FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_aresample.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 144 174 82.8%
Functions: 9 10 90.0%
Branches: 57 84 67.9%
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/opt.h"
30 #include "libavutil/samplefmt.h"
31 #include "libavutil/avassert.h"
32 #include "libswresample/swresample.h"
33 #include "avfilter.h"
34 #include "audio.h"
35 #include "filters.h"
36 #include "formats.h"
37
38 typedef struct AResampleContext {
39 const AVClass *class;
40 int sample_rate_arg;
41 double ratio;
42 struct SwrContext *swr;
43 int64_t next_pts;
44 int more_data;
45 int eof;
46 } AResampleContext;
47
48 2316 static av_cold int preinit(AVFilterContext *ctx)
49 {
50 2316 AResampleContext *aresample = ctx->priv;
51
52 2316 aresample->next_pts = AV_NOPTS_VALUE;
53 2316 aresample->swr = swr_alloc();
54
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 2316 if (!aresample->swr)
55 ✗ return AVERROR(ENOMEM);
56
57 2316 return 0;
58 }
59
60 2316 static av_cold void uninit(AVFilterContext *ctx)
61 {
62 2316 AResampleContext *aresample = ctx->priv;
63 2316 swr_free(&aresample->swr);
64 2316 }
65
66 1464 static int query_formats(const AVFilterContext *ctx,
67 AVFilterFormatsConfig **cfg_in,
68 AVFilterFormatsConfig **cfg_out)
69 {
70 1464 const AResampleContext *aresample = ctx->priv;
71 enum AVSampleFormat out_format;
72 1464 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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 1464 if (aresample->sample_rate_arg > 0)
81 639 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
82 1464 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
83 1464 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
84
85 1464 in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
86
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 1464 if ((ret = ff_formats_ref(in_formats, &cfg_in[0]->formats)) < 0)
87 ✗ return ret;
88
89 1464 in_samplerates = ff_all_samplerates();
90
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 1464 if ((ret = ff_formats_ref(in_samplerates, &cfg_in[0]->samplerates)) < 0)
91 ✗ return ret;
92
93 1464 in_layouts = ff_all_channel_counts();
94
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 1464 if ((ret = ff_channel_layouts_ref(in_layouts, &cfg_in[0]->channel_layouts)) < 0)
95 ✗ return ret;
96
97
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 1464 if(out_rate > 0) {
98 639 int ratelist[] = { out_rate, -1 };
99 639 out_samplerates = ff_make_format_list(ratelist);
100 } else {
101 825 out_samplerates = ff_all_samplerates();
102 }
103
104
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 1464 if ((ret = ff_formats_ref(out_samplerates, &cfg_out[0]->samplerates)) < 0)
105 ✗ return ret;
106
107
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 1464 if(out_format != AV_SAMPLE_FMT_NONE) {
108 ✗ int formatlist[] = { out_format, -1 };
109 ✗ out_formats = ff_make_format_list(formatlist);
110 } else
111 1464 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
112
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 1464 if ((ret = ff_formats_ref(out_formats, &cfg_out[0]->formats)) < 0)
113 ✗ return ret;
114
115 1464 av_opt_get_chlayout(aresample->swr, "ochl", 0, &out_layout);
116
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 1464 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 1464 out_layouts = ff_all_channel_counts();
121 1464 av_channel_layout_uninit(&out_layout);
122
123 1464 return ff_channel_layouts_ref(out_layouts, &cfg_out[0]->channel_layouts);
124 }
125
126
127 1464 static int config_output(AVFilterLink *outlink)
128 {
129 int ret;
130 1464 AVFilterContext *ctx = outlink->src;
131 1464 AVFilterLink *inlink = ctx->inputs[0];
132 1464 AResampleContext *aresample = ctx->priv;
133 1464 AVChannelLayout out_layout = { 0 };
134 int64_t out_rate;
135 enum AVSampleFormat out_format;
136 char inchl_buf[128], outchl_buf[128];
137
138 1464 ret = swr_alloc_set_opts2(&aresample->swr,
139 1464 &outlink->ch_layout, outlink->format, outlink->sample_rate,
140 1464 &inlink->ch_layout, inlink->format, inlink->sample_rate,
141 0, ctx);
142
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 1464 if (ret < 0)
143 ✗ return ret;
144
145 1464 ret = swr_init(aresample->swr);
146
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 1464 if (ret < 0)
147 ✗ return ret;
148
