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/mem.h"
34 #include "libavutil/opt.h"
35 #include "audio.h"
36 #include "avfilter.h"
37 #include "filters.h"
38 #include "generate_wave_table.h"
39 
40 typedef struct ChorusContext {
41  const AVClass *class;
42  float in_gain, out_gain;
43  char *delays_str;
44  char *decays_str;
45  char *speeds_str;
46  char *depths_str;
47  float *delays;
48  float *decays;
49  float *speeds;
50  float *depths;
51  uint8_t **chorusbuf;
52  int **phase;
53  int *length;
55  int *counter;
58  int channels;
60  int fade_out;
63 
64 #define OFFSET(x) offsetof(ChorusContext, x)
65 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
66 
67 static const AVOption chorus_options[] = {
68  { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
69  { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
70  { "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
71  { "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
72  { "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
73  { "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
74  { NULL }
75 };
76 
77 AVFILTER_DEFINE_CLASS(chorus);
78 
79 static void count_items(char *item_str, int *nb_items)
80 {
81  char *p;
82 
83  *nb_items = 1;
84  for (p = item_str; *p; p++) {
85  if (*p == '|')
86  (*nb_items)++;
87  }
88 
89 }
90 
91 static void fill_items(char *item_str, int *nb_items, float *items)
92 {
93  char *p, *saveptr = NULL;
94  int i, new_nb_items = 0;
95 
96  p = item_str;
97  for (i = 0; i < *nb_items; i++) {
98  char *tstr = av_strtok(p, "|", &saveptr);
99  p = NULL;
100  if (tstr)
101  new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;
102  }
103 
104  *nb_items = new_nb_items;
105 }
106 
108 {
109  ChorusContext *s = ctx->priv;
110  int nb_delays, nb_decays, nb_speeds, nb_depths;
111 
112  if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
113  av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
114  return AVERROR(EINVAL);
115  }
116 
117  count_items(s->delays_str, &nb_delays);
118  count_items(s->decays_str, &nb_decays);
119  count_items(s->speeds_str, &nb_speeds);
120  count_items(s->depths_str, &nb_depths);
121 
122  s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
123  s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
124  s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
125  s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
126 
127  if (!s->delays || !s->decays || !s->speeds || !s->depths)
128  return AVERROR(ENOMEM);
129 
130  fill_items(s->delays_str, &nb_delays, s->delays);
131  fill_items(s->decays_str, &nb_decays, s->decays);
132  fill_items(s->speeds_str, &nb_speeds, s->speeds);
133  fill_items(s->depths_str, &nb_depths, s->depths);
134 
135  if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
136  av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
137  return AVERROR(EINVAL);
138  }
139 
140  s->num_chorus = nb_delays;
141 
142  if (s->num_chorus < 1) {
143  av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
144  return AVERROR(EINVAL);
145  }
146 
147  s->length = av_calloc(s->num_chorus, sizeof(*s->length));
148  s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
149 
150  if (!s->length || !s->lookup_table)
151  return AVERROR(ENOMEM);
152 
153  s->next_pts = AV_NOPTS_VALUE;
154 
155  return 0;
156 }
157 
158 static int config_output(AVFilterLink *outlink)
159 {
160  AVFilterContext *ctx = outlink->src;
161  ChorusContext *s = ctx->priv;
162  float sum_in_volume = 1.0;
163  int n;
164 
165  s->channels = outlink->ch_layout.nb_channels;
166 
167  for (n = 0; n < s->num_chorus; n++) {
168  int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
169  int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
170 
171  s->length[n] = outlink->sample_rate / s->speeds[n];
172 
173  s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
174  if (!s->lookup_table[n])
175  return AVERROR(ENOMEM);
176 
178  s->length[n], 0., depth_samples, 0);
179  s->max_samples = FFMAX(s->max_samples, samples);
180  }
181 
182  for (n = 0; n < s->num_chorus; n++)
183  sum_in_volume += s->decays[n];
184 
185  if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
186  av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
187 
188  s->counter = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->counter));
189  if (!s->counter)
190  return AVERROR(ENOMEM);
191 
192  s->phase = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->phase));
193  if (!s->phase)
194  return AVERROR(ENOMEM);
195 
196  for (n = 0; n < outlink->ch_layout.nb_channels; n++) {
197  s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
