FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libswresample/swresample.c
Date: 2026-09-06 08:50:24
Exec Total Coverage
Lines: 479 634 75.6%
Functions: 23 24 95.8%
Branches: 359 547 65.6%

Line Branch Exec Source
1 /*
2 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3 *
4 * This file is part of libswresample
5 *
6 * libswresample is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * libswresample is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with libswresample; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/mem.h"
22 #include "libavutil/opt.h"
23 #include "swresample_internal.h"
24 #include "audioconvert.h"
25 #include "libavutil/avassert.h"
26 #include "libavutil/channel_layout.h"
27 #include "libavutil/internal.h"
28
29 #include <float.h>
30
31 #define ALIGN 32
32
33 6478 int swri_check_chlayout(struct SwrContext *s, const AVChannelLayout *chl, const char *name) {
34 char l1[1024];
35 int ret;
36
37
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6478 if (!(ret = av_channel_layout_check(chl)) || chl->nb_channels > SWR_CH_MAX) {
38 if (ret)
39 av_channel_layout_describe(chl, l1, sizeof(l1));
40 av_log(s, AV_LOG_WARNING, "%s channel layout \"%s\" is invalid or unsupported.\n", name, ret ? l1 : "");
41 return AVERROR(EINVAL);
42 }
43
44 6478 return 0;
45 }
46
47 18 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
48
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18 if(!s || s->in_convert) // s needs to be allocated but not initialized
49 return AVERROR(EINVAL);
50 18 s->channel_map = channel_map;
51 18 return 0;
52 }
53
54 1596 int swr_alloc_set_opts2(struct SwrContext **ps,
55 const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
56 const AVChannelLayout *in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
57 int log_offset, void *log_ctx) {
58 1596 struct SwrContext *s = *ps;
59 int ret;
60
61
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1596 if (!s) s = swr_alloc();
62
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1596 if (!s) return AVERROR(ENOMEM);
63
64 1596 *ps = s;
65
66 1596 s->log_level_offset = log_offset;
67 1596 s->log_ctx = log_ctx;
68
69
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1596 if ((ret = av_opt_set_chlayout(s, "ochl", out_ch_layout, 0)) < 0)
70 goto fail;
71
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1596 if ((ret = swri_check_chlayout(s, out_ch_layout, "ochl")) < 0)
72 goto fail;
73
74
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1596 if ((ret = av_opt_set_int(s, "osf", out_sample_fmt, 0)) < 0)
75 goto fail;
76
77
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1596 if ((ret = av_opt_set_int(s, "osr", out_sample_rate, 0)) < 0)
78 goto fail;
79
80
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1596 if ((ret = av_opt_set_chlayout(s, "ichl", in_ch_layout, 0)) < 0)
81 goto fail;
82
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1596 if ((ret = swri_check_chlayout(s, in_ch_layout, "ichl")) < 0)
83 goto fail;
84
85
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1596 if ((ret = av_opt_set_int(s, "isf", in_sample_fmt, 0)) < 0)
86 goto fail;
87
88
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1596 if ((ret = av_opt_set_int(s, "isr", in_sample_rate, 0)) < 0)
89 goto fail;
90
91 1596 return 0;
92 fail:
93 av_log(s, AV_LOG_ERROR, "Failed to set option\n");
94 swr_free(ps);
95 return ret;
96 }
97
98 6117 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
99 6117 a->fmt = fmt;
100 6117 a->bps = av_get_bytes_per_sample(fmt);
101 6117 a->planar= av_sample_fmt_is_planar(fmt);
102
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6117 if (a->ch_count == 1)
103 4636 a->planar = 1;
104 6117 }
105
106 34016 static void free_temp(AudioData *a){
107 34016 av_free(a->data);
108 34016 memset(a, 0, sizeof(*a));
109 34016 }
110
111 4252 static void clear_context(SwrContext *s){
112 4252 s->in_buffer_index= 0;
113 4252 s->in_buffer_count= 0;
114 4252 s->resample_in_constraint= 0;
115 4252 memset(s->in.ch, 0, sizeof(s->in.ch));
116 4252 memset(s->out.ch, 0, sizeof(s->out.ch));
117 4252 free_temp(&s->postin);
118 4252 free_temp(&s->midbuf);
119 4252 free_temp(&s->preout);
120 4252 free_temp(&s->in_buffer);
121 4252 free_temp(&s->silence);
122 4252 free_temp(&s->drop_temp);
123 4252 free_temp(&s->dither.noise);
124 4252 free_temp(&s->dither.temp);
125 4252 av_channel_layout_uninit(&s->in_ch_layout);
126 4252 av_channel_layout_uninit(&s->out_ch_layout);
127 4252 av_channel_layout_uninit(&s->used_ch_layout);
128 4252 swri_audio_convert_free(&s-> in_convert);
129 4252 swri_audio_convert_free(&s->out_convert);
130 4252 swri_audio_convert_free(&s->full_convert);
131 4252 swri_rematrix_free(s);
132
133 4252 s->delayed_samples_fixup = 0;
134 4252 s->flushed = 0;
135 4252 }
136
137 2639 av_cold void swr_free(SwrContext **ss){
138 2639 SwrContext *s= *ss;
139
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2639 if(s){
140 2602 clear_context(s);
141 2602 av_channel_layout_uninit(&s->user_in_chlayout);
142 2602 av_channel_layout_uninit(&s->user_out_chlayout);
143 2602 av_channel_layout_uninit(&s->user_used_chlayout);
144
145
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2602 if (s->resampler)
146 1619 s->resampler->free(&s->resample);
147 }
148
149 2639 av_freep(ss);
150 2639 }
151
152 17 av_cold void swr_close(SwrContext *s){
153 17 clear_context(s);
154 17 }
155
156 1633 av_cold int swr_init(struct SwrContext *s){
157 int ret;
158 char l1[1024], l2[1024];
159
160 1633 clear_context(s);
161
162
