Directory: | ../../../ffmpeg/ |
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File: | src/libavcodec/opusdec.c |
Date: | 2022-07-07 01:21:54 |
Exec | Total | Coverage | |
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Lines: | 282 | 391 | 72.1% |
Branches: | 155 | 234 | 66.2% |
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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 "codec_internal.h" | ||
47 | #include "get_bits.h" | ||
48 | #include "internal.h" | ||
49 | #include "mathops.h" | ||
50 | #include "opus.h" | ||
51 | #include "opustab.h" | ||
52 | #include "opus_celt.h" | ||
53 | |||
54 | static const uint16_t silk_frame_duration_ms[16] = { | ||
55 | 10, 20, 40, 60, | ||
56 | 10, 20, 40, 60, | ||
57 | 10, 20, 40, 60, | ||
58 | 10, 20, | ||
59 | 10, 20, | ||
60 | }; | ||
61 | |||
62 | /* number of samples of silence to feed to the resampler | ||
63 | * at the beginning */ | ||
64 | static const int silk_resample_delay[] = { | ||
65 | 4, 8, 11, 11, 11 | ||
66 | }; | ||
67 | |||
68 | 26220 | static int get_silk_samplerate(int config) | |
69 | { | ||
70 |
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26220 | if (config < 4) |
71 | 1551 | return 8000; | |
72 |
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24669 | else if (config < 8) |
73 | 1223 | return 12000; | |
74 | 23446 | return 16000; | |
75 | } | ||
76 | |||
77 | 68 | static void opus_fade(float *out, | |
78 | const float *in1, const float *in2, | ||
79 | const float *window, int len) | ||
80 | { | ||
81 | int i; | ||
82 |
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7268 | for (i = 0; i < len; i++) |
83 | 7200 | out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]); | |
84 | 68 | } | |
85 | |||
86 | 17 | static int opus_flush_resample(OpusStreamContext *s, int nb_samples) | |
87 | { | ||
88 | 17 | int celt_size = av_audio_fifo_size(s->celt_delay); | |
89 | int ret, i; | ||
90 | 17 | ret = swr_convert(s->swr, | |
91 | 17 | (uint8_t**)s->cur_out, nb_samples, | |
92 | NULL, 0); | ||
93 |
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17 | if (ret < 0) |
94 | ✗ | return ret; | |
95 |
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17 | else if (ret != nb_samples) { |
96 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n", | |
97 | ret); | ||
98 | ✗ | return AVERROR_BUG; | |
99 | } | ||
100 | |||
101 |
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17 | if (celt_size) { |
102 | ✗ | if (celt_size != nb_samples) { | |
103 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n"); | |
104 | ✗ | return AVERROR_BUG; | |
105 | } | ||
106 | ✗ | av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples); | |
107 | ✗ | for (i = 0; i < s->output_channels; i++) { | |
108 | ✗ | s->fdsp->vector_fmac_scalar(s->cur_out[i], | |
109 | ✗ | s->celt_output[i], 1.0, | |
110 | nb_samples); | ||
111 | } | ||
112 | } | ||
113 | |||
114 |
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17 | if (s->redundancy_idx) { |
115 |
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12 | for (i = 0; i < s->output_channels; i++) |
116 | 8 | opus_fade(s->cur_out[i], s->cur_out[i], | |
117 | 8 | s->redundancy_output[i] + 120 + s->redundancy_idx, | |
118 | 8 | ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx); | |
119 | 4 | s->redundancy_idx = 0; | |
120 | } | ||
121 | |||
122 | 17 | s->cur_out[0] += nb_samples; | |
123 | 17 | s->cur_out[1] += nb_samples; | |
124 | 17 | s->remaining_out_size -= nb_samples * sizeof(float); | |
125 | |||
126 | 17 | return 0; | |
127 | } | ||
128 | |||
129 | 33 | static int opus_init_resample(OpusStreamContext *s) | |
130 | { | ||
131 | static const float delay[16] = { 0.0 }; | ||
132 | 33 | const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay }; | |
133 | int ret; | ||
134 | |||
135 | 33 | av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0); | |
136 | 33 | ret = swr_init(s->swr); | |
137 |
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33 | if (ret < 0) { |