198  if (!s->phase[n])
199  return AVERROR(ENOMEM);
200  }
201 
202  s->fade_out = s->max_samples;
203 
204  return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
205  outlink->ch_layout.nb_channels,
206  s->max_samples,
207  outlink->format, 0);
208 }
209 
210 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
211 
213 {
214  AVFilterContext *ctx = inlink->dst;
215  ChorusContext *s = ctx->priv;
216  AVFrame *out_frame;
217  int c, i, n;
218 
220  out_frame = frame;
221  } else {
222  out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
223  if (!out_frame) {
225  return AVERROR(ENOMEM);
226  }
227  av_frame_copy_props(out_frame, frame);
228  }
229 
230  for (c = 0; c < inlink->ch_layout.nb_channels; c++) {
231  const float *src = (const float *)frame->extended_data[c];
232  float *dst = (float *)out_frame->extended_data[c];
233  float *chorusbuf = (float *)s->chorusbuf[c];
234  int *phase = s->phase[c];
235 
236  for (i = 0; i < frame->nb_samples; i++) {
237  float out, in = src[i];
238 
239  out = in * s->in_gain;
240 
241  for (n = 0; n < s->num_chorus; n++) {
242  out += chorusbuf[MOD(s->max_samples + s->counter[c] -
243  s->lookup_table[n][phase[n]],
244  s->max_samples)] * s->decays[n];
245  phase[n] = MOD(phase[n] + 1, s->length[n]);
246  }
247 
248  out *= s->out_gain;
249 
250  dst[i] = out;
251 
252  chorusbuf[s->counter[c]] = in;
253  s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
254  }
255  }
256 
257  s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
258 
259  if (frame != out_frame)
261 
262  return ff_filter_frame(ctx->outputs[0], out_frame);
263 }
264 
265 static int request_frame(AVFilterLink *outlink)
266 {
267  AVFilterContext *ctx = outlink->src;
268  ChorusContext *s = ctx->priv;
269  int ret;
270 
271  ret = ff_request_frame(ctx->inputs[0]);
272 
273  if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
274  int nb_samples = FFMIN(s->fade_out, 2048);
275  AVFrame *frame;
276 
277  frame = ff_get_audio_buffer(outlink, nb_samples);
278  if (!frame)
279  return AVERROR(ENOMEM);
280  s->fade_out -= nb_samples;
281 
282  av_samples_set_silence(frame->extended_data, 0,
283  frame->nb_samples,
284  outlink->ch_layout.nb_channels,
285  frame->format);
286 
287  frame->pts = s->next_pts;
288  if (s->next_pts != AV_NOPTS_VALUE)
289  s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
290 
291  ret = filter_frame(ctx->inputs[0], frame);
292  }
293 
294  return ret;
295 }
296 
298 {
299  ChorusContext *s = ctx->priv;
300  int n;
301 
302  av_freep(&s->delays);
303  av_freep(&s->decays);
304  av_freep(&s->speeds);
305  av_freep(&s->depths);
306 
307  if (s->chorusbuf)
308  av_freep(&s->chorusbuf[0]);
309  av_freep(&s->chorusbuf);
310 
311  if (s->phase)
312  for (n = 0; n < s->channels; n++)
313  av_freep(&s->phase[n]);
314  av_freep(&s->phase);
315 
316  av_freep(&s->counter);
317  av_freep(&s->length);
318 
319  if (s->lookup_table)
320  for (n = 0; n < s->num_chorus; n++)
321  av_freep(&s->lookup_table[n]);
322  av_freep(&s->lookup_table);
323 }
324 
325 static const AVFilterPad chorus_inputs[] = {
326  {
327  .name = "default",
328  .type = AVMEDIA_TYPE_AUDIO,
329  .filter_frame = filter_frame,
330  },
331 };
332 
333 static const AVFilterPad chorus_outputs[] = {
334  {
335  .name = "default",
336  .type = AVMEDIA_TYPE_AUDIO,
337  .request_frame = request_frame,
338  .config_props = config_output,
339  },
340 };
341 
343  .name = "chorus",
344  .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
345  .priv_size = sizeof(ChorusContext),
346  .priv_class = &chorus_class,
347  .init = init,
348  .uninit = uninit,
352 };
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:98
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
ChorusContext::phase
int ** phase
Definition: af_chorus.c:52
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:215
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:55
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1061
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
ChorusContext::chorusbuf
uint8_t ** chorusbuf
Definition: af_chorus.c:51
int64_t
long long int64_t
Definition: coverity.c:34
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:59
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:162
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: filters.h:262
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
av_samples_set_silence
int av_samples_set_silence(uint8_t *const *audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:246
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_chorus.c:212
AVOption
AVOption.
