FFmpeg
af_resample.c
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Lesser General Public
6  * License as published by the Free Software Foundation; either
7  * version 2.1 of the License, or (at your option) any later version.
8  *
9  * FFmpeg is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12  * Lesser General Public License for more details.
13  *
14  * You should have received a copy of the GNU Lesser General Public
15  * License along with FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 /**
20  * @file
21  * sample format and channel layout conversion audio filter
22  */
23 
24 #include "libavutil/avassert.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/common.h"
27 #include "libavutil/dict.h"
28 #include "libavutil/mathematics.h"
29 #include "libavutil/opt.h"
30 
32 
33 #include "audio.h"
34 #include "avfilter.h"
35 #include "formats.h"
36 #include "internal.h"
37 
38 typedef struct ResampleContext {
39  const AVClass *class;
42 
44  int64_t next_pts;
45  int64_t next_in_pts;
46 
47  /* set by filter_frame() to signal an output frame to request_frame() */
50 
52 {
53  ResampleContext *s = ctx->priv;
54  const AVClass *avr_class = avresample_get_class();
56 
57  while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
58  if (av_opt_find(&avr_class, e->key, NULL, 0,
60  av_dict_set(&s->options, e->key, e->value, 0);
61  }
62 
63  e = NULL;
64  while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
65  av_dict_set(opts, e->key, NULL, 0);
66 
67  /* do not allow the user to override basic format options */
68  av_dict_set(&s->options, "in_channel_layout", NULL, 0);
69  av_dict_set(&s->options, "out_channel_layout", NULL, 0);
70  av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
71  av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
72  av_dict_set(&s->options, "in_sample_rate", NULL, 0);
73  av_dict_set(&s->options, "out_sample_rate", NULL, 0);
74 
75  return 0;
76 }
77 
79 {
80  ResampleContext *s = ctx->priv;
81 
82  if (s->avr) {
84  avresample_free(&s->avr);
85  }
86  av_dict_free(&s->options);
87 }
88 
90 {
91  AVFilterLink *inlink = ctx->inputs[0];
92  AVFilterLink *outlink = ctx->outputs[0];
93  AVFilterFormats *in_formats, *out_formats, *in_samplerates, *out_samplerates;
94  AVFilterChannelLayouts *in_layouts, *out_layouts;
95  int ret;
96 
97  if (!(in_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) ||
98  !(out_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) ||
99  !(in_samplerates = ff_all_samplerates ( )) ||
100  !(out_samplerates = ff_all_samplerates ( )) ||
101  !(in_layouts = ff_all_channel_layouts ( )) ||
102  !(out_layouts = ff_all_channel_layouts ( )))
103  return AVERROR(ENOMEM);
104 
105  if ((ret = ff_formats_ref (in_formats, &inlink->out_formats )) < 0 ||
106  (ret = ff_formats_ref (out_formats, &outlink->in_formats )) < 0 ||
107  (ret = ff_formats_ref (in_samplerates, &inlink->out_samplerates )) < 0 ||
108  (ret = ff_formats_ref (out_samplerates, &outlink->in_samplerates )) < 0 ||
109  (ret = ff_channel_layouts_ref (in_layouts, &inlink->out_channel_layouts)) < 0 ||
110  (ret = ff_channel_layouts_ref (out_layouts, &outlink->in_channel_layouts)) < 0)
111  return ret;
112 
113  return 0;
114 }
115 
116 static int config_output(AVFilterLink *outlink)
117 {
118  AVFilterContext *ctx = outlink->src;
119  AVFilterLink *inlink = ctx->inputs[0];
120  ResampleContext *s = ctx->priv;
121  char buf1[64], buf2[64];
122  int ret;
123 
124  int64_t resampling_forced;
125 
126  if (s->avr) {
127  avresample_close(s->avr);
128  avresample_free(&s->avr);
129  }
130 
131  if (inlink->channel_layout == outlink->channel_layout &&
132  inlink->sample_rate == outlink->sample_rate &&
133  (inlink->format == outlink->format ||
136  av_get_planar_sample_fmt(inlink->format) ==
137  av_get_planar_sample_fmt(outlink->format))))
138  return 0;
139 
140  if (!(s->avr = avresample_alloc_context()))
141  return AVERROR(ENOMEM);
142 
143  if (s->options) {
144  int ret;
145  AVDictionaryEntry *e = NULL;
146  while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
147  av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
148 
149  ret = av_opt_set_dict(s->avr, &s->options);
150  if (ret < 0)
151  return ret;
152  }
153 
154  av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
155  av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
156  av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
157  av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
158  av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
159  av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
160 
161  if ((ret = avresample_open(s->avr)) < 0)
162  return ret;
163 
164  av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
165  s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);
166 
167  if (s->resampling) {
168  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
171  } else
172  outlink->time_base = inlink->time_base;
173 
174  av_get_channel_layout_string(buf1, sizeof(buf1),
175  -1, inlink ->channel_layout);
176  av_get_channel_layout_string(buf2, sizeof(buf2),
