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path: root/libavfilter/af_aresample.c (plain)
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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 "internal.h"
36
37typedef struct {
38 const AVClass *class;
39 int sample_rate_arg;
40 double ratio;
41 struct SwrContext *swr;
42 int64_t next_pts;
43 int more_data;
44} AResampleContext;
45
46static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
47{
48 AResampleContext *aresample = ctx->priv;
49 int ret = 0;
50
51 aresample->next_pts = AV_NOPTS_VALUE;
52 aresample->swr = swr_alloc();
53 if (!aresample->swr) {
54 ret = AVERROR(ENOMEM);
55 goto end;
56 }
57
58 if (opts) {
59 AVDictionaryEntry *e = NULL;
60
61 while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
62 if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
63 goto end;
64 }
65 av_dict_free(opts);
66 }
67 if (aresample->sample_rate_arg > 0)
68 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
69end:
70 return ret;
71}
72
73static av_cold void uninit(AVFilterContext *ctx)
74{
75 AResampleContext *aresample = ctx->priv;
76 swr_free(&aresample->swr);
77}
78
79static int query_formats(AVFilterContext *ctx)
80{
81 AResampleContext *aresample = ctx->priv;
82 enum AVSampleFormat out_format;
83 int64_t out_rate, out_layout;
84
85 AVFilterLink *inlink = ctx->inputs[0];
86 AVFilterLink *outlink = ctx->outputs[0];
87
88 AVFilterFormats *in_formats, *out_formats;
89 AVFilterFormats *in_samplerates, *out_samplerates;
90 AVFilterChannelLayouts *in_layouts, *out_layouts;
91 int ret;
92
93 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
94 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
95 av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
96
97 in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
98 if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
99 return ret;
100
101 in_samplerates = ff_all_samplerates();
102 if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
103 return ret;
104
105 in_layouts = ff_all_channel_counts();
106 if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
107 return ret;
108
109 if(out_rate > 0) {
110 int ratelist[] = { out_rate, -1 };
111 out_samplerates = ff_make_format_list(ratelist);
112 } else {
113 out_samplerates = ff_all_samplerates();
114 }
115
116 if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
117 return ret;
118
119 if(out_format != AV_SAMPLE_FMT_NONE) {
120 int formatlist[] = { out_format, -1 };
121 out_formats = ff_make_format_list(formatlist);
122 } else
123 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
124 if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0)
125 return ret;
126
127 if(out_layout) {
128 int64_t layout_list[] = { out_layout, -1 };
129 out_layouts = avfilter_make_format64_list(layout_list);
130 } else
131 out_layouts = ff_all_channel_counts();
132
133 return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
134}
135
136
137static int config_output(AVFilterLink *outlink)
138{
139 int ret;
140 AVFilterContext *ctx = outlink->src;
141 AVFilterLink *inlink = ctx->inputs[0];
142 AResampleContext *aresample = ctx->priv;
143 int64_t out_rate, out_layout;
144 enum AVSampleFormat out_format;
145 char inchl_buf[128], outchl_buf[128];
146
147 aresample->swr = swr_alloc_set_opts(aresample->swr,
148 outlink->channel_layout, outlink->format, outlink->sample_rate,
149 inlink->channel_layout, inlink->format, inlink->sample_rate,
150 0, ctx);
151 if (!aresample->swr)
152 return AVERROR(ENOMEM);
153 if (!inlink->channel_layout)
154 av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
155 if (!outlink->channel_layout)
156 av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
157
158 ret = swr_init(aresample->swr);
159 if (ret < 0)
160 return ret;
161
162 av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
163 av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
164 av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
165 outlink->time_base = (AVRational) {1, out_rate};
166
167 av_assert0(outlink->sample_rate == out_rate);
168 av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
169 av_assert0(outlink->format == out_format);
170
171 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
172
173 av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
174 av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
175
176 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
177 inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
178 outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
179 return 0;
180}
181
182static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
183{
184 AResampleContext *aresample = inlink->dst->priv;
