output_samples);
287 }
288
289 4938 ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
290
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4938 if (ret < 0)
291 return ret;
292 }
293 } else
294 4533 ff_celt_flush(s->celt);
295
296
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34393 if (s->redundancy_idx) {
297 for (i = 0; i < s->output_channels; i++)
298 opus_fade(s->cur_out[i], s->cur_out[i],
299 s->redundancy_output[i] + 120 + s->redundancy_idx,
300 ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
301 s->redundancy_idx = 0;
302 }
303
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34393 if (redundancy) {
304
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30 if (!redundancy_pos) {
305 16 ff_celt_flush(s->celt);
306 16 ret = opus_decode_redundancy(s, data + size, redundancy_size);
307
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16 if (ret < 0)
308 return ret;
309
310
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48 for (i = 0; i < s->output_channels; i++) {
311 32 opus_fade(s->cur_out[i] + samples - 120 + delayed_samples,
312 32 s->cur_out[i] + samples - 120 + delayed_samples,
313 32 s->redundancy_output[i] + 120,
314 ff_celt_window2, 120 - delayed_samples);
315
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32 if (delayed_samples)
316 8 s->redundancy_idx = 120 - delayed_samples;
317 }
318 } else {
319
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42 for (i = 0; i < s->output_channels; i++) {
320 28 memcpy(s->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
321 28 opus_fade(s->cur_out[i] + 120 + delayed_samples,
322 28 s->redundancy_output[i] + 120,
323 28 s->cur_out[i] + 120 + delayed_samples,
324 ff_celt_window2, 120);
325 }
326 }
327 }
328
329 34393 return samples;
330 }
331
332 26220 static int opus_decode_subpacket(OpusStreamContext *s,
333 const uint8_t *buf, int buf_size,
334 int nb_samples)
335 {
336 26220 int output_samples = 0;
337 26220 int flush_needed = 0;
338 int i, j, ret;
339
340 26220 s->cur_out[0] = s->out[0];
341 26220 s->cur_out[1] = s->out[1];
342 26220 s->remaining_out_size = s->out_size;
343
344 /* check if we need to flush the resampler */
345
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26220 if (swr_is_initialized(s->swr)) {
346
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9012 if (buf) {
347 int64_t cur_samplerate;
348 9011 av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
349
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9011 flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
350 } else {
351 1 flush_needed = !!s->delayed_samples;
352 }
353 }
354
355
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26220 if (!buf && !flush_needed)
356 return 0;
357
358 /* use dummy output buffers if the channel is not mapped to anything */
359
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26220 if (!s->cur_out[0] ||
360
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26220 (s->output_channels == 2 && !s->cur_out[1])) {
361 av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size,
362 s->remaining_out_size);
363 if (!s->out_dummy)
364 return AVERROR(ENOMEM);
365 if (!s->cur_out[0])
366 s->cur_out[0] = s->out_dummy;
367 if (!s->cur_out[1])
368 s->cur_out[1] = s->out_dummy;
369 }
370
371 /* flush the resampler if necessary */
372
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26220 if (flush_needed) {
373 17 ret = opus_flush_resample(s, s->delayed_samples);
374
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17 if (ret < 0) {
375 av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
376 return ret;
377 }
378 17 swr_close(s->swr);
379 17 output_samples += s->delayed_samples;
380 17 s->delayed_samples = 0;
381
382
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17 if (!buf)
383 1 goto finish;
384 }
385
386 /* decode all the frames in the packet */
387
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60612 for (i = 0; i < s->packet.frame_count; i++) {
388 34393 int size = s->packet.frame_size[i];
389 34393 int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
390
391
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34393 if (samples < 0) {
392 av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
393 if (s->avctx->err_recognition & AV_EF_EXPLODE)
394 return samples;
395
396 for (j = 0; j < s->output_channels; j++)
397 memset(s->cur_out[j], 0, s->packet.frame_duration * sizeof(float));
398 samples = s->packet.frame_duration;
399 }
400 34393 output_samples += samples;
401
402
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100113 for (j = 0; j < s->output_channels; j++)
403 65720 s->cur_out[j] += samples;
404 34393 s->remaining_out_size -= samples * sizeof(float);
405 }
406
407 26219 finish:
408 26220 s->cur_out[0] = s->cur_out[1] = NULL;
409 26220 s->remaining_out_size = 0;
410
411 26220 return output_samples;
412 }
413
414 21636 static int opus_decode_packet(AVCodecContext *avctx, void *data,
415 int *got_frame_ptr, AVPacket *avpkt)
416 {
417 21636 OpusContext *c = avctx->priv_data;
418 21636 AVFrame *frame = data;
419 21636 const uint8_t *buf = avpkt->data;
420 21636 int buf_size = avpkt->size;
421 21636 int coded_samples = 0;
422 21636 int decoded_samples = INT_MAX;
423 21636 int delayed_samples = 0;
424 int i, ret;
425
426 /* calculate the number of delayed samples */
427
