FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_aresample.c
Date: 2026-07-16 17:05:34
Exec Total Coverage
Lines: 187 223 83.9%
Functions: 8 9 88.9%
Branches: 68 102 66.7%
Line Branch Exec Source
1 /*
2 * Copyright (c) 2011 Stefano Sabatini
3 * Copyright (c) 2011 Mina Nagy Zaki
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * resampling audio filter
25 */
26
27 #include "libavutil/avstring.h"
28 #include "libavutil/channel_layout.h"
29 #include "libavutil/downmix_info.h"
30 #include "libavutil/opt.h"
31 #include "libavutil/samplefmt.h"
32 #include "libavutil/avassert.h"
33 #include "libswresample/swresample.h"
34 #include "avfilter.h"
35 #include "audio.h"
36 #include "filters.h"
37 #include "formats.h"
38
39 typedef struct AResampleContext {
40 const AVClass *class;
41 int sample_rate_arg;
42 double ratio;
43 struct SwrContext *swr;
44 int64_t next_pts;
45 int more_data;
46 } AResampleContext;
47
48 2399 static av_cold int preinit(AVFilterContext *ctx)
49 {
50 2399 AResampleContext *aresample = ctx->priv;
51
52 2399 aresample->next_pts = AV_NOPTS_VALUE;
53 2399 aresample->swr = swr_alloc();
54
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 2399 if (!aresample->swr)
55 ✗ return AVERROR(ENOMEM);
56
57 2399 return 0;
58 }
59
60 2399 static av_cold void uninit(AVFilterContext *ctx)
61 {
62 2399 AResampleContext *aresample = ctx->priv;
63 2399 swr_free(&aresample->swr);
64 2399 }
65
66 1534 static int query_formats(const AVFilterContext *ctx,
67 AVFilterFormatsConfig **cfg_in,
68 AVFilterFormatsConfig **cfg_out)
69 {
70 1534 const AResampleContext *aresample = ctx->priv;
71 enum AVSampleFormat out_format;
72 1534 AVChannelLayout out_layout = { 0 };
73 int64_t out_rate;
74
75 AVFilterFormats *in_formats, *out_formats;
76 AVFilterFormats *in_samplerates, *out_samplerates;
77 AVFilterChannelLayouts *in_layouts, *out_layouts;
78 int ret;
79
80
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 1534 if (aresample->sample_rate_arg > 0)
81 640 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
82 1534 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
83 1534 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
84
85 1534 in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
86
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 1534 if ((ret = ff_formats_ref(in_formats, &cfg_in[0]->formats)) < 0)
87 ✗ return ret;
88
89 1534 in_samplerates = ff_all_samplerates();
90
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 1534 if ((ret = ff_formats_ref(in_samplerates, &cfg_in[0]->samplerates)) < 0)
91 ✗ return ret;
92
93 1534 in_layouts = ff_all_channel_counts();
94
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 1534 if ((ret = ff_channel_layouts_ref(in_layouts, &cfg_in[0]->channel_layouts)) < 0)
95 ✗ return ret;
96
97
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 1534 if(out_rate > 0) {
98 640 int ratelist[] = { out_rate, -1 };
99 640 out_samplerates = ff_make_format_list(ratelist);
100 } else {
101 894 out_samplerates = ff_all_samplerates();
102 }
103
104
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 1534 if ((ret = ff_formats_ref(out_samplerates, &cfg_out[0]->samplerates)) < 0)
105 ✗ return ret;
106
107
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 1534 if(out_format != AV_SAMPLE_FMT_NONE) {
108 ✗ enum AVSampleFormat formatlist[] = { out_format, AV_SAMPLE_FMT_NONE };
109 ✗ out_formats = ff_make_sample_format_list(formatlist);
110 } else
111 1534 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
112
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 1534 if ((ret = ff_formats_ref(out_formats, &cfg_out[0]->formats)) < 0)
113 ✗ return ret;
114
115 1534 av_opt_get_chlayout(aresample->swr, "ochl", 0, &out_layout);
116
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 1534 if (av_channel_layout_check(&out_layout)) {
117 ✗ const AVChannelLayout layout_list[] = { out_layout, { 0 } };
118 ✗ out_layouts = ff_make_channel_layout_list(layout_list);
119 } else
120 1534 out_layouts = ff_all_channel_counts();
121 1534 av_channel_layout_uninit(&out_layout);
122
123 1534 return ff_channel_layouts_ref(out_layouts, &cfg_out[0]->channel_layouts);
124 }
125
126 #define SWR_CH_MAX 64
127
128 1534 static int config_output(AVFilterLink *outlink)
129 {
130 int ret;