138 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n"); | |
139 | ✗ | return ret; | |
140 | } | ||
141 | |||
142 | 33 | ret = swr_convert(s->swr, | |
143 | NULL, 0, | ||
144 | 33 | delayptr, silk_resample_delay[s->packet.bandwidth]); | |
145 |
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33 | if (ret < 0) { |
146 | ✗ | av_log(s->avctx, AV_LOG_ERROR, | |
147 | "Error feeding initial silence to the resampler.\n"); | ||
148 | ✗ | return ret; | |
149 | } | ||
150 | |||
151 | 33 | return 0; | |
152 | } | ||
153 | |||
154 | 30 | static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size) | |
155 | { | ||
156 | 30 | int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size); | |
157 |
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30 | if (ret < 0) |
158 | ✗ | goto fail; | |
159 | 30 | ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size); | |
160 | |||
161 | 30 | ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc, | |
162 | 30 | s->redundancy_output, | |
163 | 30 | s->packet.stereo + 1, 240, | |
164 | 30 | 0, ff_celt_band_end[s->packet.bandwidth]); | |
165 |
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30 | if (ret < 0) |
166 | ✗ | goto fail; | |
167 | |||
168 | 30 | return 0; | |
169 | ✗ | fail: | |
170 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n"); | |
171 | ✗ | return ret; | |
172 | } | ||
173 | |||
174 | 34394 | static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size) | |
175 | { | ||
176 | 34394 | int samples = s->packet.frame_duration; | |
177 | 34394 | int redundancy = 0; | |
178 | int redundancy_size, redundancy_pos; | ||
179 | int ret, i, consumed; | ||
180 | 34394 | int delayed_samples = s->delayed_samples; | |
181 | |||
182 | 34394 | ret = ff_opus_rc_dec_init(&s->rc, data, size); | |
183 |
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34394 | if (ret < 0) |
184 | ✗ | return ret; | |
185 | |||
186 | /* decode the silk frame */ | ||
187 |
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34394 | if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) { |
188 |
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9472 | if (!swr_is_initialized(s->swr)) { |
189 | 33 | ret = opus_init_resample(s); | |
190 |
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33 | if (ret < 0) |
191 | ✗ | return ret; | |
192 | } | ||
193 | |||
194 | 9472 | samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output, | |
195 | 9472 | FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND), | |
196 | 9472 | s->packet.stereo + 1, | |
197 | 9472 | silk_frame_duration_ms[s->packet.config]); | |
198 |
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9472 | if (samples < 0) { |
199 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n"); | |
200 | ✗ | return samples; | |
201 | } | ||
202 | 9472 | samples = swr_convert(s->swr, | |
203 | 9472 | (uint8_t**)s->cur_out, s->packet.frame_duration, | |
204 | 9472 | (const uint8_t**)s->silk_output, samples); | |
205 |
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9472 | if (samples < 0) { |
206 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n"); | |
207 | ✗ | return samples; | |
208 | } | ||
209 | av_assert2((samples & 7) == 0); | ||
210 | 9472 | s->delayed_samples += s->packet.frame_duration - samples; | |
211 | } else | ||
212 | 24922 | ff_silk_flush(s->silk); | |
213 | |||
214 | // decode redundancy information | ||
215 | 34394 | consumed = opus_rc_tell(&s->rc); | |
216 |
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34394 | if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8) |
217 | 4904 | redundancy = ff_opus_rc_dec_log(&s->rc, 12); | |
218 |
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29490 | else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8) |
219 | 14 | redundancy = 1; | |
220 | |||
221 |
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34394 | if (redundancy) { |
222 | 30 | redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1); | |
