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 "internal.h"
28 
29 typedef struct AudioEchoContext {
30  const AVClass *class;
31  float in_gain, out_gain;
32  char *delays, *decays;
33  float *delay, *decay;
34  int nb_echoes;
38  int *samples;
39  int64_t next_pts;
40 
42  uint8_t * const *src, uint8_t **dst,
43  int nb_samples, int channels);
45 
46 #define OFFSET(x) offsetof(AudioEchoContext, x)
47 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
48 
49 static const AVOption aecho_options[] = {
50  { "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
51  { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
52  { "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
53  { "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
54  { NULL }
55 };
56 
58 
59 static void count_items(char *item_str, int *nb_items)
60 {
61  char *p;
62 
63  *nb_items = 1;
64  for (p = item_str; *p; p++) {
65  if (*p == '|')
66  (*nb_items)++;
67  }
68 
69 }
70 
71 static void fill_items(char *item_str, int *nb_items, float *items)
72 {
73  char *p, *saveptr = NULL;
74  int i, new_nb_items = 0;
75 
76  p = item_str;
77  for (i = 0; i < *nb_items; i++) {
78  char *tstr = av_strtok(p, "|", &saveptr);
79  p = NULL;
80  if (tstr)
81  new_nb_items += av_sscanf(tstr, "%f", &items[new_nb_items]) == 1;
82  }
83 
84  *nb_items = new_nb_items;
85 }
86 
88 {
89  AudioEchoContext *s = ctx->priv;
90 
91  av_freep(&s->delay);
92  av_freep(&s->decay);
93  av_freep(&s->samples);
94 
95  if (s->delayptrs)
96  av_freep(&s->delayptrs[0]);
97  av_freep(&s->delayptrs);
98 }
99 
101 {
102  AudioEchoContext *s = ctx->priv;
103  int nb_delays, nb_decays, i;
104 
105  if (!s->delays || !s->decays) {
106  av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
107  return AVERROR(EINVAL);
108  }
109 
110  count_items(s->delays, &nb_delays);
111  count_items(s->decays, &nb_decays);
112 
113  s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
114  s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
115  if (!s->delay || !s->decay)
116  return AVERROR(ENOMEM);
117 
118  fill_items(s->delays, &nb_delays, s->delay);
119  fill_items(s->decays, &nb_decays, s->decay);
120 
121  if (nb_delays != nb_decays) {
122  av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
123  return AVERROR(EINVAL);
124  }
125 
126  s->nb_echoes = nb_delays;
127  if (!s->nb_echoes) {
128  av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
129  return AVERROR(EINVAL);
130  }
131 
132  s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
133  if (!s->samples)
134  return AVERROR(ENOMEM);
135 
136  for (i = 0; i < nb_delays; i++) {
137  if (s->delay[i] <= 0 || s->delay[i] > 90000) {
138  av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
139  return AVERROR(EINVAL);
140  }
141  if (s->decay[i] <= 0 || s->decay[i] > 1) {
142  av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
143  return AVERROR(EINVAL);
144  }
145  }
146 
148 
149  av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
150  return 0;
151 }
152 
154 {
157  static const enum AVSampleFormat sample_fmts[] = {
161  };
162  int ret;
163 
164  layouts = ff_all_channel_counts();
165  if (!layouts)
166  return AVERROR(ENOMEM);
167  ret = ff_set_common_channel_layouts(ctx, layouts);
168  if (ret < 0)
169  return ret;
170 
171  formats = ff_make_format_list(sample_fmts);
172  if (!formats)
173  return AVERROR(ENOMEM);
174  ret = ff_set_common_formats(ctx, formats);
175  if (ret < 0)
176  return ret;
177 
178  formats = ff_all_samplerates();
179  if (!formats)
180  return AVERROR(ENOMEM);
181  return ff_set_common_samplerates(ctx, formats);
182 }
183 
184 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
185 
186 #define ECHO(name, type, min, max) \
187 static void echo_samples_## name ##p(AudioEchoContext *ctx, \
188  uint8_t **delayptrs, \
189  uint8_t * const *src, uint8_t **dst, \
190  int nb_samples, int channels) \
191 { \
192  const double out_gain = ctx->out_gain; \
193  const double in_gain = ctx->in_gain; \
194  const int nb_echoes = ctx->nb_echoes; \
195  const int max_samples = ctx->max_samples; \
196  int i, j, chan, av_uninit(index); \
197  \
198  av_assert1(channels > 0); /* would corrupt delay_index */ \
199  \
200  for (chan = 0; chan < channels; chan++) { \
201  const type *s = (type *)src[chan]; \
202  type *d = (type *)dst[chan]; \
203  type *dbuf = (type *)delayptrs[chan]; \
204  \
205  index = ctx->delay_index; \
206  for (i = 0; i < nb_samples; i++, s++, d++) { \
207  double out, in; \
208  \
209  in = *s; \
210  out = in * in_gain; \
211  for (j = 0; j < nb_echoes; j++) { \
212  int ix = index + max_samples - ctx->samples[j]; \
213  ix = MOD(ix, max_samples); \
214  out += dbuf[ix] * ctx->decay[j]; \
215  } \
216  out *= out_gain; \
217  \
218  *d = av_clipd(out, min, max); \
219  dbuf[index] = in; \
220  \
221  index = MOD(index + 1, max_samples); \
222  } \
223  } \
224  ctx->delay_index = index; \
225 }
226 
227 ECHO(dbl, double, -1.0, 1.0 )
228 ECHO(flt, float, -1.0, 1.0 )
229 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
230 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
231 
232 static int config_output(AVFilterLink *outlink)
233 {
234  AVFilterContext *ctx = outlink->src;
235  AudioEchoContext *s = ctx->priv;
236  float volume = 1.0;
237  int i;
238 
239  for (i = 0; i < s->nb_echoes; i++) {
240  s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