131 1534 AVFilterContext *ctx = outlink->src;
132 1534 AVFilterLink *inlink = ctx->inputs[0];
133 1534 AResampleContext *aresample = ctx->priv;
134 1534 AVChannelLayout out_layout = { 0 };
135 int64_t out_rate;
136 const AVFrameSideData *sd;
137 enum AVSampleFormat out_format;
138 1534 enum AVMatrixEncoding matrix_encoding = AV_MATRIX_ENCODING_NONE;
139 char inchl_buf[128], outchl_buf[128];
140
141 1534 ret = swr_alloc_set_opts2(&aresample->swr,
142 1534 &outlink->ch_layout, outlink->format, outlink->sample_rate,
143 1534 &inlink->ch_layout, inlink->format, inlink->sample_rate,
144 0, ctx);
145
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 1534 if (ret < 0)
146 ✗ return ret;
147
148 1534 sd = av_frame_side_data_get(inlink->side_data, inlink->nb_side_data,
149 AV_FRAME_DATA_DOWNMIX_MATRIX);
150
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 1534 if (sd) {
151 4 const AVDownmixMatrix *dm = (AVDownmixMatrix *)sd->data;
152
153
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 8 if (inlink->ch_layout.nb_channels == dm->in_ch_count &&
154 4 !av_channel_layout_compare(&outlink->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) {
155 2 swr_set_matrix(aresample->swr,
156 2 (double *)((uint8_t *)dm + dm->coeffs_offset),
157 2 dm->in_ch_count);
158
159
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 2 switch (dm->downmix_type) {
160 ✗ case AV_DOWNMIX_TYPE_LTRT:
161 ✗ matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
162 ✗ break;
163 ✗ case AV_DOWNMIX_TYPE_DPLII:
164 ✗ matrix_encoding = AV_MATRIX_ENCODING_DPLII;
165 ✗ break;
166 2 default:
167 2 break;
168 }
169 }
170 }
171
172 1534 sd = av_frame_side_data_get(inlink->side_data, inlink->nb_side_data,
173 AV_FRAME_DATA_DOWNMIX_INFO);
174
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 1534 if (sd) {
175 5 const AVDownmixInfo *di = (AVDownmixInfo *)sd->data;
176 double center_mix_level, surround_mix_level;
177
178
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 5 switch (di->preferred_downmix_type) {
179 1 case AV_DOWNMIX_TYPE_LTRT:
180 1 matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
181 1 center_mix_level = di->center_mix_level_ltrt;
182 1 surround_mix_level = di->surround_mix_level_ltrt;
183 1 break;
184 1 case AV_DOWNMIX_TYPE_DPLII:
185 1 matrix_encoding = AV_MATRIX_ENCODING_DPLII;
186 1 center_mix_level = di->center_mix_level_ltrt;
187 1 surround_mix_level = di->surround_mix_level_ltrt;
188 1 break;
189 3 default:
190 3 center_mix_level = di->center_mix_level;
191 3 surround_mix_level = di->surround_mix_level;
192 3 break;
193 }
194
195 // Don't use Dolby Surround compatible coeffs when not downmixing to stereo
196
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 5 if (av_channel_layout_compare(&outlink->ch_layout,
197 5 &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) {
198 5 matrix_encoding = AV_MATRIX_ENCODING_NONE;
199 5 center_mix_level = di->center_mix_level;
200 5 surround_mix_level = di->surround_mix_level;
201 }
202
203 5 av_log(ctx, AV_LOG_VERBOSE, "Mix levels: center %f - "
204 "surround %f - lfe %f.\n",
205 5 center_mix_level, surround_mix_level, di->lfe_mix_level);
206
207 5 av_opt_set_double(aresample->swr, "clev", center_mix_level, 0);
208 5 av_opt_set_double(aresample->swr, "slev", surround_mix_level, 0);
209 5 av_opt_set_double(aresample->swr, "lfe_mix_level", di->lfe_mix_level, 0);
210 5 av_opt_set_int(aresample->swr, "matrix_encoding", matrix_encoding, 0);
211 }
212
213
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 1534 if (av_channel_layout_compare(&outlink->ch_layout, &inlink->ch_layout)) {
214 15 av_frame_side_data_remove_by_props(&outlink->side_data, &outlink->nb_side_data,
215 AV_SIDE_DATA_PROP_CHANNEL_DEPENDENT);
216
217
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 15 if (matrix_encoding != AV_MATRIX_ENCODING_NONE) {
218 ✗ AVFrameSideData *side_data = av_frame_side_data_new(&outlink->side_data, &outlink->nb_side_data,
219 AV_FRAME_DATA_MATRIXENCODING,
220 sizeof(matrix_encoding), 0);
221
222 ✗ if (!side_data)
223 ✗ return AVERROR(ENOMEM);
224
225 ✗ *(enum AVMatrixEncoding *)side_data->data = matrix_encoding;
226 }
227 }
228
229 1534 ret = swr_init(aresample->swr);
230
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 1534 if (ret < 0)
231 ✗ return ret;
232
