FFmpeg
af_aecho.c
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1 /*
2  * Copyright (c) 2013 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/avassert.h"
22 #include "libavutil/avstring.h"
23 #include "libavutil/opt.h"
24 #include "libavutil/samplefmt.h"
25 #include "avfilter.h"
26 #include "audio.h"
27 #include "filters.h"
28 #include "internal.h"
29 
30 typedef struct AudioEchoContext {
31  const AVClass *class;
32  float in_gain, out_gain;
33  char *delays, *decays;
34  float *delay, *decay;
35  int nb_echoes;
39  int *samples;
40  int eof;
41  int64_t next_pts;
42 
44  uint8_t * const *src, uint8_t **dst,
45  int nb_samples, int channels);
47 
48 #define OFFSET(x) offsetof(AudioEchoContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
50 
51 static const AVOption aecho_options[] = {
52  { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
53  { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
54  { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
55  { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
56  { NULL }
57 };
58 
60 
61 static void count_items(char *item_str, int *nb_items)
62 {
63  char *p;
64 
65  *nb_items = 1;
66  for (p = item_str; *p; p++) {
67  if (*p == '|')
68  (*nb_items)++;
69  }
70 
71 }
72 
73 static void fill_items(char *item_str, int *nb_items, float *items)
74 {
75  char *p, *saveptr = NULL;
76  int i, new_nb_items = 0;
77 
78  p = item_str;
79  for (i = 0; i < *nb_items; i++) {
80  char *tstr = av_strtok(p, "|", &saveptr);
81  p = NULL;
82  if (tstr)
83  new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 1;
84  }
85 
86  *nb_items = new_nb_items;
87 }
88 
90 {
91  AudioEchoContext *s = ctx->priv;
92 
93  av_freep(&s->delay);
94  av_freep(&s->decay);
95  av_freep(&s->samples);
96 
97  if (s->delayptrs)
98  av_freep(&s->delayptrs[0]);
99  av_freep(&s->delayptrs);
100 }
101 
103 {
104  AudioEchoContext *s = ctx->priv;
105  int nb_delays, nb_decays, i;
106 
107  if (!s->delays || !s->decays) {
108  av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
109  return AVERROR(EINVAL);
110  }
111 
112  count_items(s->delays, &nb_delays);
113  count_items(s->decays, &nb_decays);
114 
115  s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
116  s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
117  if (!s->delay || !s->decay)
118  return AVERROR(ENOMEM);
119 
120  fill_items(s->delays, &nb_delays, s->delay);
121  fill_items(s->decays, &nb_decays, s->decay);
122 
123  if (nb_delays != nb_decays) {
124  av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
125  return AVERROR(EINVAL);
126  }
127 
128  s->nb_echoes = nb_delays;
129  if (!s->nb_echoes) {
130  av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
131  return AVERROR(EINVAL);
132  }
133 
134  s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
135  if (!s->samples)
136  return AVERROR(ENOMEM);
137 
138  for (i = 0; i < nb_delays; i++) {
139  if (s->delay[i] <= 0 || s->delay[i] > 90000) {
140  av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
141  return AVERROR(EINVAL);
142  }
143  if (s->decay[i] <= 0 || s->decay[i] > 1) {
144  av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
145  return AVERROR(EINVAL);
146  }
147  }
148 
150 
151  av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
152  return 0;
153 }
154 
156 {
159  static const enum AVSampleFormat sample_fmts[] = {
163  };
164  int ret;
165 
166  layouts = ff_all_channel_counts();
167  if (!layouts)
168  return AVERROR(ENOMEM);
169  ret = ff_set_common_channel_layouts(ctx, layouts);
170  if (ret < 0)
171  return ret;
172 
173  formats = ff_make_format_list(sample_fmts);
174  if (!formats)
175  return AVERROR(ENOMEM);
176  ret = ff_set_common_formats(ctx, formats);
177  if (ret < 0)
178  return ret;
179 
180  formats = ff_all_samplerates();
181  if (!formats)
182  return AVERROR(ENOMEM);
183  return ff_set_common_samplerates(ctx, formats);
184 }
185 
186 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
187 
188 #define ECHO(name, type, min, max) \
189 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
190  uint8_t **delayptrs, \
191  uint8_t * const *src, uint8_t **dst, \
192  int nb_samples, int channels) \
193 { \
194  const double out_gain = ctx->out_gain; \
195  const double in_gain = ctx->in_gain; \
196  const int nb_echoes = ctx->nb_echoes; \
197  const int max_samples = ctx->max_samples; \
198  int i, j, chan, av_uninit(index); \
199  \
200  av_assert1(channels > 0); /* would corrupt delay_index */ \
201  \
202  for (chan = 0; chan < channels; chan++) { \
203  const type *s = (type *)src[chan]; \