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1633 if((unsigned) s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
163 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
164 return AVERROR(EINVAL);
165 }
166
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1633 if((unsigned) s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
167 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
168 return AVERROR(EINVAL);
169 }
170
171
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1633 if(s-> in_sample_rate <= 0){
172 av_log(s, AV_LOG_ERROR, "Requested input sample rate %d is invalid\n", s->in_sample_rate);
173 return AVERROR(EINVAL);
174 }
175
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1633 if(s->out_sample_rate <= 0){
176 av_log(s, AV_LOG_ERROR, "Requested output sample rate %d is invalid\n", s->out_sample_rate);
177 return AVERROR(EINVAL);
178 }
179
180
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1633 if (s->out_sample_fmt == AV_SAMPLE_FMT_DSD &&
181 !(s->in_sample_fmt == AV_SAMPLE_FMT_DSD &&
182 s->in_sample_rate == s->out_sample_rate &&
183 !(s->flags & SWR_FLAG_RESAMPLE))) {
184 av_log(s, AV_LOG_ERROR, "Conversion to DSD is not supported\n");
185 return AVERROR(EINVAL);
186 }
187
188 1633 s->out.ch_count = s-> user_out_chlayout.nb_channels;
189 1633 s-> in.ch_count = s-> user_in_chlayout.nb_channels;
190
191
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3266 if (swri_check_chlayout(s, &s->user_in_chlayout , "input") ||
192 1633 swri_check_chlayout(s, &s->user_out_chlayout, "output"))
193 return AVERROR(EINVAL);
194
195 1633 ret = av_channel_layout_copy(&s->in_ch_layout, &s->user_in_chlayout);
196 1633 ret |= av_channel_layout_copy(&s->out_ch_layout, &s->user_out_chlayout);
197 1633 ret |= av_channel_layout_copy(&s->used_ch_layout, &s->user_used_chlayout);
198
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1633 if (ret < 0)
199 return ret;
200
201 1633 s->int_sample_fmt= s->user_int_sample_fmt;
202
203 1633 s->dither.method = s->user_dither_method;
204
205
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1633 switch(s->engine){
206 #if CONFIG_LIBSOXR
207 case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
208 #endif
209 1633 case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
210 default:
211 av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
212 return AVERROR(EINVAL);
213 }
214
215
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1633 if (!av_channel_layout_check(&s->used_ch_layout))
216 1615 av_channel_layout_default(&s->used_ch_layout, s->in.ch_count);
217
218
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1633 if (s->used_ch_layout.nb_channels != s->in_ch_layout.nb_channels)
219 8 av_channel_layout_uninit(&s->in_ch_layout);
220
221
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1633 if (s->used_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
222 1 av_channel_layout_default(&s->used_ch_layout, s->used_ch_layout.nb_channels);
223
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1633 if (s->in_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) {
224 26 ret = av_channel_layout_copy(&s->in_ch_layout, &s->used_ch_layout);
225
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26 if (ret < 0)
226 return ret;
227 }
228
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1633 if (s->out_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
229 19 av_channel_layout_default(&s->out_ch_layout, s->out.ch_count);
230
231 1633 s->rematrix = av_channel_layout_compare(&s->out_ch_layout, &s->in_ch_layout) ||
232
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3238 s->rematrix_volume!=1.0 ||
233
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1605 s->rematrix_custom;
234
235
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1633 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
236 // DSD to PCM conversion is done in floating point
237
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992 if( s->in_sample_fmt == AV_SAMPLE_FMT_DSD
238
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5 && s->out_sample_fmt != AV_SAMPLE_FMT_DSD) {
239 5 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
240 // 16bit or less to 16bit or less with the same sample rate
241
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987 } else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
242
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579 && av_get_bytes_per_sample(s->out_sample_fmt) <= 2
243
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491 && s->out_sample_rate==s->in_sample_rate) {
244 490 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
245 // 8 -> 8, 16->8, 8->16bit
246 497 } else if( av_get_bytes_per_sample(s-> in_sample_fmt)
247
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497 +av_get_bytes_per_sample(s->out_sample_fmt) <= 3 ) {
248 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
249
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497 }else if( av_get_bytes_per_sample(s-> in_sample_fmt) <= 2
250
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89 && !s->rematrix
251
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87 && s->out_sample_rate==s->in_sample_rate
252
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86 && !(s->flags & SWR_FLAG_RESAMPLE)){
253 86 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
254
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411 }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
255
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76 && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