223 | |||
224 |
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30 | if (s->packet.mode == OPUS_MODE_HYBRID) |
225 | 16 | redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2; | |
226 | else | ||
227 | 14 | redundancy_size = size - (consumed + 7) / 8; | |
228 | 30 | size -= redundancy_size; | |
229 |
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30 | if (size < 0) { |
230 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n"); | |
231 | ✗ | return AVERROR_INVALIDDATA; | |
232 | } | ||
233 | |||
234 |
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30 | if (redundancy_pos) { |
235 | 14 | ret = opus_decode_redundancy(s, data + size, redundancy_size); | |
236 |
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14 | if (ret < 0) |
237 | ✗ | return ret; | |
238 | 14 | ff_celt_flush(s->celt); | |
239 | } | ||
240 | } | ||
241 | |||
242 | /* decode the CELT frame */ | ||
243 |
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64254 | if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) { |
244 | 29860 | float *out_tmp[2] = { s->cur_out[0], s->cur_out[1] }; | |
245 | 59720 | float **dst = (s->packet.mode == OPUS_MODE_CELT) ? | |
246 |
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29860 | out_tmp : s->celt_output; |
247 | 29860 | int celt_output_samples = samples; | |
248 | 29860 | int delay_samples = av_audio_fifo_size(s->celt_delay); | |
249 | |||
250 |
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29860 | if (delay_samples) { |
251 | ✗ | if (s->packet.mode == OPUS_MODE_HYBRID) { | |
252 | ✗ | av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples); | |
253 | |||
254 | ✗ | for (i = 0; i < s->output_channels; i++) { | |
255 | ✗ | s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0, | |
256 | delay_samples); | ||
257 | ✗ | out_tmp[i] += delay_samples; | |
258 | } | ||
259 | ✗ | celt_output_samples -= delay_samples; | |
260 | } else { | ||
261 | ✗ | av_log(s->avctx, AV_LOG_WARNING, | |
262 | "Spurious CELT delay samples present.\n"); | ||
263 | ✗ | av_audio_fifo_drain(s->celt_delay, delay_samples); | |
264 | ✗ | if (s->avctx->err_recognition & AV_EF_EXPLODE) | |
265 | ✗ | return AVERROR_BUG; | |
266 | } | ||
267 | } | ||
268 | |||
269 | 29860 | ff_opus_rc_dec_raw_init(&s->rc, data + size, size); | |
270 | |||
271 | 29860 | ret = ff_celt_decode_frame(s->celt, &s->rc, dst, | |
272 | 29860 | s->packet.stereo + 1, | |
273 | s->packet.frame_duration, | ||
274 | 29860 | (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0, | |
275 |
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29860 | ff_celt_band_end[s->packet.bandwidth]); |
276 |
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29860 | if (ret < 0) |
277 | ✗ | return ret; | |
278 | |||
279 |
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29860 | if (s->packet.mode == OPUS_MODE_HYBRID) { |
280 | 4938 | int celt_delay = s->packet.frame_duration - celt_output_samples; | |
281 | 4938 | void *delaybuf[2] = { s->celt_output[0] + celt_output_samples, | |
282 | 4938 | s->celt_output[1] + celt_output_samples }; | |
283 | |||
284 |
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14814 | for (i = 0; i < s->output_channels; i++) { |
285 | 9876 | s->fdsp->vector_fmac_scalar(out_tmp[i], | |
286 | 9876 | s->celt_output[i], 1.0, | |
287 | celt_output_samples); | ||
288 | } | ||
289 | |||
290 | 4938 | ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay); | |
291 |
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4938 | if (ret < 0) |
292 | ✗ | return ret; | |
293 | } | ||
294 | } else | ||
295 | 4534 | ff_celt_flush(s->celt); | |
296 | |||
297 |
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34394 | if (s->redundancy_idx) { |
298 | ✗ | for (i = 0; i < s->output_channels; i++) | |
299 | ✗ | opus_fade(s->cur_out[i], s->cur_out[i], | |
300 | ✗ | s->redundancy_output[i] + 120 + s->redundancy_idx, | |
301 | ✗ | ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx); | |