149 1464 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
150 1464 av_opt_get_chlayout(aresample->swr, "ochl", 0, &out_layout);
151 1464 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
152 1464 outlink->time_base = (AVRational) {1, out_rate};
153
154
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 1464 av_assert0(outlink->sample_rate == out_rate);
155
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 1464 av_assert0(!av_channel_layout_compare(&outlink->ch_layout, &out_layout));
156
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 1464 av_assert0(outlink->format == out_format);
157
158 1464 av_channel_layout_uninit(&out_layout);
159
160 1464 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
161
162 1464 av_channel_layout_describe(&inlink ->ch_layout, inchl_buf, sizeof(inchl_buf));
163 1464 av_channel_layout_describe(&outlink->ch_layout, outchl_buf, sizeof(outchl_buf));
164
165 1464 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
166 1464 inlink ->ch_layout.nb_channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
167 1464 outlink->ch_layout.nb_channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
168 1464 return 0;
169 }
170
171 213119 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
172 {
173 213119 AVFilterContext *ctx = inlink->dst;
174 213119 AResampleContext *aresample = ctx->priv;
175 213119 const int n_in = insamplesref->nb_samples;
176 int64_t delay;
177 213119 int n_out = n_in * aresample->ratio + 32;
178 213119 AVFilterLink *const outlink = inlink->dst->outputs[0];
179 AVFrame *outsamplesref;
180 int ret;
181
182 213119 delay = swr_get_delay(aresample->swr, outlink->sample_rate);
183
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 213119 if (delay > 0)
184 7740 n_out += FFMIN(delay, FFMAX(4096, n_out));
185
186 213119 outsamplesref = ff_get_audio_buffer(outlink, n_out);
187
188
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 213119 if(!outsamplesref) {
189 ✗ av_frame_free(&insamplesref);
190 ✗ return AVERROR(ENOMEM);
191 }
192
193 213119 av_frame_copy_props(outsamplesref, insamplesref);
194 213119 outsamplesref->format = outlink->format;
195 213119 ret = av_channel_layout_copy(&outsamplesref->ch_layout, &outlink->ch_layout);
196
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 213119 if (ret < 0) {
197 ✗ av_frame_free(&outsamplesref);
198 ✗ av_frame_free(&insamplesref);
199 ✗ return ret;
200 }
201 213119 outsamplesref->sample_rate = outlink->sample_rate;
202
203
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 213119 if(insamplesref->pts != AV_NOPTS_VALUE) {
204 213119 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
205 213119 int64_t outpts= swr_next_pts(aresample->swr, inpts);
206 213119 aresample->next_pts =
207
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 213119 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
208 } else {
209 ✗ outsamplesref->pts = AV_NOPTS_VALUE;
210 }
211 213119 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
212 213119 (void *)insamplesref->extended_data, n_in);
213
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 213119 if (n_out <= 0) {
214 ✗ av_frame_free(&outsamplesref);
215 ✗ av_frame_free(&insamplesref);
216 ✗ ff_inlink_request_frame(inlink);
217 ✗ return 0;
218 }
219
220 213119 aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
221
222 213119 outsamplesref->nb_samples = n_out;
223
224 213119 ret = ff_filter_frame(outlink, outsamplesref);
225 213119 av_frame_free(&insamplesref);
226 213119 return ret;
227 }
228
229 2035 static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
230 {
231 2035 AVFilterContext *ctx = outlink->src;
232 2035 AResampleContext *aresample = ctx->priv;
233 2035 AVFilterLink *const inlink = outlink->src->inputs[0];
234 AVFrame *outsamplesref;
235 2035 int n_out = 4096;
236 int64_t pts;
237
238 2035 outsamplesref = ff_get_audio_buffer(outlink, n_out);
239 2035 *outsamplesref_ret = outsamplesref;
240
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 2035 if (!outsamplesref)
241 ✗ return AVERROR(ENOMEM);
242
243 2035 pts = swr_next_pts(aresample->swr, INT64_MIN);
244
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 2035 pts = ROUNDED_DIV(pts, inlink->sample_rate);
245
246
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 2035 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
247
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 2035 if (n_out <= 0) {