256
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42 && !s->rematrix
257
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42 && s->out_sample_rate == s->in_sample_rate
258
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42 && !(s->flags & SWR_FLAG_RESAMPLE)
259
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42 && s->engine != SWR_ENGINE_SOXR){
260 42 s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
261
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369 }else if(av_get_bytes_per_sample(s->in_sample_fmt) <= 4){
262 324 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
263 }else{
264 45 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
265 }
266 }
267 1633 av_log(s, AV_LOG_DEBUG, "Using %s internally between filters\n", av_get_sample_fmt_name(s->int_sample_fmt));
268
269
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1633 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
270
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881 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
271
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694 &&s->int_sample_fmt != AV_SAMPLE_FMT_S64P
272
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694 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
273
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201 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
274 av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, s16p/s32p/s64p/fltp/dblp are supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
275 return AVERROR(EINVAL);
276 }
277
278 1633 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
279 1633 set_audiodata_fmt(&s->out, s->out_sample_fmt);
280
281
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1633 if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
282
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55 if (!s->async && s->min_compensation >= FLT_MAX/2)
283 s->async = 1;
284
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55 if (s->firstpts == AV_NOPTS_VALUE)
285 54 s->firstpts =
286 54 s->outpts = s->firstpts_in_samples * s->out_sample_rate;
287 } else
288 1578 s->firstpts = AV_NOPTS_VALUE;
289
290
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1633 if (s->async) {
291
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54 if (s->min_compensation >= FLT_MAX/2)
292 53 s->min_compensation = 0.001;
293
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54 if (s->async > 1.0001) {
294 54 s->max_soft_compensation = s->async / (double) s->in_sample_rate;
295 }
296 }
297
298
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1633 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
299 685 s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby, s->exact_rational);
300
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685 if (!s->resample) {
301 av_log(s, AV_LOG_ERROR, "Failed to initialize resampler\n");
302 return AVERROR(ENOMEM);
303 }
304 }else
305 948 s->resampler->free(&s->resample);
306
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1633 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
307
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881 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
308
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694 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
309
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201 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
310 && s->resample){
311 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16p/s32p/fltp/dblp\n");
312 ret = AVERROR(EINVAL);
313 goto fail;
314 }
315
316 #define RSC 1 //FIXME finetune
317
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1633 if(!s-> in.ch_count)
318 s-> in.ch_count = s->in_ch_layout.nb_channels;
319
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1633 if (!av_channel_layout_check(&s->used_ch_layout))
320 av_channel_layout_default(&s->used_ch_layout, s->in.ch_count);
321
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1633 if(!s->out.ch_count)
322 s->out.ch_count = s->out_ch_layout.nb_channels;
323
324
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1633 if(!s-> in.ch_count){
325 av_assert0(s->in_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC);
326 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
327 ret = AVERROR(EINVAL);
328 goto fail;
329 }
330
331 1633 av_channel_layout_describe(&s->out_ch_layout, l2, sizeof(l2));
332 1633 av_channel_layout_describe(&s->in_ch_layout, l1, sizeof(l1));
333
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1633 if (s->in_ch_layout.order != AV_CHANNEL_ORDER_UNSPEC && s->used_ch_layout.nb_channels != s->in_ch_layout.nb_channels) {
334 av_log(s, AV_LOG_ERROR, "Input channel layout %s mismatches specified channel count %d\n", l1, s->used_ch_layout.nb_channels);
335 ret = AVERROR(EINVAL);
336 goto fail;
337 }
338
339
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1633 if (( s->out_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC
340
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1633 || s-> in_ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) && s->used_ch_layout.nb_channels != s->out.ch_count && !s->rematrix_custom) {
341 av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
342 "but there is not enough information to do it\n", l1, l2);
343 ret = AVERROR(EINVAL);
344 goto fail;
345 }
346
347
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1633 av_assert0(s->used_ch_layout.nb_channels);
