FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_aresample.c
Date: 2024-07-17 14:05:47
Exec Total Coverage
Lines: 146 176 83.0%
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 #include "internal.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 int eof;
47 } AResampleContext;
48
49 2170 static av_cold int preinit(AVFilterContext *ctx)
50 {
51 2170 AResampleContext *aresample = ctx->priv;
52
53 2170 aresample->next_pts = AV_NOPTS_VALUE;
54 2170 aresample->swr = swr_alloc();
55
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2170 if (!aresample->swr)
56 return AVERROR(ENOMEM);
57
58 2170 return 0;
59 }
60
61 2170 static av_cold void uninit(AVFilterContext *ctx)
62 {
63 2170 AResampleContext *aresample = ctx->priv;
64 2170 swr_free(&aresample->swr);
65 2170 }
66
67 1318 static int query_formats(AVFilterContext *ctx)
68 {
69 1318 AResampleContext *aresample = ctx->priv;
70 enum AVSampleFormat out_format;
71 1318 AVChannelLayout out_layout = { 0 };
72 int64_t out_rate;
73
74 1318 AVFilterLink *inlink = ctx->inputs[0];
75 1318 AVFilterLink *outlink = ctx->outputs[0];
76
77 AVFilterFormats *in_formats, *out_formats;
78 AVFilterFormats *in_samplerates, *out_samplerates;
79 AVFilterChannelLayouts *in_layouts, *out_layouts;
80 int ret;
81
82
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1318 if (aresample->sample_rate_arg > 0)
83 639 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
84 1318 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
85 1318 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
86
87 1318 in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
88
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1318 if ((ret = ff_formats_ref(in_formats, &inlink->outcfg.formats)) < 0)
89 return ret;
90
91 1318 in_samplerates = ff_all_samplerates();
92
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1318 if ((ret = ff_formats_ref(in_samplerates, &inlink->outcfg.samplerates)) < 0)
93 return ret;
94
95 1318 in_layouts = ff_all_channel_counts();
96
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1318 if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->outcfg.channel_layouts)) < 0)
97 return ret;
98
99
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1318 if(out_rate > 0) {
100 639 int ratelist[] = { out_rate, -1 };
101 639 out_samplerates = ff_make_format_list(ratelist);
102 } else {
103 679 out_samplerates = ff_all_samplerates();
104 }
105
106
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1318 if ((ret = ff_formats_ref(out_samplerates, &outlink->incfg.samplerates)) < 0)
107 return ret;
108
109
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1318 if(out_format != AV_SAMPLE_FMT_NONE) {
110 int formatlist[] = { out_format, -1 };
111 out_formats = ff_make_format_list(formatlist);
112 } else
113 1318 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
114
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1318 if ((ret = ff_formats_ref(out_formats, &outlink->incfg.formats)) < 0)
115 return ret;
116
117 1318 av_opt_get_chlayout(aresample->swr, "ochl", 0, &out_layout);
118
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1318 if (av_channel_layout_check(&out_layout)) {
119 const AVChannelLayout layout_list[] = { out_layout, { 0 } };
120 out_layouts = ff_make_channel_layout_list(layout_list);
121 } else
122 1318 out_layouts = ff_all_channel_counts();
123 1318 av_channel_layout_uninit(&out_layout);
124
125 1318 return ff_channel_layouts_ref(out_layouts, &outlink->incfg.channel_layouts);
126 }
127
128
129 1318 static int config_output(AVFilterLink *outlink)
130 {
131 int ret;
132 1318 AVFilterContext *ctx = outlink->src;
133 1318 AVFilterLink *inlink = ctx->inputs[0];
134 1318 AResampleContext *aresample = ctx->priv;
135 1318 AVChannelLayout out_layout = { 0 };
136 int64_t out_rate;
137 enum AVSampleFormat out_format;
138 char inchl_buf[128], outchl_buf[128];
139
140 1318 ret = swr_alloc_set_opts2(&aresample->swr,
141 1318 &outlink->ch_layout, outlink->format, outlink->sample_rate,
142 1318 &inlink->ch_layout, inlink->format, inlink->sample_rate,
143 0, ctx);
144
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1318 if (ret < 0)
145 return ret;
146
147 1318 ret = swr_init(aresample->swr);
148
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1318 if (ret < 0)
149 return ret;
150
151 1318 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