302 | ✗ | s->redundancy_idx = 0; | |
303 | } | ||
304 |
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34394 | if (redundancy) { |
305 |
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30 | if (!redundancy_pos) { |
306 | 16 | ff_celt_flush(s->celt); | |
307 | 16 | ret = opus_decode_redundancy(s, data + size, redundancy_size); | |
308 |
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16 | if (ret < 0) |
309 | ✗ | return ret; | |
310 | |||
311 |
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48 | for (i = 0; i < s->output_channels; i++) { |
312 | 32 | opus_fade(s->cur_out[i] + samples - 120 + delayed_samples, | |
313 | 32 | s->cur_out[i] + samples - 120 + delayed_samples, | |
314 | 32 | s->redundancy_output[i] + 120, | |
315 | ff_celt_window2, 120 - delayed_samples); | ||
316 |
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32 | if (delayed_samples) |
317 | 8 | s->redundancy_idx = 120 - delayed_samples; | |
318 | } | ||
319 | } else { | ||
320 |
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42 | for (i = 0; i < s->output_channels; i++) { |
321 | 28 | memcpy(s->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float)); | |
322 | 28 | opus_fade(s->cur_out[i] + 120 + delayed_samples, | |
323 | 28 | s->redundancy_output[i] + 120, | |
324 | 28 | s->cur_out[i] + 120 + delayed_samples, | |
325 | ff_celt_window2, 120); | ||
326 | } | ||
327 | } | ||
328 | } | ||
329 | |||
330 | 34394 | return samples; | |
331 | } | ||
332 | |||
333 | 26221 | static int opus_decode_subpacket(OpusStreamContext *s, | |
334 | const uint8_t *buf, int buf_size, | ||
335 | int nb_samples) | ||
336 | { | ||
337 | 26221 | int output_samples = 0; | |
338 | 26221 | int flush_needed = 0; | |
339 | int i, j, ret; | ||
340 | |||
341 | 26221 | s->cur_out[0] = s->out[0]; | |
342 | 26221 | s->cur_out[1] = s->out[1]; | |
343 | 26221 | s->remaining_out_size = s->out_size; | |
344 | |||
345 | /* check if we need to flush the resampler */ | ||
346 |
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26221 | if (swr_is_initialized(s->swr)) { |
347 |
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9012 | if (buf) { |
348 | int64_t cur_samplerate; | ||
349 | 9011 | av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate); | |
350 |
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9011 | flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate); |
351 | } else { | ||
352 | 1 | flush_needed = !!s->delayed_samples; | |
353 | } | ||
354 | } | ||
355 | |||
356 |
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26221 | if (!buf && !flush_needed) |
357 | ✗ | return 0; | |
358 | |||
359 | /* use dummy output buffers if the channel is not mapped to anything */ | ||
360 |
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26221 | if (!s->cur_out[0] || |
361 |
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26221 | (s->output_channels == 2 && !s->cur_out[1])) { |
362 | ✗ | av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, | |
363 | ✗ | s->remaining_out_size); | |
364 | ✗ | if (!s->out_dummy) | |
365 | ✗ | return AVERROR(ENOMEM); | |
366 | ✗ | if (!s->cur_out[0]) | |
367 | ✗ | s->cur_out[0] = s->out_dummy; | |
368 | ✗ | if (!s->cur_out[1]) | |
369 | ✗ | s->cur_out[1] = s->out_dummy; | |
370 | } | ||
371 | |||
372 | /* flush the resampler if necessary */ | ||
373 |
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26221 | if (flush_needed) { |
374 | 17 | ret = opus_flush_resample(s, s->delayed_samples); | |
375 |
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17 | if (ret < 0) { |
376 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n"); | |
377 | ✗ | return ret; | |
378 | } | ||
379 | 17 | swr_close(s->swr); | |
380 | 17 | output_samples += s->delayed_samples; | |
381 | 17 | s->delayed_samples = 0; | |
382 | |||
383 |
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17 | if (!buf) |
384 | 1 | goto finish; | |
385 | } | ||
386 | |||
387 | /* decode all the frames in the packet */ | ||