348
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1633 av_assert0(s->out.ch_count);
349 1633 s->resample_first= RSC*s->out.ch_count/s->used_ch_layout.nb_channels - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
350
351 1633 s->in_buffer= s->in;
352 1633 s->silence = s->in;
353 1633 s->drop_temp= s->out;
354
355
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1633 if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
356 goto fail;
357
358
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1633 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
359 912 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
360 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
361 // fall through to the generic path for conversions that have no
362 // direct implementation (e.g. DSD input to non-float output)
363
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912 if (s->full_convert)
364 911 return 0;
365 }
366
367 722 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
368 s-> in_sample_fmt, s->used_ch_layout.nb_channels, s->channel_map, 0);
369 722 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
370 s->int_sample_fmt, s->out.ch_count, NULL, 0);
371
372
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722 if (!s->in_convert || !s->out_convert) {
373 av_log(s, AV_LOG_ERROR, "Cannot convert %s sample format to %s sample format\n",
374 av_get_sample_fmt_name(!s->in_convert ? s->in_sample_fmt : s->int_sample_fmt),
375 av_get_sample_fmt_name(!s->in_convert ? s->int_sample_fmt : s->out_sample_fmt));
376 ret = AVERROR(EINVAL);
377 goto fail;
378 }
379
380 722 s->postin= s->in;
381 722 s->preout= s->out;
382 722 s->midbuf= s->in;
383
384
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722 if(s->channel_map){
385 18 s->postin.ch_count=
386 18 s->midbuf.ch_count= s->used_ch_layout.nb_channels;
387
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18 if(s->resample)
388 s->in_buffer.ch_count= s->used_ch_layout.nb_channels;
389 }
390
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722 if(!s->resample_first){
391 352 s->midbuf.ch_count= s->out.ch_count;
392
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352 if(s->resample)
393 327 s->in_buffer.ch_count = s->out.ch_count;
394 }
395
396 722 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
397 722 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
398 722 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
399
400
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722 if(s->resample){
401 685 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
402 }
403
404
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722 av_assert0(!s->preout.count);
405 722 s->dither.noise = s->preout;
406 722 s->dither.temp = s->preout;
407
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722 if (s->dither.method > SWR_DITHER_NS) {
408 s->dither.noise.bps = 4;
409 s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP;
410 s->dither.noise_scale = 1;
411 }
412
413
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722 if(s->rematrix || s->dither.method) {
414 30 ret = swri_rematrix_init(s);
415
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30 if (ret < 0)
416 goto fail;
417 }
418
419 722 return 0;
420 fail:
421 swr_close(s);
422 return ret;
423
424 }
425
426 100519 int swri_realloc_audio(AudioData *a, int count){
427 int i, countb;
428 AudioData old;
429
430
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100519 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
431 return AVERROR(EINVAL);
432
433
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100519 if(a->count >= count)
434 96453 return 0;
435
436 4066 count*=2;
437
438 4066 countb= FFALIGN(count*a->bps, ALIGN);
439 4066 old= *a;
440
441
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4066 av_assert0(a->bps);
442
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4066 av_assert0(a->ch_count);
443
444 4066 a->data = av_calloc(countb, a->ch_count);
445
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4066 if(!a->data)
446 return AVERROR(ENOMEM);
447
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8654 for(i=0; i<a->ch_count; i++){
448
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4588 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
449
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4588 if(a->count && a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
450 }
451
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4066 if(a->count && !a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
452 4066 av_freep(&old.data);
453 4066 a->count= count;
454
455 4066 return 1;
456 }
457
458 35426 static void copy(AudioData *out, AudioData *in,
459 int count){
460
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35426 av_assert0(out->planar == in->planar);
461
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35426 av_assert0(out->bps == in->bps);
462
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35426 av_assert0(out->ch_count == in->ch_count);
463
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35426 if(out->planar){
464 int ch;
465
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91465 for(ch=0; ch<out->ch_count; ch++)
466 56039 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
467 }else
468 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
469 35426 }
470
471 555385 static void fill_audiodata(AudioData *out, uint8_t *const in_arg [SWR_CH_MAX])
472 {
473 int i;
474
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555385 if(!in_arg){
