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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 += 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 
273 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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)
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:226
#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
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
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
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:319
static AVFrame * frame
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 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
#define AVERROR(e)
Definition: error.h:43
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
#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
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
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:299
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
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
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
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:273
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
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