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47872 for (i = 0; i < c->nb_streams; i++) {
428 26236 OpusStreamContext *s = &c->streams[i];
429 26236 s->out[0] =
430 26236 s->out[1] = NULL;
431
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26236 delayed_samples = FFMAX(delayed_samples,
432 s->delayed_samples + av_audio_fifo_size(s->sync_buffer));
433 }
434
435 /* decode the header of the first sub-packet to find out the sample count */
436
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21636 if (buf) {
437 21622 OpusPacket *pkt = &c->streams[0].packet;
438 21622 ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
439
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21622 if (ret < 0) {
440 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
441 return ret;
442 }
443 21622 coded_samples += pkt->frame_count * pkt->frame_duration;
444 21622 c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
445 }
446
447 21636 frame->nb_samples = coded_samples + delayed_samples;
448
449 /* no input or buffered data => nothing to do */
450
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21636 if (!frame->nb_samples) {
451 13 *got_frame_ptr = 0;
452 13 return 0;
453 }
454
455 /* setup the data buffers */
456 21623 ret = ff_get_buffer(avctx, frame, 0);
457
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21623 if (ret < 0)
458 return ret;
459 21623 frame->nb_samples = 0;
460
461
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70997 for (i = 0; i < avctx->channels; i++) {
462 49374 ChannelMap *map = &c->channel_maps[i];
463
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49374 if (!map->copy)
464 49374 c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
465 }
466
467 /* read the data from the sync buffers */
468
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47843 for (i = 0; i < c->nb_streams; i++) {
469 26220 OpusStreamContext *s = &c->streams[i];
470 26220 float **out = s->out;
471 26220 int sync_size = av_audio_fifo_size(s->sync_buffer);
472
473 float sync_dummy[32];
474
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26220 int out_dummy = (!out[0]) | ((!out[1]) << 1);
475
476
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26220 if (!out[0])
477 out[0] = sync_dummy;
478
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26220 if (!out[1])
479 3066 out[1] = sync_dummy;
480
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26220 if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
481 return AVERROR_BUG;
482
483 26220 ret = av_audio_fifo_read(s->sync_buffer, (void**)out, sync_size);
484
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26220 if (ret < 0)
485 return ret;
486
487
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26220 if (out_dummy & 1)
488 out[0] = NULL;
489 else
490 26220 out[0] += ret;
491
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26220 if (out_dummy & 2)
492 3066 out[1] = NULL;
493 else
494 23154 out[1] += ret;
495
496 26220 s->out_size = frame->linesize[0] - ret * sizeof(float);
497 }
498
499 /* decode each sub-packet */
500
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47843 for (i = 0; i < c->nb_streams; i++) {
501 26220 OpusStreamContext *s = &c->streams[i];
502
503
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26220 if (i && buf) {
504 4597 ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
505
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4597 if (ret < 0) {
506 av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
507 return ret;
508 }
509
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4597 if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
510 av_log(avctx, AV_LOG_ERROR,
511 "Mismatching coded sample count in substream %d.\n", i);
512 return AVERROR_INVALIDDATA;
513 }
514
515 4597 s->silk_samplerate = get_silk_samplerate(s->packet.config);
516 }
517
518 26220 ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
519 coded_samples);
520
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26220 if (ret < 0)
521 return ret;
522 26220 s->decoded_samples = ret;
523 26220 decoded_samples = FFMIN(decoded_samples, ret);
524
525 26220 buf += s->packet.packet_size;
526 26220 buf_size -= s->packet.packet_size;
527 }
528
529 /* buffer the extra samples */
530
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47843 for (i = 0; i < c->nb_streams; i++) {
531 26220 OpusStreamContext *s = &c->streams[i];
532 26220 int buffer_samples = s->decoded_samples - decoded_samples;
533
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26220 if (buffer_samples) {
534 float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0],
535 s->out[1] ? s->out[1] : (float*)frame->extended_data[0] };
536 buf[0] += decoded_samples;
537 buf[1] += decoded_samples;
538 ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples);
539 if (ret < 0)
540 return ret;
541 }
542 }
543
544
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70997 for (i = 0; i < avctx->channels; i++) {
545 49374 ChannelMap *map = &c->channel_maps[i];
546
547 /* handle copied channels */
548
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49374 if (map->copy) {
549 memcpy(frame->extended_data[i],
550 frame->extended_data[map->copy_idx],
551 frame->linesize[0]);
552
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49374 } else if (map->silence) {
553 memset(frame->extended_data[i], 0, frame->linesize[0]);
554 }
555
556