475 68 memset(out->ch, 0, sizeof(out->ch));
476
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555317 }else if(out->planar){
477
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1016116 for(i=0; i<out->ch_count; i++)
478 614306 out->ch[i]= in_arg[i];
479 }else{
480
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514487 for(i=0; i<out->ch_count; i++)
481 360980 out->ch[i]= in_arg[0] + i*out->bps;
482 }
483 555385 }
484
485 137 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
486 int i;
487
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137 if(out->planar){
488
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274 for(i=0; i<out->ch_count; i++)
489 137 in_arg[i]= out->ch[i];
490 }else{
491 in_arg[0]= out->ch[0];
492 }
493 137 }
494
495 /**
496 *
497 * out may be equal in.
498 */
499 477625 static void buf_set(AudioData *out, AudioData *in, int count){
500 int ch;
501
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477625 if(in->planar){
502
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1247257 for(ch=0; ch<out->ch_count; ch++)
503 780671 out->ch[ch]= in->ch[ch] + count*out->bps;
504 }else{
505
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39741 for(ch=out->ch_count-1; ch>=0; ch--)
506 28702 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
507 }
508 477625 }
509
510 /**
511 *
512 * @return number of samples output per channel
513 */
514 19612 static int resample(SwrContext *s, AudioData *out_param, int out_count,
515 const AudioData * in_param, int in_count){
516 AudioData in, out, tmp;
517 19612 int ret_sum=0;
518 19612 int border=0;
519
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19612 int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
520
521 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
522 av_assert1(s->in_buffer.planar == in_param->planar);
523 av_assert1(s->in_buffer.fmt == in_param->fmt);
524
525 19612 tmp=out=*out_param;
526 19612 in = *in_param;
527
528 19612 border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
529 &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
530
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19612 if (border == INT_MAX) {
531 87 return 0;
532
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19525 } else if (border < 0) {
533 return border;
534
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19525 } else if (border) {
535 683 buf_set(&in, &in, border);
536 683 in_count -= border;
537 683 s->resample_in_constraint = 0;
538 }
539
540 35391 do{
541 int ret, size, consumed;
542
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54916 if(!s->resample_in_constraint && s->in_buffer_count){
543 46904 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
544 46904 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
545 46904 out_count -= ret;
546 46904 ret_sum += ret;
547 46904 buf_set(&out, &out, ret);
548 46904 s->in_buffer_count -= consumed;
549 46904 s->in_buffer_index += consumed;
550
551
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46904 if(!in_count)
552 19520 break;
553
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27384 if(s->in_buffer_count <= border){
554 17803 buf_set(&in, &in, -s->in_buffer_count);
555 17803 in_count += s->in_buffer_count;
556 17803 s->in_buffer_count=0;
557 17803 s->in_buffer_index=0;
558 17803 border = 0;
559 }
560 }
561
562
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35396 if((s->flushed || in_count > padless) && !s->in_buffer_count){
563 17847 s->in_buffer_index=0;
564 17847 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
565 17847 out_count -= ret;
566 17847 ret_sum += ret;
567 17847 buf_set(&out, &out, ret);
568 17847 in_count -= consumed;
569 17847 buf_set(&in, &in, consumed);
570 }
571
572 //TODO is this check sane considering the advanced copy avoidance below
573 35396 size= s->in_buffer_index + s->in_buffer_count + in_count;
574
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35396 if( size > s->in_buffer.count
575
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696 && s->in_buffer_count + in_count <= s->in_buffer_index){
576 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
577 copy(&s->in_buffer, &tmp, s->in_buffer_count);
578 s->in_buffer_index=0;
579 }else
580
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35396 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
581 return ret;
582
583
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35396 if(in_count){
584 35391 int count= in_count;
585
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35391 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
586
587 35391 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
588 35391 copy(&tmp, &in, /*in_*/count);
589 35391 s->in_buffer_count += count;
590 35391 in_count -= count;
591 35391 border += count;
592 35391 buf_set(&in, &in, count);
593 35391 s->resample_in_constraint= 0;
594
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35391 if(s->in_buffer_count != count || in_count)
595 35391 continue;
596
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17843 if (padless) {
597 17843 padless = 0;
598 17843 continue;
599 }
600 }
601 5 break;
602 }while(1);
603
604 19525 s->resample_in_constraint= !!out_count;
605
606 19525 return ret_sum;
607 }
608
609 278307 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
610 AudioData *in , int in_count){
611 AudioData *postin, *midbuf, *preout;
612 int ret/*, in_max*/;
613 AudioData preout_tmp, midbuf_tmp;
614
615
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278307 if(s->full_convert){
