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;
50  uint8_t **chorusbuf;
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 
152  s->next_pts = AV_NOPTS_VALUE;
153 
154  return 0;
155 }
156 
157 static int config_output(AVFilterLink *outlink)
158 {
159  AVFilterContext *ctx = outlink->src;
160  ChorusContext *s = ctx->priv;
161  float sum_in_volume = 1.0;
162  int n;
163 
164  s->channels = outlink->channels;
165 
166  for (n = 0; n < s->num_chorus; n++) {
167  int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
168  int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
169 
170  s->length[n] = outlink->sample_rate / s->speeds[n];
171 
172  s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
173  if (!s->lookup_table[n])
174  return AVERROR(ENOMEM);
175 
177  s->length[n], 0., depth_samples, 0);
178  s->max_samples = FFMAX(s->max_samples, samples);
179  }
180 
181  for (n = 0; n < s->num_chorus; n++)
182  sum_in_volume += s->decays[n];
183 
184  if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
185  av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
186 
187  s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
188  if (!s->counter)
189  return AVERROR(ENOMEM);
190 
191  s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
192  if (!s->phase)
193  return AVERROR(ENOMEM);
194 
195  for (n = 0; n < outlink->channels; n++) {
196  s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
197  if (!s->phase[n])
198  return AVERROR(ENOMEM);
199  }
200 
201  s->fade_out = s->max_samples;
202 
203  return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
204  outlink->channels,
205  s->max_samples,
206  outlink->format, 0);
207 }
208 
209 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
210 
212 {
213  AVFilterContext *ctx = inlink->dst;
214  ChorusContext *s = ctx->priv;
215  AVFrame *out_frame;
216  int c, i, n;
217 
219  out_frame = frame;
220  } else {
221  out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
222  if (!out_frame) {
224  return AVERROR(ENOMEM);
225  }
226  av_frame_copy_props(out_frame, frame);
227  }
228 
229  for (c = 0; c < inlink->channels; c++) {
230  const float *src = (const float *)frame->extended_data[c];
231  float *dst = (float *)out_frame->extended_data[c];
232  float *chorusbuf = (float *)s->chorusbuf[c];
233  int *phase = s->phase[c];
234 
235  for (i = 0; i < frame->nb_samples; i++) {
236  float out, in = src[i];
237 
238  out = in * s->in_gain;
239 
240  for (n = 0; n < s->num_chorus; n++) {
241  out += chorusbuf[MOD(s->max_samples + s->counter[c] -
242  s->lookup_table[n][phase[n]],
243  s->max_samples)] * s->decays[n];
244  phase[n] = MOD(phase[n] + 1, s->length[n]);
245  }
246 
247  out *= s->out_gain;
248 
249  dst[i] = out;
250 
251  chorusbuf[s->counter[c]] = in;
252  s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
253  }
254  }
255 
256  s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
257 
258  if (frame != out_frame)
260 
261  return ff_filter_frame(ctx->outputs[0], out_frame);
262 }
263 
264 static int request_frame(AVFilterLink *outlink)
265 {
266  AVFilterContext *ctx = outlink->src;
267  ChorusContext *s = ctx->priv;
268  int ret;
269 
270  ret = ff_request_frame(ctx->inputs[0]);
271 
272  if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
273  int nb_samples = FFMIN(s->fade_out, 2048);
274  AVFrame *frame;
275 
276  frame = ff_get_audio_buffer(outlink, nb_samples);
277  if (!frame)
278  return AVERROR(ENOMEM);
279  s->fade_out -= nb_samples;
280 
281  av_samples_set_silence(frame->extended_data, 0,
282  frame->nb_samples,
283  outlink->channels,
284  frame->format);
285 
286  frame->pts = s->next_pts;
287  if (s->next_pts != AV_NOPTS_VALUE)
288  s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
289 
290  ret = filter_frame(ctx->inputs[0], frame);
291  }
292 
293  return ret;
294 }
295 
297 {
298  ChorusContext *s = ctx->priv;
299  int n;
300 
301  av_freep(&s->delays);
302  av_freep(&s->decays);
303  av_freep(&s->speeds);
304  av_freep(&s->depths);
305 
306  if (s->chorusbuf)
307  av_freep(&s->chorusbuf[0]);
308  av_freep(&s->chorusbuf);
309 
310  if (s->phase)
311  for (n = 0; n < s->channels; n++)
312  av_freep(&s->phase[n]);
313  av_freep(&s->phase);
314 
315  av_freep(&s->counter);
316  av_freep(&s->length);
317 
318  if (s->lookup_table)
319  for (n = 0; n < s->num_chorus; n++)
320  av_freep(&s->lookup_table[n]);
321  av_freep(&s->lookup_table);
322 }
323 
324 static const AVFilterPad chorus_inputs[] = {
325  {
326  .name = "default",
327  .type = AVMEDIA_TYPE_AUDIO,
328  .filter_frame = filter_frame,
329  },
330 };
331 
332 static const AVFilterPad chorus_outputs[] = {
333  {
334  .name = "default",
335  .type = AVMEDIA_TYPE_AUDIO,
336  .request_frame = request_frame,
337  .config_props = config_output,
338  },
339 };
340 
342  .name = "chorus",
343  .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
344  .priv_size = sizeof(ChorusContext),
345  .priv_class = &chorus_class,
346  .init = init,
347  .uninit = uninit,
351 };
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:88
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
ChorusContext::phase
int ** phase
Definition: af_chorus.c:51
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:186
AVERROR
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
opt.h
out
FILE * out
Definition: movenc.c:54
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1018
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
FILTER_SINGLE_SAMPLEFMT
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: internal.h:184
ChorusContext::chorusbuf
uint8_t ** chorusbuf
Definition: af_chorus.c:50
inlink
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
Definition: filter_design.txt:212
ChorusContext::modulation
int modulation
Definition: af_chorus.c:58
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:112
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_chorus.c:211
AVOption
AVOption.