241  s->max_samples = FFMAX(s->max_samples, s->samples[i]);
242  volume += s->decay[i];
243  }
244 
245  if (s->max_samples <= 0) {
246  av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
247  return AVERROR(EINVAL);
248  }
249  s->fade_out = s->max_samples;
250 
251  if (volume * s->in_gain * s->out_gain > 1.0)
252  av_log(ctx, AV_LOG_WARNING,
253  "out_gain %f can cause saturation of output\n", s->out_gain);
254 
255  switch (outlink->format) {
256  case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
257  case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
258  case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
259  case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
260  }
261 
262 
263  if (s->delayptrs)
264  av_freep(&s->delayptrs[0]);
265  av_freep(&s->delayptrs);
266 
268  outlink->channels,
269  s->max_samples,
270  outlink->format, 0);
271 }
272 
274 {
275  AVFilterContext *ctx = inlink->dst;
276  AudioEchoContext *s = ctx->priv;
277  AVFrame *out_frame;
278 
279  if (av_frame_is_writable(frame)) {
280  out_frame = frame;
281  } else {
282  out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
283  if (!out_frame) {
284  av_frame_free(&frame);
285  return AVERROR(ENOMEM);
286  }
287  av_frame_copy_props(out_frame, frame);
288  }
289 
290  s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,
291  frame->nb_samples, inlink->channels);
292 
293  s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
294 
295  if (frame != out_frame)
296  av_frame_free(&frame);
297 
298  return ff_filter_frame(ctx->outputs[0], out_frame);
299 }
300 
301 static int request_frame(AVFilterLink *outlink)
302 {
303  AVFilterContext *ctx = outlink->src;
304  AudioEchoContext *s = ctx->priv;
305  int ret;
306 
307  ret = ff_request_frame(ctx->inputs[0]);
308 
309  if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
310  int nb_samples = FFMIN(s->fade_out, 2048);
311  AVFrame *frame;
312 
313  frame = ff_get_audio_buffer(outlink, nb_samples);
314  if (!frame)
315  return AVERROR(ENOMEM);
316  s->fade_out -= nb_samples;
317 
319  frame->nb_samples,
320  outlink->channels,
321  frame->format);
322 
323  s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,
324  frame->nb_samples, outlink->channels);
325 
326  frame->pts = s->next_pts;
327  if (s->next_pts != AV_NOPTS_VALUE)
328  s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
329 
330  return ff_filter_frame(outlink, frame);
331  }
332 
333  return ret;
334 }
335 
336 static const AVFilterPad aecho_inputs[] = {
337  {
338  .name = "default",
339  .type = AVMEDIA_TYPE_AUDIO,
340  .filter_frame = filter_frame,
341  },
342  { NULL }
343 };
344 
345 static const AVFilterPad aecho_outputs[] = {
346  {
347  .name = "default",
348  .request_frame = request_frame,
349  .config_props = config_output,
350  .type = AVMEDIA_TYPE_AUDIO,
351  },
352  { NULL }
353 };
354 
356  .name = "aecho",
357  .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
358  .query_formats = query_formats,
359  .priv_size = sizeof(AudioEchoContext),
360  .priv_class = &aecho_class,
361  .init = init,
362  .uninit = uninit,
363  .inputs = aecho_inputs,
364  .outputs = aecho_outputs,
365 };
float, planar
Definition: samplefmt.h:69
#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:549
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:59
double, planar
Definition: samplefmt.h:70
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_aecho.c:273
#define src
Definition: vp8dsp.c:254
int is_disabled
the enabled state from the last expression evaluation
Definition: avfilter.h:385
char * decays
Definition: af_aecho.c:32
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:71
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:87
float out_gain
Definition: af_aecho.c:31
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:41
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:1080
static const AVOption aecho_options[]
Definition: af_aecho.c:49
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
#define OFFSET(x)
Definition: af_aecho.c:46
uint8_t ** delayptrs
Definition: af_aecho.c:36
#define ECHO(name, type, min, max)
Definition: af_aecho.c:186
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:301
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define av_log(a,...)
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
#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:568
static int query_formats(AVFilterContext *ctx)
Definition: af_aecho.c:153
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:232
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:355
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:345
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
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
int64_t next_pts
Definition: af_aecho.c:39
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:100
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
char * delays
Definition: af_aecho.c:32
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:33
#define A
Definition: af_aecho.c:47
static const AVFilterPad aecho_inputs[]
Definition: af_aecho.c:336
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
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
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:410
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:33
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
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