177  -1, outlink->channel_layout);
178  av_log(ctx, AV_LOG_VERBOSE,
179  "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
180  av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
181  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
182 
183  return 0;
184 }
185 
186 static int request_frame(AVFilterLink *outlink)
187 {
188  AVFilterContext *ctx = outlink->src;
189  ResampleContext *s = ctx->priv;
190  int ret = 0;
191 
192  s->got_output = 0;
193  while (ret >= 0 && !s->got_output)
194  ret = ff_request_frame(ctx->inputs[0]);
195 
196  /* flush the lavr delay buffer */
197  if (ret == AVERROR_EOF && s->avr) {
198  AVFrame *frame;
199  int nb_samples = avresample_get_out_samples(s->avr, 0);
200 
201  if (!nb_samples)
202  return ret;
203 
204  frame = ff_get_audio_buffer(outlink, nb_samples);
205  if (!frame)
206  return AVERROR(ENOMEM);
207 
208  ret = avresample_convert(s->avr, frame->extended_data,
209  frame->linesize[0], nb_samples,
210  NULL, 0, 0);
211  if (ret <= 0) {
212  av_frame_free(&frame);
213  return (ret == 0) ? AVERROR_EOF : ret;
214  }
215 
216  frame->nb_samples = ret;
217  frame->pts = s->next_pts;
218  return ff_filter_frame(outlink, frame);
219  }
220  return ret;
221 }
222 
224 {
225  AVFilterContext *ctx = inlink->dst;
226  ResampleContext *s = ctx->priv;
227  AVFilterLink *outlink = ctx->outputs[0];
228  int ret;
229 
230  if (s->avr) {
231  AVFrame *out;
232  int delay, nb_samples;
233 
234  /* maximum possible samples lavr can output */
235  delay = avresample_get_delay(s->avr);
236  nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
237 
238  out = ff_get_audio_buffer(outlink, nb_samples);
239  if (!out) {
240  ret = AVERROR(ENOMEM);
241  goto fail;
242  }
243 
244  ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
245  nb_samples, in->extended_data, in->linesize[0],
246  in->nb_samples);
247  if (ret <= 0) {
248  av_frame_free(&out);
249  if (ret < 0)
250  goto fail;
251  }
252 
254 
255  if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {
256  if (in->pts == AV_NOPTS_VALUE) {
257  av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
258  "assuming 0.\n");
259  s->next_pts = 0;
260  } else
261  s->next_pts = av_rescale_q(in->pts, inlink->time_base,
262  outlink->time_base);
263  }
264 
265  if (ret > 0) {
266  out->nb_samples = ret;
267 
268  ret = av_frame_copy_props(out, in);
269  if (ret < 0) {
270  av_frame_free(&out);
271  goto fail;
272  }
273 
274  if (s->resampling) {
275  out->sample_rate = outlink->sample_rate;
276  /* Only convert in->pts if there is a discontinuous jump.
277  This ensures that out->pts tracks the number of samples actually
278  output by the resampler in the absence of such a jump.
279  Otherwise, the rounding in av_rescale_q() and av_rescale()
280  causes off-by-1 errors. */
281  if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
282  out->pts = av_rescale_q(in->pts, inlink->time_base,
283  outlink->time_base) -
284  av_rescale(delay, outlink->sample_rate,
285  inlink->sample_rate);
286  } else
287  out->pts = s->next_pts;
288 
289  s->next_pts = out->pts + out->nb_samples;
290  s->next_in_pts = in->pts + in->nb_samples;
291  } else
292  out->pts = in->pts;
293 
294  ret = ff_filter_frame(outlink, out);
295  s->got_output = 1;
296  }
297 
298 fail:
299  av_frame_free(&in);
300  } else {
301  in->format = outlink->format;
302  ret = ff_filter_frame(outlink, in);
303  s->got_output = 1;
304  }
305 
306  return ret;
307 }
308 
309 static const AVClass *resample_child_class_next(const AVClass *prev)
310 {
311  return prev ? NULL : avresample_get_class();
312 }
313 
314 static void *resample_child_next(void *obj, void *prev)
315 {
316  ResampleContext *s = obj;
317  return prev ? NULL : s->avr;
318 }
319 
320 static const AVClass resample_class = {
321  .class_name = "resample",
322  .item_name = av_default_item_name,
323  .version = LIBAVUTIL_VERSION_INT,
324  .child_class_next = resample_child_class_next,
326 };
327 
329  {
330  .name = "default",
331  .type = AVMEDIA_TYPE_AUDIO,
332  .filter_frame = filter_frame,
333  },
334  { NULL }
335 };
336 
338  {
339  .name = "default",
340  .type = AVMEDIA_TYPE_AUDIO,
341  .config_props = config_output,
342  .request_frame = request_frame
343  },
344  { NULL }
345 };
346 
348  .name = "resample",
349  .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
350  .priv_size = sizeof(ResampleContext),
351  .priv_class = &resample_class,
352  .init_dict = init,
353  .uninit = uninit,
355  .inputs = avfilter_af_resample_inputs,
356  .outputs = avfilter_af_resample_outputs,
357 };
#define NULL
Definition: coverity.c:32
static const AVFilterPad avfilter_af_resample_outputs[]
Definition: af_resample.c:337
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
attribute_deprecated int avresample_open(AVAudioResampleContext *avr)
Definition: utils.c:36
attribute_deprecated int avresample_get_delay(AVAudioResampleContext *avr)
Definition: resample.c:438
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
Main libavfilter public API header.