616
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257133 av_assert0(!s->resample);
617 257133 swri_audio_convert(s->full_convert, out, in, in_count);
618 257133 return out_count;
619 }
620
621 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
622 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
623
624
1/2
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21174 if((ret=swri_realloc_audio(&s->postin, in_count))<0)
625 return ret;
626
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21174 if(s->resample_first){
627
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14808 av_assert0(s->midbuf.ch_count == s->used_ch_layout.nb_channels);
628
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14808 if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
629 return ret;
630 }else{
631
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6366 av_assert0(s->midbuf.ch_count == s->out.ch_count);
632
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6366 if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
633 return ret;
634 }
635
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✓ Branch 2 taken 21174 times.
21174 if((ret=swri_realloc_audio(&s->preout, out_count))<0)
636 return ret;
637
638 21174 postin= &s->postin;
639
640 21174 midbuf_tmp= s->midbuf;
641 21174 midbuf= &midbuf_tmp;
642 21174 preout_tmp= s->preout;
643 21174 preout= &preout_tmp;
644
645
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21174 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
646 14988 postin= in;
647
648
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21174 if(s->resample_first ? !s->resample : !s->rematrix)
649 5928 midbuf= postin;
650
651
4/4
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21174 if(s->resample_first ? !s->rematrix : !s->resample)
652 15696 preout= midbuf;
653
654
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21174 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
655
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14029 && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
656
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14029 if(preout==in){
657 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
658 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
659 copy(out, in, out_count);
660 return out_count;
661 }
662
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14029 else if(preout==postin) preout= midbuf= postin= out;
663
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14029 else if(preout==midbuf) preout= midbuf= out;
664 1043 else preout= out;
665 }
666
667
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21174 if(in != postin){
668 6186 swri_audio_convert(s->in_convert, postin, in, in_count);
669 }
670
671
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21174 if(s->resample_first){
672
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14808 if(postin != midbuf)
673
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14506 if ((out_count = resample(s, midbuf, out_count, postin, in_count)) < 0)
674 return out_count;
675
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14808 if(midbuf != preout)
676 372 swri_rematrix(s, preout, midbuf, out_count, preout==out);
677 }else{
678
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6366 if(postin != midbuf)
679 740 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
680
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6366 if(midbuf != preout)
681
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5106 if ((out_count = resample(s, preout, out_count, midbuf, in_count)) < 0)
682 return out_count;
683 }
684
685
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21174 if(preout != out && out_count){
686 6815 AudioData *conv_src = preout;
687
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6815 if(s->dither.method){
688 int ch;
689 int dither_count= FFMAX(out_count, 1<<16);
690
691 if (preout == in) {
692 conv_src = &s->dither.temp;
693 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
694 return ret;
695 }
696
697 if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
698 return ret;
699 if(ret)
700 for(ch=0; ch<s->dither.noise.ch_count; ch++)
701 if((ret=swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, (12345678913579ULL*ch + 3141592) % 2718281828U, s->dither.noise.fmt))<0)
702 return ret;
703 av_assert0(s->dither.noise.ch_count == preout->ch_count);
704
705 if(s->dither.noise_pos + out_count > s->dither.noise.count)
706 s->dither.noise_pos = 0;
707
708 if (s->dither.method < SWR_DITHER_NS){
709 if (s->mix_2_1_simd) {
710 int len1= out_count&~15;
711 int off = len1 * preout->bps;
712
713 if(len1)
714 for(ch=0; ch<preout->ch_count; ch++)
715 s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, &s->native_simd_one, 0, 0, len1);
716 if(out_count != len1)
717 for(ch=0; ch<preout->ch_count; ch++)
718 s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off, &s->native_one, 0, 0, out_count - len1);
719 } else {
720 for(ch=0; ch<preout->ch_count; ch++)
721 s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, &s->native_one, 0, 0, out_count);
722 }
723 } else {
724 switch(s->int_sample_fmt) {
725 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
726 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
727 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
728 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
729 }
730 }
731 s->dither.noise_pos += out_count;
732 }
733 //FIXME packed doesn't need more than 1 chan here!