388 |
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60614 | for (i = 0; i < s->packet.frame_count; i++) { |
389 | 34394 | int size = s->packet.frame_size[i]; | |
390 | 34394 | int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size); | |
391 | |||
392 |
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34394 | if (samples < 0) { |
393 | ✗ | av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n"); | |
394 | ✗ | if (s->avctx->err_recognition & AV_EF_EXPLODE) | |
395 | ✗ | return samples; | |
396 | |||
397 | ✗ | for (j = 0; j < s->output_channels; j++) | |
398 | ✗ | memset(s->cur_out[j], 0, s->packet.frame_duration * sizeof(float)); | |
399 | ✗ | samples = s->packet.frame_duration; | |
400 | } | ||
401 | 34394 | output_samples += samples; | |
402 | |||
403 |
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100115 | for (j = 0; j < s->output_channels; j++) |
404 | 65721 | s->cur_out[j] += samples; | |
405 | 34394 | s->remaining_out_size -= samples * sizeof(float); | |
406 | } | ||
407 | |||
408 | 26220 | finish: | |
409 | 26221 | s->cur_out[0] = s->cur_out[1] = NULL; | |
410 | 26221 | s->remaining_out_size = 0; | |
411 | |||
412 | 26221 | return output_samples; | |
413 | } | ||
414 | |||
415 | 21637 | static int opus_decode_packet(AVCodecContext *avctx, AVFrame *frame, | |
416 | int *got_frame_ptr, AVPacket *avpkt) | ||
417 | { | ||
418 | 21637 | OpusContext *c = avctx->priv_data; | |
419 | 21637 | const uint8_t *buf = avpkt->data; | |
420 | 21637 | int buf_size = avpkt->size; | |
421 | 21637 | int coded_samples = 0; | |
422 | 21637 | int decoded_samples = INT_MAX; | |
423 | 21637 | int delayed_samples = 0; | |
424 | int i, ret; | ||
425 | |||
426 | /* calculate the number of delayed samples */ | ||
427 |
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47874 | for (i = 0; i < c->nb_streams; i++) { |
428 | 26237 | OpusStreamContext *s = &c->streams[i]; | |
429 | 26237 | s->out[0] = | |
430 | 26237 | s->out[1] = NULL; | |
431 |
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26237 | 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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21637 | if (buf) { |
437 | 21623 | OpusPacket *pkt = &c->streams[0].packet; | |
438 | 21623 | ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1); | |
439 |
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21623 | if (ret < 0) { |
440 | ✗ | av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n"); | |
441 | ✗ | return ret; | |
442 | } | ||
443 | 21623 | coded_samples += pkt->frame_count * pkt->frame_duration; | |
444 | 21623 | c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config); | |
445 | } | ||
446 | |||
447 | 21637 | frame->nb_samples = coded_samples + delayed_samples; | |
448 | |||
449 | /* no input or buffered data => nothing to do */ | ||
450 |
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21637 | if (!frame->nb_samples) { |
451 | 13 | *got_frame_ptr = 0; | |
452 | 13 | return 0; | |
453 | } | ||
454 | |||
455 | /* setup the data buffers */ | ||
456 | 21624 | ret = ff_get_buffer(avctx, frame, 0); | |
457 |
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21624 | if (ret < 0) |
458 | ✗ | return ret; | |
459 | 21624 | frame->nb_samples = 0; | |
460 | |||
461 |
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70999 | for (i = 0; i < avctx->ch_layout.nb_channels; i++) { |
462 | 49375 | ChannelMap *map = &c->channel_maps[i]; | |
463 |
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49375 | if (!map->copy) |
464 | 49375 | 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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47845 | for (i = 0; i < c->nb_streams; i++) { |
469 | 26221 | OpusStreamContext *s = &c->streams[i]; | |
470 | 26221 | float **out = s->out; | |
471 | 26221 | int sync_size = av_audio_fifo_size(s->sync_buffer); | |
472 | |||
473 | float sync_dummy[32]; | ||
474 |
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26221 | int out_dummy = (!out[0]) | ((!out[1]) << 1); |