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49374 if (c->gain_i && decoded_samples > 0) {
557 c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
558 (float*)frame->extended_data[i],
559 c->gain, FFALIGN(decoded_samples, 8));
560 }
561 }
562
563 21623 frame->nb_samples = decoded_samples;
564 21623 *got_frame_ptr = !!decoded_samples;
565
566 21623 return avpkt->size;
567 }
568
569 static av_cold void opus_decode_flush(AVCodecContext *ctx)
570 {
571 OpusContext *c = ctx->priv_data;
572 int i;
573
574 for (i = 0; i < c->nb_streams; i++) {
575 OpusStreamContext *s = &c->streams[i];
576
577 memset(&s->packet, 0, sizeof(s->packet));
578 s->delayed_samples = 0;
579
580 av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
581 swr_close(s->swr);
582
583 av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer));
584
585 ff_silk_flush(s->silk);
586 ff_celt_flush(s->celt);
587 }
588 }
589
590 33 static av_cold int opus_decode_close(AVCodecContext *avctx)
591 {
592 33 OpusContext *c = avctx->priv_data;
593 int i;
594
595
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80 for (i = 0; i < c->nb_streams; i++) {
596 47 OpusStreamContext *s = &c->streams[i];
597
598 47 ff_silk_free(&s->silk);
599 47 ff_celt_free(&s->celt);
600
601 47 av_freep(&s->out_dummy);
602 47 s->out_dummy_allocated_size = 0;
603
604 47 av_audio_fifo_free(s->sync_buffer);
605 47 av_audio_fifo_free(s->celt_delay);
606 47 swr_free(&s->swr);
607 }
608
609 33 av_freep(&c->streams);
610
611 33 c->nb_streams = 0;
612
613 33 av_freep(&c->channel_maps);
614 33 av_freep(&c->fdsp);
615
616 33 return 0;
617 }
618
619 33 static av_cold int opus_decode_init(AVCodecContext *avctx)
620 {
621 33 OpusContext *c = avctx->priv_data;
622 int ret, i, j;
623
624 33 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
625 33 avctx->sample_rate = 48000;
626
627 33 c->fdsp = avpriv_float_dsp_alloc(0);
628
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33 if (!c->fdsp)
629 return AVERROR(ENOMEM);
630
631 /* find out the channel configuration */
632 33 ret = ff_opus_parse_extradata(avctx, c);
633
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33 if (ret < 0)
634 return ret;
635
636 /* allocate and init each independent decoder */
637 33 c->streams = av_calloc(c->nb_streams, sizeof(*c->streams));
638
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33 if (!c->streams) {
639 c->nb_streams = 0;
640 return AVERROR(ENOMEM);
641 }
642
643
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80 for (i = 0; i < c->nb_streams; i++) {
644 47 OpusStreamContext *s = &c->streams[i];
645 uint64_t layout;
646
647
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47 s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
648
649 47 s->avctx = avctx;
650
651
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131 for (j = 0; j < s->output_channels; j++) {
652 84 s->silk_output[j] = s->silk_buf[j];
653 84 s->celt_output[j] = s->celt_buf[j];
654 84 s->redundancy_output[j] = s->redundancy_buf[j];
655 }
656
657 47 s->fdsp = c->fdsp;
658
659 47 s->swr =swr_alloc();
660
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47 if (!s->swr)
661 return AVERROR(ENOMEM);
662
663
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47 layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
664 47 av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
665 47 av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
666 47 av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
667 47 av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
668 47 av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
669 47 av_opt_set_int(s->swr, "filter_size", 16, 0);
670
671 47 ret = ff_silk_init(avctx, &s->silk, s->output_channels);
672
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47 if (ret < 0)
673 return ret;
674
675 47 ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);
676
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47 if (ret < 0)
677 return ret;
678
679 47 s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
680 s->output_channels, 1024);
681
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47 if (!s->celt_delay)
682 return AVERROR(ENOMEM);
683
684 47 s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt,
685 s->output_channels, 32);
686
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47 if (!s->sync_buffer)
687 return AVERROR(ENOMEM);
688 }
689
690 33 return 0;
691 }
692
693 #define OFFSET(x) offsetof(OpusContext, x)
694 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
695 static const AVOption opus_options[] = {
696 { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },
697 { NULL },
698 };
699
700 static const AVClass opus_class = {
701 .class_name = "Opus Decoder",
702 .item_name = av_default_item_name,
703 .option = opus_options,
704 .version = LIBAVUTIL_VERSION_INT,
705 };
706
707 const AVCodec ff_opus_decoder = {
708 .name = "opus",
709 .long_name = NULL_IF_CONFIG_SMALL("Opus"),
710 .priv_class = &opus_class,
711 .type = AVMEDIA_TYPE_AUDIO,
712 .id = AV_CODEC_ID_OPUS,
713 .priv_data_size = sizeof(OpusContext),
714 .init = opus_decode_init,
715 .close = opus_decode_close,
716 .decode = opus_decode_packet,
717 .flush = opus_decode_flush,
718 .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | AV_CODEC_CAP_CHANNEL_CONF,
719 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
720 };
721