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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;
44  int more_data;
46 
48 {
49  AResampleContext *aresample = ctx->priv;
50  int ret = 0;
51 
52  aresample->next_pts = AV_NOPTS_VALUE;
53  aresample->swr = swr_alloc();
54  if (!aresample->swr) {
55  ret = AVERROR(ENOMEM);
56  goto end;
57  }
58 
59  if (opts) {
61 
62  while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
63  if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
64  goto end;
65  }
66  av_dict_free(opts);
67  }
68  if (aresample->sample_rate_arg > 0)
69  av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
70 end:
71  return ret;
72 }
73 
74 static av_cold void uninit(AVFilterContext *ctx)
75 {
76  AResampleContext *aresample = ctx->priv;
77  swr_free(&aresample->swr);
78 }
79 
81 {
82  AResampleContext *aresample = ctx->priv;
83  int out_rate = av_get_int(aresample->swr, "osr", NULL);
84  uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
85  enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
86 
87  AVFilterLink *inlink = ctx->inputs[0];
88  AVFilterLink *outlink = ctx->outputs[0];
89 
90  AVFilterFormats *in_formats, *out_formats;
91  AVFilterFormats *in_samplerates, *out_samplerates;
92  AVFilterChannelLayouts *in_layouts, *out_layouts;
93 
94 
95  in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
96  if (!in_formats)
97  return AVERROR(ENOMEM);
98  ff_formats_ref (in_formats, &inlink->out_formats);
99 
100  in_samplerates = ff_all_samplerates();
101  if (!in_samplerates)
102  return AVERROR(ENOMEM);
103  ff_formats_ref (in_samplerates, &inlink->out_samplerates);
104 
105  in_layouts = ff_all_channel_counts();
106  if (!in_layouts)
107  return AVERROR(ENOMEM);
108  ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
109 
110  if(out_rate > 0) {
111  int ratelist[] = { out_rate, -1 };
112  out_samplerates = ff_make_format_list(ratelist);
113  } else {
114  out_samplerates = ff_all_samplerates();
115  }
116  if (!out_samplerates) {
117  av_log(ctx, AV_LOG_ERROR, "Cannot allocate output samplerates.\n");
118  return AVERROR(ENOMEM);
119  }
120 
121  ff_formats_ref(out_samplerates, &outlink->in_samplerates);
122 
123  if(out_format != AV_SAMPLE_FMT_NONE) {
124  int formatlist[] = { out_format, -1 };
125  out_formats = ff_make_format_list(formatlist);
126  } else
127  out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
128  ff_formats_ref(out_formats, &outlink->in_formats);
129 
130  if(out_layout) {
131  int64_t layout_list[] = { out_layout, -1 };
132  out_layouts = avfilter_make_format64_list(layout_list);
133  } else
134  out_layouts = ff_all_channel_counts();
135  ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
136 
137  return 0;
138 }
139 
140 
141 static int config_output(AVFilterLink *outlink)
142 {
143  int ret;
144  AVFilterContext *ctx = outlink->src;
145  AVFilterLink *inlink = ctx->inputs[0];
146  AResampleContext *aresample = ctx->priv;
147  int out_rate;
148  uint64_t out_layout;
149  enum AVSampleFormat out_format;
150  char inchl_buf[128], outchl_buf[128];
151 
152  aresample->swr = swr_alloc_set_opts(aresample->swr,
153  outlink->channel_layout, outlink->format, outlink->sample_rate,
154  inlink->channel_layout, inlink->format, inlink->sample_rate,
155  0, ctx);
156  if (!aresample->swr)
157  return AVERROR(ENOMEM);
158  if (!inlink->channel_layout)
159  av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
160  if (!outlink->channel_layout)
161  av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
162 
163  ret = swr_init(aresample->swr);
164  if (ret < 0)
165  return ret;
166 
167  out_rate = av_get_int(aresample->swr, "osr", NULL);
168  out_layout = av_get_int(aresample->swr, "ocl", NULL);
169  out_format = av_get_int(aresample->swr, "osf", NULL);
170  outlink->time_base = (AVRational) {1, out_rate};
171 
172  av_assert0(outlink->sample_rate == out_rate);
173  av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
174  av_assert0(outlink->format == out_format);
175 
176  aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
177 
178  av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
179  av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
180 
181  av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
182  inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
183  outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
184  return 0;
185 }
186 
187 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
188 {
189  AResampleContext *aresample = inlink->dst->priv;
190  const int n_in = insamplesref->nb_samples;
191  int64_t delay;
192  int n_out = n_in * aresample->ratio + 32;
193  AVFilterLink *const outlink = inlink->dst->outputs[0];