475 | |||
476 |
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26221 | if (!out[0]) |
477 | ✗ | out[0] = sync_dummy; | |
478 |
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26221 | if (!out[1]) |
479 | 3067 | out[1] = sync_dummy; | |
480 |
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26221 | if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy)) |
481 | ✗ | return AVERROR_BUG; | |
482 | |||
483 | 26221 | ret = av_audio_fifo_read(s->sync_buffer, (void**)out, sync_size); | |
484 |
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26221 | if (ret < 0) |
485 | ✗ | return ret; | |
486 | |||
487 |
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26221 | if (out_dummy & 1) |
488 | ✗ | out[0] = NULL; | |
489 | else | ||
490 | 26221 | out[0] += ret; | |
491 |
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26221 | if (out_dummy & 2) |
492 | 3067 | out[1] = NULL; | |
493 | else | ||
494 | 23154 | out[1] += ret; | |
495 | |||
496 | 26221 | s->out_size = frame->linesize[0] - ret * sizeof(float); | |
497 | } | ||
498 | |||
499 | /* decode each sub-packet */ | ||
500 |
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47845 | for (i = 0; i < c->nb_streams; i++) { |
501 | 26221 | OpusStreamContext *s = &c->streams[i]; | |
502 | |||
503 |
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26221 | 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 | 26221 | ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size, | |
519 | coded_samples); | ||
520 |
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26221 | if (ret < 0) |
521 | ✗ | return ret; | |
522 | 26221 | s->decoded_samples = ret; | |
523 | 26221 | decoded_samples = FFMIN(decoded_samples, ret); | |
524 | |||
525 | 26221 | buf += s->packet.packet_size; | |
526 | 26221 | buf_size -= s->packet.packet_size; | |
527 | } | ||
528 | |||
529 | /* buffer the extra samples */ | ||
530 |
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47845 | for (i = 0; i < c->nb_streams; i++) { |
531 | 26221 | OpusStreamContext *s = &c->streams[i]; | |
532 | 26221 | int buffer_samples = s->decoded_samples - decoded_samples; | |
533 |
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26221 | 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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70999 | for (i = 0; i < avctx->ch_layout.nb_channels; i++) { |
545 | 49375 | ChannelMap *map = &c->channel_maps[i]; | |
546 | |||
547 | /* handle copied channels */ | ||
548 |
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49375 | 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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49375 | } else if (map->silence) { |
553 | ✗ | memset(frame->extended_data[i], 0, frame->linesize[0]); | |
554 | } | ||
555 | |||
556 |
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49375 | 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 | 21624 | frame->nb_samples = decoded_samples; | |
564 | 21624 | *got_frame_ptr = !!decoded_samples; | |
565 | |||
566 | 21624 | 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 | 34 | static av_cold int opus_decode_close(AVCodecContext *avctx) | |
591 | { | ||
592 | 34 | OpusContext *c = avctx->priv_data; | |
593 | int i; | ||
594 | |||
595 |
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82 | for (i = 0; i < c->nb_streams; i++) { |
596 | 48 | OpusStreamContext *s = &c->streams[i]; | |
597 | |||
598 | 48 | ff_silk_free(&s->silk); | |
599 | 48 | ff_celt_free(&s->celt); | |
600 | |||
601 | 48 | av_freep(&s->out_dummy); | |
602 | 48 | s->out_dummy_allocated_size = 0; | |
603 | |||
604 | 48 | av_audio_fifo_free(s->sync_buffer); | |
605 | 48 | av_audio_fifo_free(s->celt_delay); | |
606 | 48 | swr_free(&s->swr); | |
607 | } | ||
608 | |||
609 | 34 | av_freep(&c->streams); | |
610 | |||
611 | 34 | c->nb_streams = 0; | |
612 | |||
613 | 34 | av_freep(&c->channel_maps); | |
614 | 34 | av_freep(&c->fdsp); | |
