FFmpeg
af_apulsator.c
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1 /*
2  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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/opt.h"
23 #include "avfilter.h"
24 #include "internal.h"
25 #include "audio.h"
26 
29 
30 typedef struct SimpleLFO {
31  double phase;
32  double freq;
33  double offset;
34  double amount;
35  double pwidth;
36  int mode;
37  int srate;
38 } SimpleLFO;
39 
40 typedef struct AudioPulsatorContext {
41  const AVClass *class;
42  int mode;
43  double level_in;
44  double level_out;
45  double amount;
46  double offset_l;
47  double offset_r;
48  double pwidth;
49  double bpm;
50  double hertz;
51  int ms;
52  int timing;
53 
56 
57 #define OFFSET(x) offsetof(AudioPulsatorContext, x)
58 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
59 
60 static const AVOption apulsator_options[] = {
61  { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
62  { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
63  { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, "mode" },
64  { "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, "mode" },
65  { "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, "mode" },
66  { "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, "mode" },
67  { "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, "mode" },
68  { "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, "mode" },
69  { "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS },
70  { "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS },
71  { "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
72  { "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS },
73  { "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, "timing" },
74  { "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, "timing" },
75  { "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, "timing" },
76  { "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, "timing" },
77  { "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS },
78  { "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS },
79  { "hz", "set frequency", OFFSET(hertz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS },
80  { NULL }
81 };
82 
83 AVFILTER_DEFINE_CLASS(apulsator);
84 
85 static void lfo_advance(SimpleLFO *lfo, unsigned count)
86 {
87  lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
88  if (lfo->phase >= 1)
89  lfo->phase = fmod(lfo->phase, 1);
90 }
91 
92 static double lfo_get_value(SimpleLFO *lfo)
93 {
94  double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
95  double val;
96 
97  if (phs > 1)
98  phs = fmod(phs, 1.);
99 
100  switch (lfo->mode) {
101  case SINE:
102  val = sin(phs * 2 * M_PI);
103  break;
104  case TRIANGLE:
105  if (phs > 0.75)
106  val = (phs - 0.75) * 4 - 1;
107  else if (phs > 0.25)
108  val = -4 * phs + 2;
109  else
110  val = phs * 4;
111  break;
112  case SQUARE:
113  val = phs < 0.5 ? -1 : +1;
114  break;
115  case SAWUP:
116  val = phs * 2 - 1;
117  break;
118  case SAWDOWN:
119  val = 1 - phs * 2;
120  break;
121  default: av_assert0(0);
122  }
123 
124  return val * lfo->amount;
125 }
126 
128 {
129  AVFilterContext *ctx = inlink->dst;
130  AVFilterLink *outlink = ctx->outputs[0];
131  AudioPulsatorContext *s = ctx->priv;
132  const double *src = (const double *)in->data[0];
133  const int nb_samples = in->nb_samples;
134  const double level_out = s->level_out;
135  const double level_in = s->level_in;
136  const double amount = s->amount;
137  AVFrame *out;
138  double *dst;
139  int n;
140 
142  out = in;
143  } else {
144  out = ff_get_audio_buffer(inlink, in->nb_samples);
145  if (!out) {
146  av_frame_free(&in);
147  return AVERROR(ENOMEM);
148  }
150  }
151  dst = (double *)out->data[0];
152 
153  for (n = 0; n < nb_samples; n++) {
154  double outL;
155  double outR;
156  double inL = src[0] * level_in;
157  double inR = src[1] * level_in;
158  double procL = inL;
159  double procR = inR;
160 
161  procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
162  procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
163 
164  outL = procL + inL * (1 - amount);
165  outR = procR + inR * (1 - amount);
166 
167  outL *= level_out;
168  outR *= level_out;
169 
170  dst[0] = outL;
171  dst[1] = outR;
172 
173  lfo_advance(&s->lfoL, 1);
174  lfo_advance(&s->lfoR, 1);
175 
176  dst += 2;
177  src += 2;
178  }
179 
180  if (in != out)
181  av_frame_free(&in);
182 
183  return ff_filter_frame(outlink, out);
184 }
185 
187 {
190  int ret;
191 
192  if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
193  (ret = ff_set_common_formats (ctx , formats )) < 0 ||
196  return ret;
197 
200 }
201 
203 {
204  AVFilterContext *ctx = inlink->dst;
205  AudioPulsatorContext *s = ctx->priv;
206  double freq;
207 
208  switch (s->timing) {
209  case UNIT_BPM: freq = s->bpm / 60; break;
210  case UNIT_MS: freq = 1 / (s->ms / 1000.); break;
211  case UNIT_HZ: freq = s->hertz; break;
212  default: av_assert0(0);
213  }
214 
215  s->lfoL.freq = freq;
216  s->lfoR.freq = freq;
217  s->lfoL.mode = s->mode;
218  s->lfoR.mode = s->mode;
219  s->lfoL.offset = s->offset_l;
220  s->lfoR.offset = s->offset_r;
221  s->lfoL.srate = inlink->sample_rate;
222  s->lfoR.srate = inlink->sample_rate;
223  s->lfoL.amount = s->amount;
224  s->lfoR.amount = s->amount;
225  s->lfoL.pwidth = s->pwidth;
226  s->lfoR.pwidth = s->pwidth;
227 
228  return 0;
229 }
230 
231 static const AVFilterPad inputs[] = {
232  {
233  .name = "default",
234  .type = AVMEDIA_TYPE_AUDIO,
235  .config_props = config_input,
236  .filter_frame = filter_frame,
237  },
238  { NULL }
239 };
240 
241 static const AVFilterPad outputs[] = {
242  {
243  .name = "default",
244  .type = AVMEDIA_TYPE_AUDIO,
245  },
246  { NULL }
247 };
248 
250  .name = "apulsator",
251  .description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
252  .priv_size = sizeof(AudioPulsatorContext),
253  .priv_class = &apulsator_class,
255  .inputs = inputs,
256  .outputs = outputs,
257 };
formats
formats
Definition: signature.h:48
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:86
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:86
SQUARE
@ SQUARE
Definition: af_apulsator.c:27
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:1094
AudioPulsatorContext
Definition: af_apulsator.c:40
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
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:204
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:324
AVOption
AVOption.