734 6815 swri_audio_convert(s->out_convert, out, conv_src, out_count);
735 }
736 21174 return out_count;
737 }
738
739 315129 int swr_is_initialized(struct SwrContext *s) {
740 315129 return !!s->in_buffer.ch_count;
741 }
742
743 279085 int attribute_align_arg swr_convert(struct SwrContext *s,
744 uint8_t * const *out_arg, int out_count,
745 const uint8_t * const *in_arg, int in_count)
746 {
747 279085 AudioData * in= &s->in;
748 279085 AudioData *out= &s->out;
749
750
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279085 if (!swr_is_initialized(s)) {
751 av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
752 return AVERROR(EINVAL);
753 }
754 #if defined(ASSERT_LEVEL) && ASSERT_LEVEL >1
755 int max_output = swr_get_out_samples(s, in_count);
756 #endif
757
758
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279137 while(s->drop_output > 0){
759 int ret;
760 uint8_t *tmp_arg[SWR_CH_MAX];
761 #define MAX_DROP_STEP 16384
762
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104 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
763 52 return ret;
764
765 104 reversefill_audiodata(&s->drop_temp, tmp_arg);
766 104 s->drop_output *= -1; //FIXME find a less hackish solution
767
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104 ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
768 104 s->drop_output *= -1;
769 104 in_count = 0;
770
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104 if(ret>0) {
771 52 s->drop_output -= ret;
772
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52 if (!s->drop_output && !out_arg)
773 return 0;
774 52 continue;
775 }
776
777
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52 av_assert0(s->drop_output);
778 52 return 0;
779 }
780
781
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279033 if(!in_arg){
782
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1989 if(s->resample){
783
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1297 if (!s->flushed)
784 659 s->resampler->flush(s);
785 1297 s->resample_in_constraint = 0;
786 1297 s->flushed = 1;
787
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692 }else if(!s->in_buffer_count){
788 692 return 0;
789 }
790 }else
791 277044 fill_audiodata(in , (void*)in_arg);
792
793 278341 fill_audiodata(out, out_arg);
794
795
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278341 if(s->resample){
796 19612 int ret = swr_convert_internal(s, out, out_count, in, in_count);
797
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19612 if(ret>0 && !s->drop_output)
798 18817 s->outpts += ret * (int64_t)s->in_sample_rate;
799
800 av_assert2(max_output < 0 || ret <= max_output);
801
802 19612 return ret;
803 }else{
804 258729 AudioData tmp= *in;
805 258729 int ret2=0;
806 int ret, size;
807 258729 size = FFMIN(out_count, s->in_buffer_count);
808
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258729 if(size){
809 125 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
810 125 ret= swr_convert_internal(s, out, size, &tmp, size);
811
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125 if(ret<0)
812 return ret;
813 125 ret2= ret;
814 125 s->in_buffer_count -= ret;
815 125 s->in_buffer_index += ret;
816 125 buf_set(out, out, ret);
817 125 out_count -= ret;
818
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125 if(!s->in_buffer_count)
819 4 s->in_buffer_index = 0;
820 }
821
822
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258729 if(in_count){
823 258604 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
824
825
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258604 if(in_count > out_count) { //FIXME move after swr_convert_internal
826
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35 if( size > s->in_buffer.count
827
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5 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
828 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
829 copy(&s->in_buffer, &tmp, s->in_buffer_count);
830 s->in_buffer_index=0;
831 }else
832
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35 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
833 return ret;
834 }
835
836
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258604 if(out_count){
837 258570 size = FFMIN(in_count, out_count);
838 258570 ret= swr_convert_internal(s, out, size, in, size);
839
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258570 if(ret<0)
840 return ret;
841 258570 buf_set(in, in, ret);
842 258570 in_count -= ret;
843 258570 ret2 += ret;
844 }
845
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258604 if(in_count){
846 35 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
847 35 copy(&tmp, in, in_count);
848 35 s->in_buffer_count += in_count;
849 }
850 }
851
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258729 if(ret2>0 && !s->drop_output)
852 258694 s->outpts += ret2 * (int64_t)s->in_sample_rate;
853 av_assert2(max_output < 0 || ret2 < 0 || ret2 <= max_output);
854 258729 return ret2;
855 }
856 }
857
858 52 int swr_drop_output(struct SwrContext *s, int count){
