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af_aresample.c
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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"
29 #include "libavutil/opt.h"
30 #include "libavutil/samplefmt.h"
31 #include "libavutil/avassert.h"
33 #include "avfilter.h"
34 #include "audio.h"
35 #include "internal.h"
36 
37 typedef struct {
38  const AVClass *class;
40  double ratio;
41  struct SwrContext *swr;
42  int64_t next_pts;
45 
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);
69 end:
70  return ret;
71 }
72 
73 static av_cold void uninit(AVFilterContext *ctx)
74 {
75  AResampleContext *aresample = ctx->priv;
76  swr_free(&aresample->swr);
77 }
78 
80 {
81  AResampleContext *aresample = ctx->priv;
82  int out_rate = av_get_int(aresample->swr, "osr", NULL);
83  uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
84  enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
85 
86  AVFilterLink *inlink = ctx->inputs[0];
87  AVFilterLink *outlink = ctx->outputs[0];
88 
90  AVFilterFormats *out_formats;
91  AVFilterFormats *in_samplerates = ff_all_samplerates();
92  AVFilterFormats *out_samplerates;
94  AVFilterChannelLayouts *out_layouts;
95 
96  ff_formats_ref (in_formats, &inlink->out_formats);
97  ff_formats_ref (in_samplerates, &inlink->out_samplerates);
98  ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
99 
100  if(out_rate > 0) {
101  int ratelist[] = { out_rate, -1 };
102  out_samplerates = ff_make_format_list(ratelist);
103  } else {
104  out_samplerates = ff_all_samplerates();
105  }
106  ff_formats_ref(out_samplerates, &outlink->in_samplerates);
107 
108  if(out_format != AV_SAMPLE_FMT_NONE) {
109  int formatlist[] = { out_format, -1 };
110  out_formats = ff_make_format_list(formatlist);
111  } else
112  out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
113  ff_formats_ref(out_formats, &outlink->in_formats);
114 
115  if(out_layout) {
116  int64_t layout_list[] = { out_layout, -1 };
117  out_layouts = avfilter_make_format64_list(layout_list);
118  } else
119  out_layouts = ff_all_channel_counts();
120  ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
121 
122  return 0;
123 }
124 
125 
126 static int config_output(AVFilterLink *outlink)
127 {
128  int ret;
129  AVFilterContext *ctx = outlink->src;
130  AVFilterLink *inlink = ctx->inputs[0];
131  AResampleContext *aresample = ctx->priv;
132  int out_rate;
133  uint64_t out_layout;
134  enum AVSampleFormat out_format;
135  char inchl_buf[128], outchl_buf[128];
136 
137  aresample->swr = swr_alloc_set_opts(aresample->swr,
138  outlink->channel_layout, outlink->format, outlink->sample_rate,
139  inlink->channel_layout, inlink->format, inlink->sample_rate,
140  0, ctx);
141  if (!aresample->swr)
142  return AVERROR(ENOMEM);
143  if (!inlink->channel_layout)
144  av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
145  if (!outlink->channel_layout)
146  av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
147 
148  ret = swr_init(aresample->swr);
149  if (ret < 0)
150  return ret;
151 
152  out_rate = av_get_int(aresample->swr, "osr", NULL);
153  out_layout = av_get_int(aresample->swr, "ocl", NULL);
154  out_format = av_get_int(aresample->swr, "osf", NULL);
155  outlink->time_base = (AVRational) {1, out_rate};
156 
157  av_assert0(outlink->sample_rate == out_rate);
158  av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
159  av_assert0(outlink->format == out_format);
160 
161  aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
162 
163  av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
164  av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
165 
166  av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
167  inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
168  outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
169  return 0;
170 }
171 
172 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
173 {
174  AResampleContext *aresample = inlink->dst->priv;
175  const int n_in = insamplesref->nb_samples;