Definition: opt.h:248
AudioPulsatorContext::timing
int timing
Definition: af_apulsator.c:52
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:149
SimpleLFO::phase
double phase
Definition: af_apulsator.c:31
UNIT_BPM
@ UNIT_BPM
Definition: af_apulsator.c:28
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:65
lfo_advance
static void lfo_advance(SimpleLFO *lfo, unsigned count)
Definition: af_apulsator.c:85
apulsator_options
static const AVOption apulsator_options[]
Definition: af_apulsator.c:60
val
static double val(void *priv, double ch)
Definition: aeval.c:76
SimpleLFO::srate
int srate
Definition: af_apulsator.c:37
AudioPulsatorContext::bpm
double bpm
Definition: af_apulsator.c:49
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:91
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
avassert.h
UNIT_MS
@ UNIT_MS
Definition: af_apulsator.c:28
ff_set_common_formats
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:587
SimpleLFO::pwidth
double pwidth
Definition: af_apulsator.c:35
ff_add_channel_layout
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:338
s
#define s(width, name)
Definition: cbs_vp9.c:257
NB_TIMINGS
@ NB_TIMINGS
Definition: af_apulsator.c:28
AudioPulsatorContext::level_out
double level_out
Definition: af_apulsator.c:44
PulsatorTimings
PulsatorTimings
Definition: af_apulsator.c:28
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_apulsator.c:127
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_apulsator.c:186
ctx
AVFormatContext * ctx
Definition: movenc.c:48
SimpleLFO
Definition: af_apulsator.c:30
AudioPulsatorContext::lfoL
SimpleLFO lfoL
Definition: af_apulsator.c:54
if
if(ret)
Definition: filter_design.txt:179
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
fabs
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
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:659
AudioPulsatorContext::amount
double amount
Definition: af_apulsator.c:45
src
#define src
Definition: vp8dsp.c:255
ff_add_format
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:332
TRIANGLE
@ TRIANGLE
Definition: af_apulsator.c:27
PulsatorModes
PulsatorModes
Definition: af_apulsator.c:27
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(apulsator)
AudioPulsatorContext::hertz
double hertz
Definition: af_apulsator.c:50
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
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
FFMAX
#define FFMAX(a, b)
Definition: common.h:103
ff_af_apulsator
AVFilter ff_af_apulsator
Definition: af_apulsator.c:249
SAWUP
@ SAWUP
Definition: af_apulsator.c:27
AudioPulsatorContext::lfoR
SimpleLFO lfoR
Definition: af_apulsator.c:54
UNIT_HZ
@ UNIT_HZ
Definition: af_apulsator.c:28
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:595
FLAGS
#define FLAGS
Definition: af_apulsator.c:58
FFMIN
#define FFMIN(a, b)
Definition: common.h:105
M_PI
#define M_PI
Definition: mathematics.h:52
internal.h
AudioPulsatorContext::level_in
double level_in
Definition: af_apulsator.c:43
SimpleLFO::freq
double freq
Definition: af_apulsator.c:32
layout
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 layout
Definition: filter_design.txt:18
in
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
Definition: audio_convert.c:326
lfo_get_value
static double lfo_get_value(SimpleLFO *lfo)
Definition: af_apulsator.c:92
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
SimpleLFO::mode
int mode
Definition: af_apulsator.c:36
AVFilter
Filter definition.
Definition: avfilter.h:145
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_apulsator.c:202
AudioPulsatorContext::pwidth
double pwidth
Definition: af_apulsator.c:48
ret
ret
Definition: filter_design.txt:187
NB_MODES
@ NB_MODES
Definition: af_apulsator.c:27
SINE
@ SINE
Definition: af_apulsator.c:27
AudioPulsatorContext::ms
int ms
Definition: af_apulsator.c:51
SimpleLFO::offset
double offset
Definition: af_apulsator.c:33
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
SAWDOWN
@ SAWDOWN
Definition: af_apulsator.c:27
mode
mode
Definition: ebur128.h:83
OFFSET
#define OFFSET(x)
Definition: af_apulsator.c:57
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
avfilter.h
outputs
static const AVFilterPad outputs[]
Definition: af_apulsator.c:241
AudioPulsatorContext::mode
int mode
Definition: af_apulsator.c:42
AVFilterContext
An instance of a filter.
Definition: avfilter.h:341
audio.h
AudioPulsatorContext::offset_r
double offset_r
Definition: af_apulsator.c:47
inputs
static const AVFilterPad inputs[]
Definition: af_apulsator.c:231
SimpleLFO::amount
double amount
Definition: af_apulsator.c:34
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
AudioPulsatorContext::offset_l
double offset_l
Definition: af_apulsator.c:46
ff_set_common_channel_layouts
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:568