FFmpeg
af_chorus.c
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1 /*
2  * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
3  * This source code is freely redistributable and may be used for
4  * any purpose. This copyright notice must be maintained.
5  * Juergen Mueller And Sundry Contributors are not responsible for
6  * the consequences of using this software.
7  *
8  * Copyright (c) 2015 Paul B Mahol
9  *
10  * This file is part of FFmpeg.
11  *
12  * FFmpeg is free software; you can redistribute it and/or
13  * modify it under the terms of the GNU Lesser General Public
14  * License as published by the Free Software Foundation; either
15  * version 2.1 of the License, or (at your option) any later version.
16  *
17  * FFmpeg is distributed in the hope that it will be useful,
18  * but WITHOUT ANY WARRANTY; without even the implied warranty of
19  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20  * Lesser General Public License for more details.
21  *
22  * You should have received a copy of the GNU Lesser General Public
23  * License along with FFmpeg; if not, write to the Free Software
24  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25  */
26 
27 /**
28  * @file
29  * chorus audio filter
30  */
31 
32 #include "libavutil/avstring.h"
33 #include "libavutil/opt.h"
34 #include "audio.h"
35 #include "avfilter.h"
36 #include "internal.h"
37 #include "generate_wave_table.h"
38 
39 typedef struct ChorusContext {
40  const AVClass *class;
41  float in_gain, out_gain;
42  char *delays_str;
43  char *decays_str;
44  char *speeds_str;
45  char *depths_str;
46  float *delays;
47  float *decays;
48  float *speeds;
49  float *depths;
51  int **phase;
52  int *length;
54  int *counter;
57  int channels;
59  int fade_out;
60  int64_t next_pts;
62 
63 #define OFFSET(x) offsetof(ChorusContext, x)
64 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
65 
66 static const AVOption chorus_options[] = {
67  { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
68  { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
69  { "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
70  { "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
71  { "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
72  { "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
73  { NULL }
74 };
75 
76 AVFILTER_DEFINE_CLASS(chorus);
77 
78 static void count_items(char *item_str, int *nb_items)
79 {
80  char *p;
81 
82  *nb_items = 1;
83  for (p = item_str; *p; p++) {
84  if (*p == '|')
85  (*nb_items)++;
86  }
87 
88 }
89 
90 static void fill_items(char *item_str, int *nb_items, float *items)
91 {
92  char *p, *saveptr = NULL;
93  int i, new_nb_items = 0;
94 
95  p = item_str;
96  for (i = 0; i < *nb_items; i++) {
97  char *tstr = av_strtok(p, "|", &saveptr);
98  p = NULL;
99  if (tstr)
100  new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;
101  }
102 
103  *nb_items = new_nb_items;
104 }
105 
107 {
108  ChorusContext *s = ctx->priv;
109  int nb_delays, nb_decays, nb_speeds, nb_depths;
110 
111  if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
112  av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
113  return AVERROR(EINVAL);
114  }
115 
116  count_items(s->delays_str, &nb_delays);
117  count_items(s->decays_str, &nb_decays);
118  count_items(s->speeds_str, &nb_speeds);
119  count_items(s->depths_str, &nb_depths);
120 
121  s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
122  s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
123  s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
124  s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
125 
126  if (!s->delays || !s->decays || !s->speeds || !s->depths)
127  return AVERROR(ENOMEM);
128 
129  fill_items(s->delays_str, &nb_delays, s->delays);
130  fill_items(s->decays_str, &nb_decays, s->decays);
131  fill_items(s->speeds_str, &nb_speeds, s->speeds);
132  fill_items(s->depths_str, &nb_depths, s->depths);
133 
134  if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
135  av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
136  return AVERROR(EINVAL);
137  }
138 
139  s->num_chorus = nb_delays;
140 
141  if (s->num_chorus < 1) {
142  av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
143  return AVERROR(EINVAL);
144  }
145 
146  s->length = av_calloc(s->num_chorus, sizeof(*s->length));
147  s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
148 
149  if (!s->length || !s->lookup_table)
150  return AVERROR(ENOMEM);
151 
153 
154  return 0;
155 }
156 
158 {
161  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 static int config_output(AVFilterLink *outlink)
187 {
188  AVFilterContext *ctx = outlink->src;
189  ChorusContext *s = ctx->priv;
190  float sum_in_volume = 1.0;
191  int n;
192 
193  s->channels = outlink->channels;
194 