176  int64_t delay;
177  int n_out = n_in * aresample->ratio + 32;
178  AVFilterLink *const outlink = inlink->dst->outputs[0];
179  AVFrame *outsamplesref;
180  int ret;
181 
182  delay = swr_get_delay(aresample->swr, outlink->sample_rate);
183  if (delay > 0)
184  n_out += delay;
185 
186  outsamplesref = ff_get_audio_buffer(outlink, n_out);
187 
188  if(!outsamplesref)
189  return AVERROR(ENOMEM);
190 
191  av_frame_copy_props(outsamplesref, insamplesref);
192  outsamplesref->format = outlink->format;
193  av_frame_set_channels(outsamplesref, outlink->channels);
194  outsamplesref->channel_layout = outlink->channel_layout;
195  outsamplesref->sample_rate = outlink->sample_rate;
196 
197  if(insamplesref->pts != AV_NOPTS_VALUE) {
198  int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
199  int64_t outpts= swr_next_pts(aresample->swr, inpts);
200  aresample->next_pts =
201  outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
202  } else {
203  outsamplesref->pts = AV_NOPTS_VALUE;
204  }
205  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
206  (void *)insamplesref->extended_data, n_in);
207  if (n_out <= 0) {
208  av_frame_free(&outsamplesref);
209  av_frame_free(&insamplesref);
210  return 0;
211  }
212 
213  outsamplesref->nb_samples = n_out;
214 
215  ret = ff_filter_frame(outlink, outsamplesref);
216  aresample->req_fullfilled= 1;
217  av_frame_free(&insamplesref);
218  return ret;
219 }
220 
221 static int request_frame(AVFilterLink *outlink)
222 {
223  AVFilterContext *ctx = outlink->src;
224  AResampleContext *aresample = ctx->priv;
225  AVFilterLink *const inlink = outlink->src->inputs[0];
226  int ret;
227 
228  aresample->req_fullfilled = 0;
229  do{
230  ret = ff_request_frame(ctx->inputs[0]);
231  }while(!aresample->req_fullfilled && ret>=0);
232 
233  if (ret == AVERROR_EOF) {
234  AVFrame *outsamplesref;
235  int n_out = 4096;
236  int64_t pts;
237 
238  outsamplesref = ff_get_audio_buffer(outlink, n_out);
239  if (!outsamplesref)
240  return AVERROR(ENOMEM);
241 
242  pts = swr_next_pts(aresample->swr, INT64_MIN);
243  pts = ROUNDED_DIV(pts, inlink->sample_rate);
244 
245  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
246  if (n_out <= 0) {
247  av_frame_free(&outsamplesref);
248  return (n_out == 0) ? AVERROR_EOF : n_out;
249  }
250 
251  outsamplesref->sample_rate = outlink->sample_rate;
252  outsamplesref->nb_samples = n_out;
253 
254  outsamplesref->pts = pts;
255 
256  return ff_filter_frame(outlink, outsamplesref);
257  }
258  return ret;
259 }
260 
261 static const AVClass *resample_child_class_next(const AVClass *prev)
262 {
263  return prev ? NULL : swr_get_class();
264 }
265 
266 static void *resample_child_next(void *obj, void *prev)
267 {
268  AResampleContext *s = obj;
269  return prev ? NULL : s->swr;
270 }
271 
272 #define OFFSET(x) offsetof(AResampleContext, x)
273 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
274 
275 static const AVOption options[] = {
276  {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
277  {NULL}
278 };
279 
280 static const AVClass aresample_class = {
281  .class_name = "aresample",
282  .item_name = av_default_item_name,
283  .option = options,
284  .version = LIBAVUTIL_VERSION_INT,
285  .child_class_next = resample_child_class_next,
287 };
288 
289 static const AVFilterPad aresample_inputs[] = {
290  {
291  .name = "default",
292  .type = AVMEDIA_TYPE_AUDIO,
293  .filter_frame = filter_frame,
294  },
295  { NULL }
296 };
297 
298 static const AVFilterPad aresample_outputs[] = {
299  {
300  .name = "default",
301  .config_props = config_output,
302  .request_frame = request_frame,
303  .type = AVMEDIA_TYPE_AUDIO,
304  },
305  { NULL }
306 };
307 
309  .name = "aresample",
310  .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
311  .init_dict = init_dict,
312  .uninit = uninit,
313  .query_formats = query_formats,
314  .priv_size = sizeof(AResampleContext),
315  .priv_class = &aresample_class,
316  .inputs = aresample_inputs,
317  .outputs = aresample_outputs,
318 };