233 1534 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
234 1534 av_opt_get_chlayout(aresample->swr, "ochl", 0, &out_layout);
235 1534 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
236 1534 outlink->time_base = (AVRational) {1, out_rate};
237
238
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 1534 av_assert0(outlink->sample_rate == out_rate);
239
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 1534 av_assert0(!av_channel_layout_compare(&outlink->ch_layout, &out_layout));
240
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 1534 av_assert0(outlink->format == out_format);
241
242 1534 av_channel_layout_uninit(&out_layout);
243
244 1534 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
245
246 1534 av_channel_layout_describe(&inlink ->ch_layout, inchl_buf, sizeof(inchl_buf));
247 1534 av_channel_layout_describe(&outlink->ch_layout, outchl_buf, sizeof(outchl_buf));
248
249 1534 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
250 1534 inlink ->ch_layout.nb_channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
251 1534 outlink->ch_layout.nb_channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
252 1534 return 0;
253 }
254
255 263783 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref, AVFrame **outsamplesref_ret)
256 {
257 263783 AVFilterContext *ctx = inlink->dst;
258 263783 AResampleContext *aresample = ctx->priv;
259 263783 const int n_in = insamplesref->nb_samples;
260 int64_t delay;
261 263783 int n_out = n_in * aresample->ratio + 32;
262 263783 AVFilterLink *const outlink = inlink->dst->outputs[0];
263 AVFrame *outsamplesref;
264 int ret;
265
266 263783 *outsamplesref_ret = NULL;
267 263783 delay = swr_get_delay(aresample->swr, outlink->sample_rate);
268
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 263783 if (delay > 0)
269 7957 n_out += FFMIN(delay, FFMAX(4096, n_out));
270
271 263783 outsamplesref = ff_get_audio_buffer(outlink, n_out);
272
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 263783 if (!outsamplesref)
273 ✗ return AVERROR(ENOMEM);
274
275 263783 av_frame_copy_props(outsamplesref, insamplesref);
276 263783 outsamplesref->format = outlink->format;
277 263783 ret = av_channel_layout_copy(&outsamplesref->ch_layout, &outlink->ch_layout);
278
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 263783 if (ret < 0) {
279 ✗ av_frame_free(&outsamplesref);
280 ✗ return ret;
281 }
282 263783 outsamplesref->sample_rate = outlink->sample_rate;
283
284
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 263783 if (av_channel_layout_compare(&outsamplesref->ch_layout, &insamplesref->ch_layout))
285 543 av_frame_side_data_remove_by_props(&outsamplesref->side_data, &outsamplesref->nb_side_data,
286 AV_SIDE_DATA_PROP_CHANNEL_DEPENDENT);
287
288
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 263783 if(insamplesref->pts != AV_NOPTS_VALUE) {
289 263783 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
290 263783 int64_t outpts= swr_next_pts(aresample->swr, inpts);
291 263783 aresample->next_pts =
292
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 263783 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
293 } else {
294 ✗ outsamplesref->pts = AV_NOPTS_VALUE;
295 }
296 263783 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
297 263783 (void *)insamplesref->extended_data, n_in);
298
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 263783 if (n_out <= 0) {
299 ✗ av_frame_free(&outsamplesref);
300 ✗ return 0;
301 }
302
303 263783 aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
304
305 263783 outsamplesref->nb_samples = n_out;
306
307 263783 *outsamplesref_ret = outsamplesref;
308 263783 return 1;
309 }
310
311 2076 static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
312 {
313 2076 AVFilterContext *ctx = outlink->src;
314 2076 AResampleContext *aresample = ctx->priv;
315 2076 AVFilterLink *const inlink = outlink->src->inputs[0];
316 AVFrame *outsamplesref;
317 2076 int n_out = 4096;
318 int64_t pts;
319
320 2076 outsamplesref = ff_get_audio_buffer(outlink, n_out);
321 2076 *outsamplesref_ret = outsamplesref;
322
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 2076 if (!outsamplesref)
323 ✗ return AVERROR(ENOMEM);
324
325 2076 pts = swr_next_pts(aresample->swr, INT64_MIN);
326
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 2076 pts = ROUNDED_DIV(pts, inlink->sample_rate);