859 const uint8_t *tmp_arg[SWR_CH_MAX];
860 52 s->drop_output += count;
861
862
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52 if(s->drop_output <= 0)
863 return 0;
864
865 52 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
866 52 return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
867 }
868
869 33 int swr_inject_silence(struct SwrContext *s, int count){
870 int ret, i;
871 uint8_t *tmp_arg[SWR_CH_MAX];
872
873
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33 if(count <= 0)
874 return 0;
875
876 #define MAX_SILENCE_STEP 16384
877
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62 while (count > MAX_SILENCE_STEP) {
878
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29 if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
879 return ret;
880 29 count -= MAX_SILENCE_STEP;
881 }
882
883
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33 if((ret=swri_realloc_audio(&s->silence, count))<0)
884 return ret;
885
886 {
887
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66 int fill = s->silence.fmt == AV_SAMPLE_FMT_DSD ? 0x69 :
888
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33 s->silence.bps == 1 ? 0x80 : 0;
889
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66 if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
890 33 memset(s->silence.ch[i], fill, count*s->silence.bps);
891 } else
892 memset(s->silence.ch[0], fill, count*s->silence.bps*s->silence.ch_count);
893 }
894
895 33 reversefill_audiodata(&s->silence, tmp_arg);
896 33 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
897 33 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
898 33 return ret;
899 }
900
901 531930 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
902
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531930 if (s->resampler && s->resample){
903 17294 return s->resampler->get_delay(s, base);
904 }else{
905 514636 return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
906 }
907 }
908
909 int swr_get_out_samples(struct SwrContext *s, int in_samples)
910 {
911 int64_t out_samples;
912
913 if (in_samples < 0)
914 return AVERROR(EINVAL);
915
916 if (s->resampler && s->resample) {
917 if (!s->resampler->get_out_samples)
918 return AVERROR(ENOSYS);
919 out_samples = s->resampler->get_out_samples(s, in_samples);
920 } else {
921 out_samples = s->in_buffer_count + in_samples;
922 av_assert0(s->out_sample_rate == s->in_sample_rate);
923 }
924
925 if (out_samples > INT_MAX)
926 return AVERROR(EINVAL);
927
928 return out_samples;
929 }
930
931 127 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
932 int ret;
933
934
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127 if (!s || compensation_distance < 0)
935 return AVERROR(EINVAL);
936
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127 if (!compensation_distance && sample_delta)
937 return AVERROR(EINVAL);
938
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127 if (!s->resample) {
939 1 s->flags |= SWR_FLAG_RESAMPLE;
940 1 ret = swr_init(s);
941
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1 if (ret < 0)
942 return ret;
943 }
944
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127 if (!s->resampler->set_compensation){
945 return AVERROR(EINVAL);
946 }else{
947 127 return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
948 }
949 }
950
951 268064 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
952
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268064 if(pts == INT64_MIN)
953 2099 return s->outpts;
954
955
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265965 if (s->firstpts == AV_NOPTS_VALUE)
956 1512 s->outpts = s->firstpts = pts;
957
958
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265965 if(s->min_compensation >= FLT_MAX) {
959 264734 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
960 } else {
961 1231 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
962 1231 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
963
964
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1231 if(fabs(fdelta) > s->min_compensation) {
965
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239 if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
966 int ret;
967
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56 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
968 52 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
969
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56 if(ret<0){
970 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
971 }
972
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127 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
973 127 int duration = s->out_sample_rate * s->soft_compensation_duration;
974
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127 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
975 127 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
976 127 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
977 127 swr_set_compensation(s, comp, duration);
978 }
979 }
980
981 1231 return s->outpts;
982 }
983 }
984