204  type *d = (type *)dst[chan]; \
205  type *dbuf = (type *)delayptrs[chan]; \
206  \
207  index = ctx->delay_index; \
208  for (i = 0; i < nb_samples; i++, s++, d++) { \
209  double out, in; \
210  \
211  in = *s; \
212  out = in * in_gain; \
213  for (j = 0; j < nb_echoes; j++) { \
214  int ix = index + max_samples - ctx->samples[j]; \
215  ix = MOD(ix, max_samples); \
216  out += dbuf[ix] * ctx->decay[j]; \
217  } \
218  out *= out_gain; \
219  \
220  *d = av_clipd(out, min, max); \
221  dbuf[index] = in; \
222  \
223  index = MOD(index + 1, max_samples); \
224  } \
225  } \
226  ctx->delay_index = index; \
227 }
228 
229 ECHO(dbl, double, -1.0, 1.0 )
230 ECHO(flt, float, -1.0, 1.0 )
231 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
232 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
233 
234 static int config_output(AVFilterLink *outlink)
235 {
236  AVFilterContext *ctx = outlink->src;
237  AudioEchoContext *s = ctx->priv;
238  float volume = 1.0;
239  int i;
240 
241  for (i = 0; i < s->nb_echoes; i++) {
242  s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
243  s->max_samples = FFMAX(s->max_samples, s->samples[i]);
244  volume += s->decay[i];
245  }
246 
247  if (s->max_samples <= 0) {
248  av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
249  return AVERROR(EINVAL);
250  }
251  s->fade_out = s->max_samples;
252 
253  if (volume * s->in_gain * s->out_gain > 1.0)
254  av_log(ctx, AV_LOG_WARNING,
255  "out_gain %f can cause saturation of output\n", s->out_gain);
256 
257  switch (outlink->format) {
258  case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
259  case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
260  case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
261  case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
262  }
263 
264 
265  if (s->delayptrs)
266  av_freep(&s->delayptrs[0]);
267  av_freep(&s->delayptrs);
268 
270  outlink->channels,
271  s->max_samples,
272  outlink->format, 0);
273 }
274 
276 {
277  AVFilterContext *ctx = inlink->dst;
278  AudioEchoContext *s = ctx->priv;
279  AVFrame *out_frame;
280 
281  if (av_frame_is_writable(frame)) {
282  out_frame = frame;
283  } else {
284  out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
285  if (!out_frame) {
286  av_frame_free(&frame);
287  return AVERROR(ENOMEM);
288  }
289  av_frame_copy_props(out_frame, frame);
290  }
291 
292  s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
293  frame->nb_samples, inlink->channels);
294 
295  s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
296 
297  if (frame != out_frame)
298  av_frame_free(&frame);
299 
300  return ff_filter_frame(ctx->outputs[0], out_frame);
301 }
302 
303 static int request_frame(AVFilterLink *outlink)
304 {
305  AVFilterContext *ctx = outlink->src;
306  AudioEchoContext *s = ctx->priv;
307  int nb_samples = FFMIN(s->fade_out, 2048);
308  AVFrame *frame = ff_get_audio_buffer(outlink, nb_samples);
309 
310  if (!frame)
311  return AVERROR(ENOMEM);
312  s->fade_out -= nb_samples;
313 
315  frame->nb_samples,
316  outlink->channels,
317  frame->format);
318 
319  s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
320  frame->nb_samples, outlink->channels);
321 
322  frame->pts = s->next_pts;
323  if (s->next_pts != AV_NOPTS_VALUE)
324  s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
325 
326  return ff_filter_frame(outlink, frame);
327 }
328 
330 {
331  AVFilterLink *inlink = ctx->inputs[0];
332  AVFilterLink *outlink = ctx->outputs[0];
333  AudioEchoContext *s = ctx->priv;
334  AVFrame *in;
335  int ret, status;
336  int64_t pts;
337 
338  FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
339 
340  ret = ff_inlink_consume_frame(inlink, &in);
341  if (ret < 0)
342  return ret;
343  if (ret > 0)
344  return filter_frame(inlink, in);
345 
346  if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
347  if (status == AVERROR_EOF)
348  s->eof = 1;
349  }
350 
351  if (s->eof && s->fade_out <= 0) {
353  return 0;
354  }
355 
356  if (!s->eof)
357  FF_FILTER_FORWARD_WANTED(outlink, inlink);
358 
359  return request_frame(outlink);
360 }
361 
362 static const AVFilterPad aecho_inputs[] = {
363  {
364  .name = "default",
365  .type = AVMEDIA_TYPE_AUDIO,
366  },
367  { NULL }
368 };
369 
370 static const AVFilterPad aecho_outputs[] = {
371  {
372  .name = "default",
373  .config_props = config_output,
374  .type = AVMEDIA_TYPE_AUDIO,
375  },
376  { NULL }
377 };
378 
380  .name = "aecho",
381  .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
382  .query_formats = query_formats,
383  .priv_size = sizeof(AudioEchoContext),