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
attribute_deprecated int avresample_convert(AVAudioResampleContext *avr, uint8_t **output, int out_plane_size, int out_samples, uint8_t *const *input, int in_plane_size, int in_samples)
Definition: utils.c:330
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
const char * name
Pad name.
Definition: internal.h:60
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Public dictionary API.
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:435
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
enum AVSampleFormat format
Definition: resample.h:48
#define av_cold
Definition: attributes.h:82
AVAudioResampleContext * avr
Definition: af_resample.c:40
AVOptions.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
AVDictionaryEntry * av_dict_get(const AVDictionary *m, const char *key, const AVDictionaryEntry *prev, int flags)
Get a dictionary entry with matching key.
Definition: dict.c:40
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
Get the planar alternative form of the given sample format.
Definition: samplefmt.c:84
attribute_deprecated int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
Definition: utils.c:753
#define av_log(a,...)
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_resample.c:78
attribute_deprecated void avresample_close(AVAudioResampleContext *avr)
Definition: utils.c:262
AVFilterFormats * ff_all_formats(enum AVMediaType type)
Return a list of all formats supported by FFmpeg for the given media type.
Definition: formats.c:350
static int config_output(AVFilterLink *outlink)
Definition: af_resample.c:116
A filter pad used for either input or output.
Definition: internal.h:54
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
void *(* child_next)(void *obj, void *prev)
Return next AVOptions-enabled child or NULL.
Definition: log.h:113
static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
Definition: af_resample.c:51
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
void av_dict_free(AVDictionary **pm)
Free all the memory allocated for an AVDictionary struct and all keys and values. ...
Definition: dict.c:203
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:568
simple assert() macros that are a bit more flexible than ISO C assert().
AVDictionary * options
Definition: af_resample.c:41
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
AVFilter ff_af_resample
Definition: af_resample.c:347
#define fail()
Definition: checkasm.h:120
attribute_deprecated int avresample_available(AVAudioResampleContext *avr)
Definition: utils.c:748
const AVOption * av_opt_find(void *obj, const char *name, const char *unit, int opt_flags, int search_flags)
Look for an option in an object.
Definition: opt.c:1608
AVDictionary * opts
Definition: movenc.c:50
static const AVClass resample_class
Definition: af_resample.c:320
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
external API header
static void * resample_child_next(void *obj, void *prev)
Definition: af_resample.c:314
#define AV_OPT_SEARCH_CHILDREN
Search in possible children of the given object first.
Definition: opt.h:556
int ff_formats_ref(AVFilterFormats *f, AVFilterFormats **ref)
Add *ref as a new reference to formats.
Definition: formats.c:440
static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
Definition: af_aresample.c:46
AVFormatContext * ctx
Definition: movenc.c:48
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define s(width, name)
Definition: cbs_vp9.c:257
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (...
Definition: formats.c:401
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
attribute_deprecated const AVClass * avresample_get_class(void)
Definition: options.c:110
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:368
int av_opt_get_int(void *obj, const char *name, int search_flags, int64_t *out_val)
Definition: opt.c:888
sample_rate
int av_opt_set_dict(void *obj, AVDictionary **options)
Set all the options from a given dictionary on an object.
Definition: opt.c:1603
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
Definition: frame.h:326
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:70
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:467
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
const char * name
Filter name.
Definition: avfilter.h:148
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_resample.c:223
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int request_frame(AVFilterLink *outlink)
Definition: af_resample.c:186
attribute_deprecated AVAudioResampleContext * avresample_alloc_context(void)
Definition: options.c:96
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
static int query_formats(AVFilterContext *ctx)
Definition: af_resample.c:89
common internal and external API header
int64_t next_in_pts
Definition: af_resample.c:45
#define AV_OPT_SEARCH_FAKE_OBJ
The obj passed to av_opt_find() is fake – only a double pointer to AVClass instead of a required poi...
Definition: opt.h:565
char * key
Definition: dict.h:86
char * value
Definition: dict.h:87
attribute_deprecated void avresample_free(AVAudioResampleContext **avr)
Definition: utils.c:278
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
FILE * out
Definition: movenc.c:54
#define AV_DICT_IGNORE_SUFFIX
Return first entry in a dictionary whose first part corresponds to the search key, ignoring the suffix of the found key string.
Definition: dict.h:70
static const AVClass * resample_child_class_next(const AVClass *prev)
Definition: af_resample.c:309
static const AVFilterPad avfilter_af_resample_inputs[]
Definition: af_resample.c:328
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:407
internal API functions
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
int64_t next_pts
Definition: af_resample.c:44