Definition: opt.h:247
ChorusContext::channels
int channels
Definition: af_chorus.c:57
ff_request_frame
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:420
ChorusContext::speeds_str
char * speeds_str
Definition: af_chorus.c:44
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_chorus.c:296
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:153
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
fill_items
static void fill_items(char *item_str, int *nb_items, float *items)
Definition: af_chorus.c:90
count_items
static void count_items(char *item_str, int *nb_items)
Definition: af_chorus.c:78
MOD
#define MOD(a, b)
Definition: af_chorus.c:209
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:50
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
ChorusContext::depths_str
char * depths_str
Definition: af_chorus.c:45
av_cold
#define av_cold
Definition: attributes.h:90
ChorusContext::fade_out
int fade_out
Definition: af_chorus.c:59
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(chorus)
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_strtok
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:186
ChorusContext::delays_str
char * delays_str
Definition: af_chorus.c:42
ff_af_chorus
const AVFilter ff_af_chorus
Definition: af_chorus.c:341
ctx
AVFormatContext * ctx
Definition: movenc.c:48
ChorusContext::out_gain
float out_gain
Definition: af_chorus.c:41
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:141
ChorusContext::lookup_table
int32_t ** lookup_table
Definition: af_chorus.c:53
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: internal.h:191
WAVE_SIN
@ WAVE_SIN
Definition: generate_wave_table.h:25
ChorusContext::decays_str
char * decays_str
Definition: af_chorus.c:43
ChorusContext::in_gain
float in_gain
Definition: af_chorus.c:41
if
if(ret)
Definition: filter_design.txt:179
av_realloc_f
#define av_realloc_f(p, o, n)
Definition: tableprint_vlc.h:33
ChorusContext
Definition: af_chorus.c:39
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
NULL
#define NULL
Definition: coverity.c:32
av_frame_copy_props
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:537
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_chorus.c:157
src
#define src
Definition: vp8dsp.c:255
ChorusContext::next_pts
int64_t next_pts
Definition: af_chorus.c:60
c
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
ff_generate_wave_table
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
Definition: generate_wave_table.c:24
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
chorus_inputs
static const AVFilterPad chorus_inputs[]
Definition: af_chorus.c:324
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
A
#define A
Definition: af_chorus.c:64
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:473
ChorusContext::delays
float * delays
Definition: af_chorus.c:46
OFFSET
#define OFFSET(x)
Definition: af_chorus.c:63
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:227
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
chorus_outputs
static const AVFilterPad chorus_outputs[]
Definition: af_chorus.c:332
chorus_options
static const AVOption chorus_options[]
Definition: af_chorus.c:66
ChorusContext::counter
int * counter
Definition: af_chorus.c:54
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:350
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:56
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:271
av_samples_set_silence
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:244
AVFilter
Filter definition.
Definition: avfilter.h:149
ret
ret
Definition: filter_design.txt:187
frame
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
Definition: filter_design.txt:264
ChorusContext::speeds
float * speeds
Definition: af_chorus.c:48
ChorusContext::length
int * length
Definition: af_chorus.c:52
request_frame
static int request_frame(AVFilterLink *outlink)
Definition: af_chorus.c:264
ChorusContext::max_samples
int max_samples
Definition: af_chorus.c:56
generate_wave_table.h
avfilter.h
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
ChorusContext::depths
float * depths
Definition: af_chorus.c:49
AVFilterContext
An instance of a filter.
Definition: avfilter.h:386
audio.h
av_samples_alloc_array_and_samples
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:205
ChorusContext::decays
float * decays
Definition: af_chorus.c:47
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_chorus.c:106
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:192
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
int32_t
int32_t
Definition: audioconvert.c:56
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Definition: opt.h:228
int
int
Definition: ffmpeg_filter.c:156
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:62
ChorusContext::num_chorus
int num_chorus
Definition: af_chorus.c:55