327
328
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 2076 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
329
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 2076 if (n_out <= 0) {
330 1315 av_frame_free(&outsamplesref);
331 1315 return n_out;
332 }
333
334 761 outsamplesref->sample_rate = outlink->sample_rate;
335 761 outsamplesref->nb_samples = n_out;
336
337 761 outsamplesref->pts = pts;
338
339 761 return 1;
340 }
341
342 531891 static int activate(AVFilterContext *ctx)
343 {
344 531891 AVFilterLink *inlink = ctx->inputs[0];
345 531891 AVFilterLink *outlink = ctx->outputs[0];
346 531891 AResampleContext *aresample = ctx->priv;
347 AVFrame *frame;
348 531891 int ret = 0, status;
349 int64_t pts;
350
351
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 531891 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
352
353 // First try to get data from the internal buffers
354
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 530751 if (aresample->more_data) {
355 AVFrame *outsamplesref;
356
357 127 ret = flush_frame(outlink, 0, &outsamplesref);
358
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 127 if (ret < 0)
359 123 return ret;
360
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 127 if (ret > 0)
361 123 return ff_filter_frame(outlink, outsamplesref);
362 }
363 530628 aresample->more_data = 0;
364
365 // Then consume frames from inlink
366
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 530628 while ((ret = ff_inlink_consume_frame(inlink, &frame))) {
367 AVFrame *outsamplesref;
368
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 263783 if (ret < 0)
369 263783 return ret;
370
371 263783 ret = filter_frame(inlink, frame, &outsamplesref);
372 263783 av_frame_free(&frame);
373
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 263783 if (ret < 0)
374 ✗ return ret;
375
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 263783 if (ret > 0)
376 263783 return ff_filter_frame(outlink, outsamplesref);
377 }
378
379 // If we hit the end flush
380
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 266845 if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
381 AVFrame *outsamplesref;
382
383 1949 ret = flush_frame(outlink, 1, &outsamplesref);
384
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 1949 if (ret < 0)
385 ✗ return ret;
386
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 1949 if (ret > 0)
387 638 return ff_filter_frame(outlink, outsamplesref);
388 1311 ff_outlink_set_status(outlink, status, aresample->next_pts);
389 1311 return 0;
390 }
391
392 // If not, request more data from the input
393
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 264896 FF_FILTER_FORWARD_WANTED(outlink, inlink);
394
395 ✗ return FFERROR_NOT_READY;
396 }
397
398 ✗ static const AVClass *resample_child_class_iterate(void **iter)
399 {
400 ✗ const AVClass *c = *iter ? NULL : swr_get_class();
401 ✗ *iter = (void*)(uintptr_t)c;
402 ✗ return c;
403 }
404
405 6693 static void *resample_child_next(void *obj, void *prev)
406 {
407 6693 AResampleContext *s = obj;
408
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 6693 return prev ? NULL : s->swr;
409 }
410
411 #define OFFSET(x) offsetof(AResampleContext, x)
412 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
413
414 static const AVOption options[] = {
415 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
416 {NULL}
417 };
418
419 static const AVClass aresample_class = {
420 .class_name = "aresample",
421 .item_name = av_default_item_name,
422 .option = options,
423 .version = LIBAVUTIL_VERSION_INT,
424 .child_class_iterate = resample_child_class_iterate,
425 .child_next = resample_child_next,
426 };
427
428 static const AVFilterPad aresample_outputs[] = {
429 {
430 .name = "default",
431 .config_props = config_output,
432 .type = AVMEDIA_TYPE_AUDIO,
433 },
434 };
435
436 const FFFilter ff_af_aresample = {
437 .p.name = "aresample",
438 .p.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
439 .p.priv_class = &aresample_class,
440 .preinit = preinit,
441 .activate = activate,
442 .uninit = uninit,
443 .priv_size = sizeof(AResampleContext),
444 FILTER_INPUTS(ff_audio_default_filterpad),
445 FILTER_OUTPUTS(aresample_outputs),
446 FILTER_QUERY_FUNC2(query_formats),
447 };
448