Definition: opt.h:429
ChorusContext::channels
int channels
Definition: af_chorus.c:58
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:475
ChorusContext::speeds_str
char * speeds_str
Definition: af_chorus.c:45
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_chorus.c:297
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:205
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:327
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:30
fill_items
static void fill_items(char *item_str, int *nb_items, float *items)
Definition: af_chorus.c:91
count_items
static void count_items(char *item_str, int *nb_items)
Definition: af_chorus.c:79
MOD
#define MOD(a, b)
Definition: af_chorus.c:210
AVFilterPad
A filter pad used for either input or output.
Definition: filters.h:38
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:209
ChorusContext::depths_str
char * depths_str
Definition: af_chorus.c:46
av_cold
#define av_cold
Definition: attributes.h:90
ChorusContext::fade_out
int fade_out
Definition: af_chorus.c:60
s
#define s(width, name)
Definition: cbs_vp9.c:198
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:178
filters.h
ChorusContext::delays_str
char * delays_str
Definition: af_chorus.c:43
ff_af_chorus
const AVFilter ff_af_chorus
Definition: af_chorus.c:342
ctx
AVFormatContext * ctx
Definition: movenc.c:49
ChorusContext::out_gain
float out_gain
Definition: af_chorus.c:42
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:142
ChorusContext::lookup_table
int32_t ** lookup_table
Definition: af_chorus.c:54
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: filters.h:263
WAVE_SIN
@ WAVE_SIN
Definition: generate_wave_table.h:25
ChorusContext::decays_str
char * decays_str
Definition: af_chorus.c:44
ChorusContext::in_gain
float in_gain
Definition: af_chorus.c:42
if
if(ret)
Definition: filter_design.txt:179
av_realloc_f
#define av_realloc_f(p, o, n)
Definition: tableprint_vlc.h:32
ChorusContext
Definition: af_chorus.c:40
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:75
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:725
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_chorus.c:158
ChorusContext::next_pts
int64_t next_pts
Definition: af_chorus.c:61
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
chorus_inputs
static const AVFilterPad chorus_inputs[]
Definition: af_chorus.c:325
FILTER_SINGLE_SAMPLEFMT
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: filters.h:255
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:94
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
A
#define A
Definition: af_chorus.c:65
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:661
ChorusContext::delays
float * delays
Definition: af_chorus.c:47
OFFSET
#define OFFSET(x)
Definition: af_chorus.c:64
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Underlying C type is float.
Definition: opt.h:271
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
chorus_outputs
static const AVFilterPad chorus_outputs[]
Definition: af_chorus.c:333
chorus_options
static const AVOption chorus_options[]
Definition: af_chorus.c:67
ChorusContext::counter
int * counter
Definition: af_chorus.c:55
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:450
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVFilterPad::name
const char * name
Pad name.
Definition: filters.h:44
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
AVFilter
Filter definition.
Definition: avfilter.h:201
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:49
ChorusContext::length
int * length
Definition: af_chorus.c:53
request_frame
static int request_frame(AVFilterLink *outlink)
Definition: af_chorus.c:265
ChorusContext::max_samples
int max_samples
Definition: af_chorus.c:57
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:50
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
mem.h
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:207
ChorusContext::decays
float * decays
Definition: af_chorus.c:48
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_chorus.c:107
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
int32_t
int32_t
Definition: audioconvert.c:56
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Underlying C type is a uint8_t* that is either NULL or points to a C string allocated with the av_mal...
Definition: opt.h:276
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:59
src
#define src
Definition: vp8dsp.c:248
ChorusContext::num_chorus
int num_chorus
Definition: af_chorus.c:56