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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  double ratio;
39  struct SwrContext *swr;
40  int64_t next_pts;
43 
44 static av_cold int init(AVFilterContext *ctx, const char *args)
45 {
46  AResampleContext *aresample = ctx->priv;
47  int ret = 0;
48  char *argd = av_strdup(args);
49 
50  aresample->next_pts = AV_NOPTS_VALUE;
51  aresample->swr = swr_alloc();
52  if (!aresample->swr) {
53  ret = AVERROR(ENOMEM);
54  goto end;
55  }
56 
57  if (args) {
58  char *ptr = argd, *token;
59 
60  while (token = av_strtok(ptr, ":", &ptr)) {
61  char *value;
62  av_strtok(token, "=", &value);
63 
64  if (value) {
65  if ((ret = av_opt_set(aresample->swr, token, value, 0)) < 0)
66  goto end;
67  } else {
68  int out_rate;
69  if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
70  goto end;
71  if ((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
72  goto end;
73  }
74  }
75  }
76 end:
77  av_free(argd);
78  return ret;
79 }
80 
81 static av_cold void uninit(AVFilterContext *ctx)
82 {
83  AResampleContext *aresample = ctx->priv;
84  swr_free(&aresample->swr);
85 }
86 
88 {
89  AResampleContext *aresample = ctx->priv;
90  int out_rate = av_get_int(aresample->swr, "osr", NULL);
91  uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
92  enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
93 
94  AVFilterLink *inlink = ctx->inputs[0];
95  AVFilterLink *outlink = ctx->outputs[0];
96 
98  AVFilterFormats *out_formats;
99  AVFilterFormats *in_samplerates = ff_all_samplerates();
100  AVFilterFormats *out_samplerates;
102  AVFilterChannelLayouts *out_layouts;
103 
104  ff_formats_ref (in_formats, &inlink->out_formats);
105  ff_formats_ref (in_samplerates, &inlink->out_samplerates);
106  ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
107 
108  if(out_rate > 0) {
109  out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
110  } else {
111  out_samplerates = ff_all_samplerates();
112  }
113  ff_formats_ref(out_samplerates, &outlink->in_samplerates);
114 
115  if(out_format != AV_SAMPLE_FMT_NONE) {
116  out_formats = ff_make_format_list((int[]){ out_format, -1 });
117  } else
118  out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
119  ff_formats_ref(out_formats, &outlink->in_formats);
120 
121  if(out_layout) {
122  out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
123  } else
124  out_layouts = ff_all_channel_layouts();
125  ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
126 
127  return 0;
128 }
129 
130 
131 static int config_output(AVFilterLink *outlink)
132 {
133  int ret;
134  AVFilterContext *ctx = outlink->src;
135  AVFilterLink *inlink = ctx->inputs[0];
136  AResampleContext *aresample = ctx->priv;
137  int out_rate;
138  uint64_t out_layout;
139  enum AVSampleFormat out_format;
140  char inchl_buf[128], outchl_buf[128];
141 
142  aresample->swr = swr_alloc_set_opts(aresample->swr,
143  outlink->channel_layout, outlink->format, outlink->sample_rate,
144  inlink->channel_layout, inlink->format, inlink->sample_rate,
145  0, ctx);
146  if (!aresample->swr)
147  return AVERROR(ENOMEM);
148 
149  ret = swr_init(aresample->swr);
150  if (ret < 0)
151  return ret;
152 
153  out_rate = av_get_int(aresample->swr, "osr", NULL);
154  out_layout = av_get_int(aresample->swr, "ocl", NULL);
155  out_format = av_get_int(aresample->swr, "osf", NULL);
156  outlink->time_base = (AVRational) {1, out_rate};
157 
158  av_assert0(outlink->sample_rate == out_rate);
159  av_assert0(outlink->channel_layout == out_layout);
160  av_assert0(outlink->format == out_format);
161 
162  aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
163 
164  av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), -1, inlink ->channel_layout);