152 1318 av_opt_get_chlayout(aresample->swr, "ochl", 0, &out_layout);
153 1318 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
154 1318 outlink->time_base = (AVRational) {1, out_rate};
155
156
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1318 av_assert0(outlink->sample_rate == out_rate);
157
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1318 av_assert0(!av_channel_layout_compare(&outlink->ch_layout, &out_layout));
158
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1318 av_assert0(outlink->format == out_format);
159
160 1318 av_channel_layout_uninit(&out_layout);
161
162 1318 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
163
164 1318 av_channel_layout_describe(&inlink ->ch_layout, inchl_buf, sizeof(inchl_buf));
165 1318 av_channel_layout_describe(&outlink->ch_layout, outchl_buf, sizeof(outchl_buf));
166
167 1318 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
168 1318 inlink ->ch_layout.nb_channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
169 1318 outlink->ch_layout.nb_channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
170 1318 return 0;
171 }
172
173 212754 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
174 {
175 212754 AVFilterContext *ctx = inlink->dst;
176 212754 AResampleContext *aresample = ctx->priv;
177 212754 const int n_in = insamplesref->nb_samples;
178 int64_t delay;
179 212754 int n_out = n_in * aresample->ratio + 32;
180 212754 AVFilterLink *const outlink = inlink->dst->outputs[0];
181 AVFrame *outsamplesref;
182 int ret;
183
184 212754 delay = swr_get_delay(aresample->swr, outlink->sample_rate);
185
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212754 if (delay > 0)
186 7740 n_out += FFMIN(delay, FFMAX(4096, n_out));
187
188 212754 outsamplesref = ff_get_audio_buffer(outlink, n_out);
189
190
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212754 if(!outsamplesref) {
191 av_frame_free(&insamplesref);
192 return AVERROR(ENOMEM);
193 }
194
195 212754 av_frame_copy_props(outsamplesref, insamplesref);
196 212754 outsamplesref->format = outlink->format;
197 212754 ret = av_channel_layout_copy(&outsamplesref->ch_layout, &outlink->ch_layout);
198
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212754 if (ret < 0) {
199 av_frame_free(&outsamplesref);
200 av_frame_free(&insamplesref);
201 return ret;
202 }
203 212754 outsamplesref->sample_rate = outlink->sample_rate;
204
205
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212754 if(insamplesref->pts != AV_NOPTS_VALUE) {
206 212754 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
207 212754 int64_t outpts= swr_next_pts(aresample->swr, inpts);
208 212754 aresample->next_pts =
209
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212754 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
210 } else {
211 outsamplesref->pts = AV_NOPTS_VALUE;
212 }
213 212754 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
214 212754 (void *)insamplesref->extended_data, n_in);
215
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212754 if (n_out <= 0) {
216 av_frame_free(&outsamplesref);
217 av_frame_free(&insamplesref);
218 ff_inlink_request_frame(inlink);
219 return 0;
220 }
221
222 212754 aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
223
224 212754 outsamplesref->nb_samples = n_out;
225
226 212754 ret = ff_filter_frame(outlink, outsamplesref);
227 212754 av_frame_free(&insamplesref);
228 212754 return ret;
229 }
230
231 1969 static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
232 {
233 1969 AVFilterContext *ctx = outlink->src;
234 1969 AResampleContext *aresample = ctx->priv;
235 1969 AVFilterLink *const inlink = outlink->src->inputs[0];
236 AVFrame *outsamplesref;
237 1969 int n_out = 4096;
238 int64_t pts;
239
240 1969 outsamplesref = ff_get_audio_buffer(outlink, n_out);
241 1969 *outsamplesref_ret = outsamplesref;
242
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1969 if (!outsamplesref)
243 return AVERROR(ENOMEM);
244
245 1969 pts = swr_next_pts(aresample->swr, INT64_MIN);
246
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1969 pts = ROUNDED_DIV(pts, inlink->sample_rate);
247
248
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1969 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
249
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1969 if (n_out <= 0) {