384  .priv_class = &aecho_class,
385  .init = init,
386  .activate = activate,
387  .uninit = uninit,
388  .inputs = aecho_inputs,
389  .outputs = aecho_outputs,
390 };
float, planar
Definition: samplefmt.h:69
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1494
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:550
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
#define av_realloc_f(p, o, n)
AVOption.
Definition: opt.h:246
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
Main libavfilter public API header.
channels
Definition: aptx.c:30
AVFILTER_DEFINE_CLASS(aecho)
static void count_items(char *item_str, int *nb_items)
Definition: af_aecho.c:61
double, planar
Definition: samplefmt.h:70
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_aecho.c:275
#define src
Definition: vp8dsp.c:254
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition: filters.h:189
char * decays
Definition: af_aecho.c:33
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
Definition: samplefmt.c:198
static void fill_items(char *item_str, int *nb_items, float *items)
Definition: af_aecho.c:73
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
const char * name
Pad name.
Definition: internal.h:60
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aecho.c:89
float out_gain
Definition: af_aecho.c:32
void(* echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs, uint8_t *const *src, uint8_t **dst, int nb_samples, int channels)
Definition: af_aecho.c:43
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
static const AVOption aecho_options[]
Definition: af_aecho.c:51
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
#define OFFSET(x)
Definition: af_aecho.c:48
uint8_t ** delayptrs
Definition: af_aecho.c:37
#define ECHO(name, type, min, max)
Definition: af_aecho.c:188
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
static int request_frame(AVFilterLink *outlink)
Definition: af_aecho.c:303
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define av_log(a,...)
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
Definition: filters.h:199
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
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
Definition: avfilter.c:1449
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:569
static int query_formats(AVFilterContext *ctx)
Definition: af_aecho.c:155
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:237
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
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
static int config_output(AVFilterLink *outlink)
Definition: af_aecho.c:234
simple assert() macros that are a bit more flexible than ISO C assert().
#define FFMAX(a, b)
Definition: common.h:94
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Definition: avsscanf.c:962
#define FFMIN(a, b)
Definition: common.h:96
signed 32 bits, planar
Definition: samplefmt.h:68
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int32_t
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
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:368
AVFilter ff_af_aecho
Definition: af_aecho.c:379
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
static const AVFilterPad aecho_outputs[]
Definition: af_aecho.c:370
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
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
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
const char * name
Filter name.
Definition: avfilter.h:148
int64_t next_pts
Definition: af_aecho.c:41
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
static av_cold int init(AVFilterContext *ctx)
Definition: af_aecho.c:102
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
static int activate(AVFilterContext *ctx)
Definition: af_aecho.c:329
static int64_t pts
char * delays
Definition: af_aecho.c:33
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
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
Definition: avstring.c:184
float * delay
Definition: af_aecho.c:34
FF_FILTER_FORWARD_WANTED(outlink, inlink)
#define A
Definition: af_aecho.c:49
static const AVFilterPad aecho_inputs[]
Definition: af_aecho.c:362
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
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
formats
Definition: signature.h:48
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
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:409
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
float * decay
Definition: af_aecho.c:34
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
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