248 1277 av_frame_free(&outsamplesref);
249
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 1277 return (n_out == 0) ? AVERROR_EOF : n_out;
250 }
251
252 758 outsamplesref->sample_rate = outlink->sample_rate;
253 758 outsamplesref->nb_samples = n_out;
254
255 758 outsamplesref->pts = pts;
256
257 758 return 0;
258 }
259
260 214270 static int request_frame(AVFilterLink *outlink)
261 {
262 214270 AVFilterContext *ctx = outlink->src;
263 214270 AVFilterLink *inlink = ctx->inputs[0];
264 214270 AResampleContext *aresample = ctx->priv;
265 214270 int ret = 0, status;
266 int64_t pts;
267
268 // First try to get data from the internal buffers
269
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 214270 if (aresample->more_data) {
270 AVFrame *outsamplesref;
271
272
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 127 if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
273 123 return ff_filter_frame(outlink, outsamplesref);
274 }
275 }
276 214147 aresample->more_data = 0;
277
278
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 214147 if (!aresample->eof && ff_inlink_acknowledge_status(inlink, &status, &pts))
279 1273 aresample->eof = 1;
280
281 // Second request more data from the input
282
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 214147 if (!aresample->eof)
283
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 212239 FF_FILTER_FORWARD_WANTED(outlink, inlink);
284
285 // Third if we hit the end flush
286
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 1908 if (aresample->eof) {
287 AVFrame *outsamplesref;
288
289
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 1908 if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0) {
290
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 1273 if (ret == AVERROR_EOF) {
291 1273 ff_outlink_set_status(outlink, AVERROR_EOF, aresample->next_pts);
292 1273 return 0;
293 }
294 ✗ return ret;
295 }
296
297 635 return ff_filter_frame(outlink, outsamplesref);
298 }
299
300 ✗ ff_filter_set_ready(ctx, 100);
301 ✗ return 0;
302 }
303
304 428467 static int activate(AVFilterContext *ctx)
305 {
306 428467 AResampleContext *aresample = ctx->priv;
307 428467 AVFilterLink *inlink = ctx->inputs[0];
308 428467 AVFilterLink *outlink = ctx->outputs[0];
309
310
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 428467 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
311
312
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 427389 if (!aresample->eof && ff_inlink_queued_frames(inlink)) {
313 213119 AVFrame *frame = NULL;
314 int ret;
315
316 213119 ret = ff_inlink_consume_frame(inlink, &frame);
317
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 213119 if (ret < 0)
318 213119 return ret;
319
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 213119 if (ret > 0)
320 213119 return filter_frame(inlink, frame);
321 }
322
323 214270 return request_frame(outlink);
324 }
325
326 ✗ static const AVClass *resample_child_class_iterate(void **iter)
327 {
328 ✗ const AVClass *c = *iter ? NULL : swr_get_class();
329 ✗ *iter = (void*)(uintptr_t)c;
330 ✗ return c;
331 }
332
333 6685 static void *resample_child_next(void *obj, void *prev)
334 {
335 6685 AResampleContext *s = obj;
336
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 6685 return prev ? NULL : s->swr;
337 }
338
339 #define OFFSET(x) offsetof(AResampleContext, x)
340 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
341
342 static const AVOption options[] = {
343 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
344 {NULL}
345 };
346
347 static const AVClass aresample_class = {
348 .class_name = "aresample",
349 .item_name = av_default_item_name,
350 .option = options,
351 .version = LIBAVUTIL_VERSION_INT,
352 .child_class_iterate = resample_child_class_iterate,
353 .child_next = resample_child_next,
354 };
355
356 static const AVFilterPad aresample_outputs[] = {
357 {
358 .name = "default",
359 .config_props = config_output,
360 .type = AVMEDIA_TYPE_AUDIO,
361 },
362 };
363
364 const AVFilter ff_af_aresample = {
365 .name = "aresample",
366 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
367 .preinit = preinit,
368 .activate = activate,
369 .uninit = uninit,
370 .priv_size = sizeof(AResampleContext),
371 .priv_class = &aresample_class,
372 FILTER_INPUTS(ff_audio_default_filterpad),
373 FILTER_OUTPUTS(aresample_outputs),
374 FILTER_QUERY_FUNC2(query_formats),
375 };
376