195  for (n = 0; n < s->num_chorus; n++) {
196  int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
197  int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
198 
199  s->length[n] = outlink->sample_rate / s->speeds[n];
200 
201  s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
202  if (!s->lookup_table[n])
203  return AVERROR(ENOMEM);
204 
206  s->length[n], 0., depth_samples, 0);
207  s->max_samples = FFMAX(s->max_samples, samples);
208  }
209 
210  for (n = 0; n < s->num_chorus; n++)
211  sum_in_volume += s->decays[n];
212 
213  if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
214  av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
215 
216  s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
217  if (!s->counter)
218  return AVERROR(ENOMEM);
219 
220  s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
221  if (!s->phase)
222  return AVERROR(ENOMEM);
223 
224  for (n = 0; n < outlink->channels; n++) {
225  s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
226  if (!s->phase[n])
227  return AVERROR(ENOMEM);
228  }
229 
230  s->fade_out = s->max_samples;
231 
233  outlink->channels,
234  s->max_samples,
235  outlink->format, 0);
236 }
237 
238 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
239 
241 {
242  AVFilterContext *ctx = inlink->dst;
243  ChorusContext *s = ctx->priv;
244  AVFrame *out_frame;
245  int c, i, n;
246 
247  if (av_frame_is_writable(frame)) {
248  out_frame = frame;
249  } else {
250  out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
251  if (!out_frame) {
252  av_frame_free(&frame);
253  return AVERROR(ENOMEM);
254  }
255  av_frame_copy_props(out_frame, frame);
256  }
257 
258  for (c = 0; c < inlink->channels; c++) {
259  const float *src = (const float *)frame->extended_data[c];
260  float *dst = (float *)out_frame->extended_data[c];
261  float *chorusbuf = (float *)s->chorusbuf[c];
262  int *phase = s->phase[c];
263 
264  for (i = 0; i < frame->nb_samples; i++) {
265  float out, in = src[i];
266 
267  out = in * s->in_gain;
268 
269  for (n = 0; n < s->num_chorus; n++) {
270  out += chorusbuf[MOD(s->max_samples + s->counter[c] -
271  s->lookup_table[n][phase[n]],
272  s->max_samples)] * s->decays[n];
273  phase[n] = MOD(phase[n] + 1, s->length[n]);
274  }
275 
276  out *= s->out_gain;
277 
278  dst[i] = out;
279 
280  chorusbuf[s->counter[c]] = in;
281  s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
282  }
283  }
284 
285  s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
286 
287  if (frame != out_frame)
288  av_frame_free(&frame);
289 
290  return ff_filter_frame(ctx->outputs[0], out_frame);
291 }
292 
293 static int request_frame(AVFilterLink *outlink)
294 {
295  AVFilterContext *ctx = outlink->src;
296  ChorusContext *s = ctx->priv;
297  int ret;
298 
299  ret = ff_request_frame(ctx->inputs[0]);
300 
301  if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
302  int nb_samples = FFMIN(s->fade_out, 2048);
303  AVFrame *frame;
304 
305  frame = ff_get_audio_buffer(outlink, nb_samples);
306  if (!frame)
307  return AVERROR(ENOMEM);
308  s->fade_out -= nb_samples;
309 
311  frame->nb_samples,
312  outlink->channels,
313  frame->format);
314 
315  frame->pts = s->next_pts;
316  if (s->next_pts != AV_NOPTS_VALUE)
317  s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
318 
319  ret = filter_frame(ctx->inputs[0], frame);
320  }
321 
322  return ret;
323 }
324 
326 {
327  ChorusContext *s = ctx->priv;
328  int n;
329 
330  av_freep(&s->delays);
331  av_freep(&s->decays);
332  av_freep(&s->speeds);
333  av_freep(&s->depths);
334 
335  if (s->chorusbuf)
336  av_freep(&s->chorusbuf[0]);
337  av_freep(&s->chorusbuf);
338 
339  if (s->phase)
340  for (n = 0; n < s->channels; n++)
341  av_freep(&s->phase[n]);
342  av_freep(&s->phase);
343 
344  av_freep(&s->counter);
345  av_freep(&s->length);
346 
347  if (s->lookup_table)
348  for (n = 0; n < s->num_chorus; n++)
349  av_freep(&s->lookup_table[n]);
350  av_freep(&s->lookup_table);
351 }
352 
353 static const AVFilterPad chorus_inputs[] = {
354  {
355  .name = "default",
356  .type = AVMEDIA_TYPE_AUDIO,
357  .filter_frame = filter_frame,
358  },
359  { NULL }
360 };
361 
362 static const AVFilterPad chorus_outputs[] = {
363  {
364  .name = "default",
365  .type = AVMEDIA_TYPE_AUDIO,
366  .request_frame = request_frame,
367  .config_props = config_output,
368  },
369  { NULL }
370 };
371 
373  .name = "chorus",
374  .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
375  .query_formats = query_formats,
376  .priv_size = sizeof(ChorusContext),
377  .priv_class = &chorus_class,
378  .init = init,
379  .uninit = uninit,
380  .inputs = chorus_inputs,
381  .outputs = chorus_outputs,
382 };
float, planar
Definition: samplefmt.h:69
char * speeds_str
Definition: af_chorus.c:44
#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.