615 | |||
616 | 34 | return 0; | |
617 | } | ||
618 | |||
619 | 34 | static av_cold int opus_decode_init(AVCodecContext *avctx) | |
620 | { | ||
621 | 34 | OpusContext *c = avctx->priv_data; | |
622 | int ret, i, j; | ||
623 | |||
624 | 34 | avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; | |
625 | 34 | avctx->sample_rate = 48000; | |
626 | |||
627 | 34 | c->fdsp = avpriv_float_dsp_alloc(0); | |
628 |
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34 | if (!c->fdsp) |
629 | ✗ | return AVERROR(ENOMEM); | |
630 | |||
631 | /* find out the channel configuration */ | ||
632 | 34 | ret = ff_opus_parse_extradata(avctx, c); | |
633 |
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34 | if (ret < 0) |
634 | ✗ | return ret; | |
635 | |||
636 | /* allocate and init each independent decoder */ | ||
637 | 34 | c->streams = av_calloc(c->nb_streams, sizeof(*c->streams)); | |
638 |
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34 | if (!c->streams) { |
639 | ✗ | c->nb_streams = 0; | |
640 | ✗ | return AVERROR(ENOMEM); | |
641 | } | ||
642 | |||
643 |
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82 | for (i = 0; i < c->nb_streams; i++) { |
644 | 48 | OpusStreamContext *s = &c->streams[i]; | |
645 | uint64_t layout; | ||
646 | |||
647 |
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48 | s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1; |
648 | |||
649 | 48 | s->avctx = avctx; | |
650 | |||
651 |
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133 | for (j = 0; j < s->output_channels; j++) { |
652 | 85 | s->silk_output[j] = s->silk_buf[j]; | |
653 | 85 | s->celt_output[j] = s->celt_buf[j]; | |
654 | 85 | s->redundancy_output[j] = s->redundancy_buf[j]; | |
655 | } | ||
656 | |||
657 | 48 | s->fdsp = c->fdsp; | |
658 | |||
659 | 48 | s->swr =swr_alloc(); | |
660 |
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48 | if (!s->swr) |
661 | ✗ | return AVERROR(ENOMEM); | |
662 | |||
663 |
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48 | layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; |
664 | 48 | av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0); | |
665 | 48 | av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0); | |
666 | 48 | av_opt_set_int(s->swr, "in_channel_layout", layout, 0); | |
667 | 48 | av_opt_set_int(s->swr, "out_channel_layout", layout, 0); | |
668 | 48 | av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0); | |
669 | 48 | av_opt_set_int(s->swr, "filter_size", 16, 0); | |
670 | |||
671 | 48 | ret = ff_silk_init(avctx, &s->silk, s->output_channels); | |
672 |
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48 | if (ret < 0) |
673 | ✗ | return ret; | |
674 | |||
675 | 48 | ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv); | |
676 |
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48 | if (ret < 0) |
677 | ✗ | return ret; | |
678 | |||
679 | 48 | s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt, | |
680 | s->output_channels, 1024); | ||
681 |
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48 | if (!s->celt_delay) |
682 | ✗ | return AVERROR(ENOMEM); | |
683 | |||
684 | 48 | s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt, | |
685 | s->output_channels, 32); | ||
686 |
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48 | if (!s->sync_buffer) |
687 | ✗ | return AVERROR(ENOMEM); | |
688 | } | ||
689 | |||
690 | 34 | 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 FFCodec ff_opus_decoder = { | ||
708 | .p.name = "opus", | ||
709 | .p.long_name = NULL_IF_CONFIG_SMALL("Opus"), | ||
710 | .p.priv_class = &opus_class, | ||
711 | .p.type = AVMEDIA_TYPE_AUDIO, | ||
712 | .p.id = AV_CODEC_ID_OPUS, | ||
713 | .priv_data_size = sizeof(OpusContext), | ||
714 | .init = opus_decode_init, | ||
715 | .close = opus_decode_close, | ||
716 | FF_CODEC_DECODE_CB(opus_decode_packet), | ||
717 | .flush = opus_decode_flush, | ||
718 | .p.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 |