165  av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);
166 
167  av_log(ctx, AV_LOG_VERBOSE, "chl:%s fmt:%s r:%dHz -> chl:%s fmt:%s r:%dHz\n",
168  inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
169  outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
170  return 0;
171 }
172 
173 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
174 {
175  AResampleContext *aresample = inlink->dst->priv;
176  const int n_in = insamplesref->audio->nb_samples;
177  int n_out = n_in * aresample->ratio * 2 + 256;
178  AVFilterLink *const outlink = inlink->dst->outputs[0];
179  AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
180  int ret;
181 
182  if(!outsamplesref)
183  return AVERROR(ENOMEM);
184 
185  avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
186  outsamplesref->format = outlink->format;
187  outsamplesref->audio->channels = outlink->channels;
188  outsamplesref->audio->channel_layout = outlink->channel_layout;
189  outsamplesref->audio->sample_rate = outlink->sample_rate;
190 
191  if(insamplesref->pts != AV_NOPTS_VALUE) {
192  int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
193  int64_t outpts= swr_next_pts(aresample->swr, inpts);
194  aresample->next_pts =
195  outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
196  } else {
197  outsamplesref->pts = AV_NOPTS_VALUE;
198  }
199  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
200  (void *)insamplesref->extended_data, n_in);
201  if (n_out <= 0) {
202  avfilter_unref_buffer(outsamplesref);
203  avfilter_unref_buffer(insamplesref);
204  return 0;
205  }
206 
207  outsamplesref->audio->nb_samples = n_out;
208 
209  ret = ff_filter_frame(outlink, outsamplesref);
210  aresample->req_fullfilled= 1;
211  avfilter_unref_buffer(insamplesref);
212  return ret;
213 }
214 
215 static int request_frame(AVFilterLink *outlink)
216 {
217  AVFilterContext *ctx = outlink->src;
218  AResampleContext *aresample = ctx->priv;
219  AVFilterLink *const inlink = outlink->src->inputs[0];
220  int ret;
221 
222  aresample->req_fullfilled = 0;
223  do{
224  ret = ff_request_frame(ctx->inputs[0]);
225  }while(!aresample->req_fullfilled && ret>=0);
226 
227  if (ret == AVERROR_EOF) {
228  AVFilterBufferRef *outsamplesref;
229  int n_out = 4096;
230 
231  outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
232  if (!outsamplesref)
233  return AVERROR(ENOMEM);
234  n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
235  if (n_out <= 0) {
236  avfilter_unref_buffer(outsamplesref);
237  return (n_out == 0) ? AVERROR_EOF : n_out;
238  }
239 
240  outsamplesref->audio->sample_rate = outlink->sample_rate;
241  outsamplesref->audio->nb_samples = n_out;
242 #if 0
243  outsamplesref->pts = aresample->next_pts;
244  if(aresample->next_pts != AV_NOPTS_VALUE)
245  aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
246 #else
247  outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
248  outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
249 #endif
250 
251  ff_filter_frame(outlink, outsamplesref);
252  return 0;
253  }
254  return ret;
255 }
256 
257 static const AVFilterPad aresample_inputs[] = {
258  {
259  .name = "default",
260  .type = AVMEDIA_TYPE_AUDIO,
261  .filter_frame = filter_frame,
262  .min_perms = AV_PERM_READ,
263  },
264  { NULL },
265 };
266 
267 static const AVFilterPad aresample_outputs[] = {
268  {
269  .name = "default",
270  .config_props = config_output,
271  .request_frame = request_frame,
272  .type = AVMEDIA_TYPE_AUDIO,
273  },
274  { NULL },
275 };
276 
278  .name = "aresample",
279  .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
280  .init = init,
281  .uninit = uninit,
282  .query_formats = query_formats,
283  .priv_size = sizeof(AResampleContext),
284  .inputs = aresample_inputs,
285  .outputs = aresample_outputs,
286 };