static void count_items(char *item_str, int *nb_items)
Definition: af_chorus.c:78
int * length
Definition: af_chorus.c:52
#define src
Definition: vp8dsp.c:254
int is_disabled
the enabled state from the last expression evaluation
Definition: avfilter.h:385
AVFilter ff_af_chorus
Definition: af_chorus.c:372
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
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
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
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
char * depths_str
Definition: af_chorus.c:45
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
AVOptions.
static int query_formats(AVFilterContext *ctx)
Definition: af_chorus.c:157
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
#define AVERROR_EOF
End of file.
Definition: error.h:55
float * delays
Definition: af_chorus.c:46
signed 32 bits
Definition: samplefmt.h:62
#define av_log(a,...)
uint8_t ** chorusbuf
Definition: af_chorus.c:50
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
AVFILTER_DEFINE_CLASS(chorus)
#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
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
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_chorus.c:325
int64_t next_pts
Definition: af_chorus.c:60
static const AVFilterPad chorus_inputs[]
Definition: af_chorus.c:353
#define MOD(a, b)
Definition: af_chorus.c:238
#define FFMAX(a, b)
Definition: common.h:94
float * decays
Definition: af_chorus.c:47
int num_chorus
Definition: af_chorus.c:55
char * decays_str
Definition: af_chorus.c:43
float * depths
Definition: af_chorus.c:49
#define FFMIN(a, b)
Definition: common.h:96
int32_t ** lookup_table
Definition: af_chorus.c:53
static int request_frame(AVFilterLink *outlink)
Definition: af_chorus.c:293
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
int n
Definition: avisynth_c.h:760
char * delays_str
Definition: af_chorus.c:42
#define A
Definition: af_chorus.c:64
int ** phase
Definition: af_chorus.c:51
static int config_output(AVFilterLink *outlink)
Definition: af_chorus.c:186
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
A list of supported channel layouts.
Definition: formats.h:85
if(ret)
int modulation
Definition: af_chorus.c:58
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:368
float * speeds
Definition: af_chorus.c:48
static av_cold int init(AVFilterContext *ctx)
Definition: af_chorus.c:106
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 AVOption chorus_options[]
Definition: af_chorus.c:66
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
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
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
int * counter
Definition: af_chorus.c:54
int
A list of supported formats for one end of a filter link.
Definition: formats.h:64
#define OFFSET(x)
Definition: af_chorus.c:63
An instance of a filter.
Definition: avfilter.h:338
float in_gain
Definition: af_chorus.c:41
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
Filter the word “frame” indicates either a video frame or a group of audio samples
#define av_freep(p)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:407
formats
Definition: signature.h:48
static const AVFilterPad chorus_outputs[]
Definition: af_chorus.c:362
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
int max_samples
Definition: af_chorus.c:56
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_chorus.c:240
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
for(j=16;j >0;--j)
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
static void fill_items(char *item_str, int *nb_items, float *items)
Definition: af_chorus.c:90
float out_gain
Definition: af_chorus.c:41