250 1211 av_frame_free(&outsamplesref);
251
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1211 return (n_out == 0) ? AVERROR_EOF : n_out;
252 }
253
254 758 outsamplesref->sample_rate = outlink->sample_rate;
255 758 outsamplesref->nb_samples = n_out;
256
257 758 outsamplesref->pts = pts;
258
259 758 return 0;
260 }
261
262 214013 static int request_frame(AVFilterLink *outlink)
263 {
264 214013 AVFilterContext *ctx = outlink->src;
265 214013 AVFilterLink *inlink = ctx->inputs[0];
266 214013 AResampleContext *aresample = ctx->priv;
267 214013 int ret = 0, status;
268 int64_t pts;
269
270 // First try to get data from the internal buffers
271
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214013 if (aresample->more_data) {
272 AVFrame *outsamplesref;
273
274
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127 if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
275 123 return ff_filter_frame(outlink, outsamplesref);
276 }
277 }
278 213890 aresample->more_data = 0;
279
280
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213890 if (!aresample->eof && ff_inlink_acknowledge_status(inlink, &status, &pts))
281 1207 aresample->eof = 1;
282
283 // Second request more data from the input
284
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213890 if (!aresample->eof)
285
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212048 FF_FILTER_FORWARD_WANTED(outlink, inlink);
286
287 // Third if we hit the end flush
288
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1842 if (aresample->eof) {
289 AVFrame *outsamplesref;
290
291
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1842 if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0) {
292
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1207 if (ret == AVERROR_EOF) {
293 1207 ff_outlink_set_status(outlink, AVERROR_EOF, aresample->next_pts);
294 1207 return 0;
295 }
296 return ret;
297 }
298
299 635 return ff_filter_frame(outlink, outsamplesref);
300 }
301
302 ff_filter_set_ready(ctx, 100);
303 return 0;
304 }
305
306 427700 static int activate(AVFilterContext *ctx)
307 {
308 427700 AResampleContext *aresample = ctx->priv;
309 427700 AVFilterLink *inlink = ctx->inputs[0];
310 427700 AVFilterLink *outlink = ctx->outputs[0];
311
312
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427700 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
313
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426767 if (!aresample->eof && ff_inlink_queued_frames(inlink)) {
315 212754 AVFrame *frame = NULL;
316 int ret;
317
318 212754 ret = ff_inlink_consume_frame(inlink, &frame);
319
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212754 if (ret < 0)
320 212754 return ret;
321
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212754 if (ret > 0)
322 212754 return filter_frame(inlink, frame);
323 }
324
325 214013 return request_frame(outlink);
326 }
327
328 static const AVClass *resample_child_class_iterate(void **iter)
329 {
330 const AVClass *c = *iter ? NULL : swr_get_class();
331 *iter = (void*)(uintptr_t)c;
332 return c;
333 }
334
335 6685 static void *resample_child_next(void *obj, void *prev)
336 {
337 6685 AResampleContext *s = obj;
338
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6685 return prev ? NULL : s->swr;
339 }
340
341 #define OFFSET(x) offsetof(AResampleContext, x)
342 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
343
344 static const AVOption options[] = {
345 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
346 {NULL}
347 };
348
349 static const AVClass aresample_class = {
350 .class_name = "aresample",
351 .item_name = av_default_item_name,
352 .option = options,
353 .version = LIBAVUTIL_VERSION_INT,
354 .child_class_iterate = resample_child_class_iterate,
355 .child_next = resample_child_next,
356 };
357
358 static const AVFilterPad aresample_outputs[] = {
359 {
360 .name = "default",
361 .config_props = config_output,
362 .type = AVMEDIA_TYPE_AUDIO,
363 },
364 };
365
366 const AVFilter ff_af_aresample = {
367 .name = "aresample",
368 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
369 .preinit = preinit,
370 .activate = activate,
371 .uninit = uninit,
372 .priv_size = sizeof(AResampleContext),
373 .priv_class = &aresample_class,
374 FILTER_INPUTS(ff_audio_default_filterpad),
375 FILTER_OUTPUTS(aresample_outputs),
376 FILTER_QUERY_FUNC(query_formats),
377 };
378