185 const int n_in = insamplesref->nb_samples;
186 int64_t delay;
187 int n_out = n_in * aresample->ratio + 32;
188 AVFilterLink *const outlink = inlink->dst->outputs[0];
189 AVFrame *outsamplesref;
190 int ret;
191
192 delay = swr_get_delay(aresample->swr, outlink->sample_rate);
193 if (delay > 0)
194 n_out += FFMIN(delay, FFMAX(4096, n_out));
195
196 outsamplesref = ff_get_audio_buffer(outlink, n_out);
197
198 if(!outsamplesref)
199 return AVERROR(ENOMEM);
200
201 av_frame_copy_props(outsamplesref, insamplesref);
202 outsamplesref->format = outlink->format;
203 av_frame_set_channels(outsamplesref, outlink->channels);
204 outsamplesref->channel_layout = outlink->channel_layout;
205 outsamplesref->sample_rate = outlink->sample_rate;
206
207 if(insamplesref->pts != AV_NOPTS_VALUE) {
208 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
209 int64_t outpts= swr_next_pts(aresample->swr, inpts);
210 aresample->next_pts =
211 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
212 } else {
213 outsamplesref->pts = AV_NOPTS_VALUE;
214 }
215 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
216 (void *)insamplesref->extended_data, n_in);
217 if (n_out <= 0) {
218 av_frame_free(&outsamplesref);
219 av_frame_free(&insamplesref);
220 return 0;
221 }
222
223 aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
224
225 outsamplesref->nb_samples = n_out;
226
227 ret = ff_filter_frame(outlink, outsamplesref);
228 av_frame_free(&insamplesref);
229 return ret;
230}
231
232static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
233{
234 AVFilterContext *ctx = outlink->src;
235 AResampleContext *aresample = ctx->priv;
236 AVFilterLink *const inlink = outlink->src->inputs[0];
237 AVFrame *outsamplesref;
238 int n_out = 4096;
239 int64_t pts;
240
241 outsamplesref = ff_get_audio_buffer(outlink, n_out);
242 *outsamplesref_ret = outsamplesref;
243 if (!outsamplesref)
244 return AVERROR(ENOMEM);
245
246 pts = swr_next_pts(aresample->swr, INT64_MIN);
247 pts = ROUNDED_DIV(pts, inlink->sample_rate);
248
249 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
250 if (n_out <= 0) {
251 av_frame_free(&outsamplesref);
252 return (n_out == 0) ? AVERROR_EOF : n_out;
253 }
254
255 outsamplesref->sample_rate = outlink->sample_rate;
256 outsamplesref->nb_samples = n_out;
257
258 outsamplesref->pts = pts;
259
260 return 0;
261}
262
263static int request_frame(AVFilterLink *outlink)
264{
265 AVFilterContext *ctx = outlink->src;
266 AResampleContext *aresample = ctx->priv;
267 int ret;
268
269 // First try to get data from the internal buffers
270 if (aresample->more_data) {
271 AVFrame *outsamplesref;
272
273 if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
274 return ff_filter_frame(outlink, outsamplesref);
275 }
276 }
277 aresample->more_data = 0;
278
279 // Second request more data from the input
280 ret = ff_request_frame(ctx->inputs[0]);
281
282 // Third if we hit the end flush
283 if (ret == AVERROR_EOF) {
284 AVFrame *outsamplesref;
285
286 if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
287 return ret;
288
289 return ff_filter_frame(outlink, outsamplesref);
290 }
291 return ret;
292}
293
294static const AVClass *resample_child_class_next(const AVClass *prev)
295{
296 return prev ? NULL : swr_get_class();
297}
298
299static void *resample_child_next(void *obj, void *prev)
300{
301 AResampleContext *s = obj;
302 return prev ? NULL : s->swr;
303}
304
305#define OFFSET(x) offsetof(AResampleContext, x)
306#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
307
308static const AVOption options[] = {
309 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
310 {NULL}
311};
312
313static const AVClass aresample_class = {
314 .class_name = "aresample",
315 .item_name = av_default_item_name,
316 .option = options,
317 .version = LIBAVUTIL_VERSION_INT,
318 .child_class_next = resample_child_class_next,
319 .child_next = resample_child_next,
320};
321
322static const AVFilterPad aresample_inputs[] = {
323 {
324 .name = "default",
325 .type = AVMEDIA_TYPE_AUDIO,
326 .filter_frame = filter_frame,
327 },
328 { NULL }
329};
330
331static const AVFilterPad aresample_outputs[] = {
332 {
333 .name = "default",
334 .config_props = config_output,
335 .request_frame = request_frame,
336 .type = AVMEDIA_TYPE_AUDIO,
337 },
338 { NULL }
339};
340
341AVFilter ff_af_aresample = {
342 .name = "aresample",
343 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
344 .init_dict = init_dict,
345 .uninit = uninit,
346 .query_formats = query_formats,
347 .priv_size = sizeof(AResampleContext),
348 .priv_class = &aresample_class,
349 .inputs = aresample_inputs,
350 .outputs = aresample_outputs,
351};
352
generated by cgit at 2024-05-17 06:08:05 (GMT)