194  AVFrame *outsamplesref;
195  int ret;
196 
197  delay = swr_get_delay(aresample->swr, outlink->sample_rate);
198  if (delay > 0)
199  n_out += FFMIN(delay, FFMAX(4096, n_out));
200 
201  outsamplesref = ff_get_audio_buffer(outlink, n_out);
202 
203  if(!outsamplesref)
204  return AVERROR(ENOMEM);
205 
206  av_frame_copy_props(outsamplesref, insamplesref);
207  outsamplesref->format = outlink->format;
208  av_frame_set_channels(outsamplesref, outlink->channels);
209  outsamplesref->channel_layout = outlink->channel_layout;
210  outsamplesref->sample_rate = outlink->sample_rate;
211 
212  if(insamplesref->pts != AV_NOPTS_VALUE) {
213  int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
214  int64_t outpts= swr_next_pts(aresample->swr, inpts);
215  aresample->next_pts =
216  outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
217  } else {
218  outsamplesref->pts = AV_NOPTS_VALUE;
219  }
220  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
221  (void *)insamplesref->extended_data, n_in);
222  if (n_out <= 0) {
223  av_frame_free(&outsamplesref);
224  av_frame_free(&insamplesref);
225  return 0;
226  }
227 
228  aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
229 
230  outsamplesref->nb_samples = n_out;
231 
232  ret = ff_filter_frame(outlink, outsamplesref);
233  aresample->req_fullfilled= 1;
234  av_frame_free(&insamplesref);
235  return ret;
236 }
237 
238 static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
239 {
240  AVFilterContext *ctx = outlink->src;
241  AResampleContext *aresample = ctx->priv;
242  AVFilterLink *const inlink = outlink->src->inputs[0];
243  AVFrame *outsamplesref;
244  int n_out = 4096;
245  int64_t pts;
246 
247  outsamplesref = ff_get_audio_buffer(outlink, n_out);
248  *outsamplesref_ret = outsamplesref;
249  if (!outsamplesref)
250  return AVERROR(ENOMEM);
251 
252  pts = swr_next_pts(aresample->swr, INT64_MIN);
253  pts = ROUNDED_DIV(pts, inlink->sample_rate);
254 
255  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
256  if (n_out <= 0) {
257  av_frame_free(&outsamplesref);
258  return (n_out == 0) ? AVERROR_EOF : n_out;
259  }
260 
261  outsamplesref->sample_rate = outlink->sample_rate;
262  outsamplesref->nb_samples = n_out;
263 
264  outsamplesref->pts = pts;
265 
266  return 0;
267 }
268 
269 static int request_frame(AVFilterLink *outlink)
270 {
271  AVFilterContext *ctx = outlink->src;
272  AResampleContext *aresample = ctx->priv;
273  int ret;
274 
275  // First try to get data from the internal buffers
276  if (aresample->more_data) {
277  AVFrame *outsamplesref;
278 
279  if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
280  return ff_filter_frame(outlink, outsamplesref);
281  }
282  }
283  aresample->more_data = 0;
284 
285  // Second request more data from the input
286  aresample->req_fullfilled = 0;
287  do{
288  ret = ff_request_frame(ctx->inputs[0]);
289  }while(!aresample->req_fullfilled && ret>=0);
290 
291  // Third if we hit the end flush
292  if (ret == AVERROR_EOF) {
293  AVFrame *outsamplesref;
294 
295  if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
296  return ret;
297 
298  return ff_filter_frame(outlink, outsamplesref);
299  }
300  return ret;
301 }
302 
303 static const AVClass *resample_child_class_next(const AVClass *prev)
304 {
305  return prev ? NULL : swr_get_class();
306 }
307 
308 static void *resample_child_next(void *obj, void *prev)
309 {
310  AResampleContext *s = obj;
311  return prev ? NULL : s->swr;
312 }
313 
314 #define OFFSET(x) offsetof(AResampleContext, x)
315 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
316 
317 static const AVOption options[] = {
318  {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
319  {NULL}
320 };
321 
322 static const AVClass aresample_class = {
323  .class_name = "aresample",
324  .item_name = av_default_item_name,
325  .option = options,
326  .version = LIBAVUTIL_VERSION_INT,
327  .child_class_next = resample_child_class_next,
329 };
330 
331 static const AVFilterPad aresample_inputs[] = {
332  {
333  .name = "default",
334  .type = AVMEDIA_TYPE_AUDIO,
335  .filter_frame = filter_frame,
336  },
337  { NULL }
338 };
339 
340 static const AVFilterPad aresample_outputs[] = {
341  {
342  .name = "default",
343  .config_props = config_output,
344  .request_frame = request_frame,
345  .type = AVMEDIA_TYPE_AUDIO,
346  },
347  { NULL }
348 };
349 
351  .name = "aresample",
352  .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
353  .init_dict = init_dict,
354  .uninit = uninit,
355  .query_formats = query_formats,
356  .priv_size = sizeof(AResampleContext),
357  .priv_class = &aresample_class,
358  .inputs = aresample_inputs,
359  .outputs = aresample_outputs,
360 };