FFmpeg
af_aphaser.c
Go to the documentation of this file.
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 /**
22  * @file
23  * phaser audio filter
24  */
25 
26 #include "libavutil/avassert.h"
27 #include "libavutil/opt.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "internal.h"
31 #include "generate_wave_table.h"
32 
33 typedef struct AudioPhaserContext {
34  const AVClass *class;
35  double in_gain, out_gain;
36  double delay;
37  double decay;
38  double speed;
39 
40  int type;
41 
43  double *delay_buffer;
44 
47 
49 
51  uint8_t * const *src, uint8_t **dst,
52  int nb_samples, int channels);
54 
55 #define OFFSET(x) offsetof(AudioPhaserContext, x)
56 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
57 
58 static const AVOption aphaser_options[] = {
59  { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
60  { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
61  { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
62  { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
63  { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
64  { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
65  { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
66  { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
67  { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
68  { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
69  { NULL }
70 };
71 
72 AVFILTER_DEFINE_CLASS(aphaser);
73 
75 {
76  AudioPhaserContext *s = ctx->priv;
77 
78  if (s->in_gain > (1 - s->decay * s->decay))
79  av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
80  if (s->in_gain / (1 - s->decay) > 1 / s->out_gain)
81  av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
82 
83  return 0;
84 }
85 
87 {
90  static const enum AVSampleFormat sample_fmts[] = {
96  };
97  int ret;
98 
99  layouts = ff_all_channel_counts();
100  if (!layouts)
101  return AVERROR(ENOMEM);
102  ret = ff_set_common_channel_layouts(ctx, layouts);
103  if (ret < 0)
104  return ret;
105 
106  formats = ff_make_format_list(sample_fmts);
107  if (!formats)
108  return AVERROR(ENOMEM);
109  ret = ff_set_common_formats(ctx, formats);
110  if (ret < 0)
111  return ret;
112 
113  formats = ff_all_samplerates();
114  if (!formats)
115  return AVERROR(ENOMEM);
116  return ff_set_common_samplerates(ctx, formats);
117 }
118 
119 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
120 
121 #define PHASER_PLANAR(name, type) \
122 static void phaser_## name ##p(AudioPhaserContext *s, \
123  uint8_t * const *ssrc, uint8_t **ddst, \
124  int nb_samples, int channels) \
125 { \
126  int i, c, delay_pos, modulation_pos; \
127  \
128  av_assert0(channels > 0); \
129  for (c = 0; c < channels; c++) { \
130  type *src = (type *)ssrc[c]; \
131  type *dst = (type *)ddst[c]; \
132  double *buffer = s->delay_buffer + \
133  c * s->delay_buffer_length; \
134  \
135  delay_pos = s->delay_pos; \
136  modulation_pos = s->modulation_pos; \
137  \
138  for (i = 0; i < nb_samples; i++, src++, dst++) { \
139  double v = *src * s->in_gain + buffer[ \
140  MOD(delay_pos + s->modulation_buffer[ \
141  modulation_pos], \
142  s->delay_buffer_length)] * s->decay; \
143  \
144  modulation_pos = MOD(modulation_pos + 1, \
145  s->modulation_buffer_length); \
146  delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
147  buffer[delay_pos] = v; \
148  \
149  *dst = v * s->out_gain; \
150  } \
151  } \
152  \
153  s->delay_pos = delay_pos; \
154  s->modulation_pos = modulation_pos; \
155 }
156 
157 #define PHASER(name, type) \
158 static void phaser_## name (AudioPhaserContext *s, \
159  uint8_t * const *ssrc, uint8_t **ddst, \
160  int nb_samples, int channels) \
161 { \
162  int i, c, delay_pos, modulation_pos; \
163  type *src = (type *)ssrc[0]; \
164  type *dst = (type *)ddst[0]; \
165  double *buffer = s->delay_buffer; \
166  \
167  delay_pos = s->delay_pos; \
168  modulation_pos = s->modulation_pos; \
169  \
170  for (i = 0; i < nb_samples; i++) { \
171  int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
172  s->delay_buffer_length) * channels; \
173  int npos; \
174  \
175  delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
176  npos = delay_pos * channels; \
177  for (c = 0; c < channels; c++, src++, dst++) { \
178  double v = *src * s->in_gain + buffer[pos + c] * s->decay; \
179  \
180  buffer[npos + c] = v; \
181  \
182  *dst = v * s->out_gain; \
183  } \
184  \
185  modulation_pos = MOD(modulation_pos + 1, \
186  s->modulation_buffer_length); \
187  } \
188  \
189  s->delay_pos = delay_pos; \
190  s->modulation_pos = modulation_pos; \
191 }
192 
193 PHASER_PLANAR(dbl, double)
194 PHASER_PLANAR(flt, float)
195 PHASER_PLANAR(s16, int16_t)
197 
198 PHASER(dbl, double)
199 PHASER(flt, float)
200 PHASER(s16, int16_t)
201 PHASER(s32, int32_t)
202 
203 static int config_output(AVFilterLink *outlink)
204 {
205  AudioPhaserContext *s = outlink->src->priv;
206  AVFilterLink *inlink = outlink->src->inputs[0];
207 
208  s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
209  if (s->delay_buffer_length <= 0) {
210  av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
211  return AVERROR(EINVAL);
212  }
213  s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->channels);
214  s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
216 
217  if (!s->modulation_buffer || !s->delay_buffer)
218  return AVERROR(ENOMEM);
219 
222  1., s->delay_buffer_length, M_PI / 2.0);
223 
224  s->delay_pos = s->modulation_pos = 0;
225 
226  switch (inlink->format) {
227  case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break;
228  case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
229  case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break;
230  case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
231  case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break;
232  case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
233  case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break;
234  case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
235  default: av_assert0(0);
236  }
237 
238  return 0;
239 }
240 
242 {
243  AudioPhaserContext *s = inlink->dst->priv;
244  AVFilterLink *outlink = inlink->dst->outputs[0];
245  AVFrame *outbuf;
246 
247  if (av_frame_is_writable(inbuf)) {
248  outbuf = inbuf;
249  } else {
250  outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples);
251  if (!outbuf) {
252  av_frame_free(&inbuf);
253  return AVERROR(ENOMEM);
254  }
255  av_frame_copy_props(outbuf, inbuf);
256  }
257 
258  s->phaser(s, inbuf->extended_data, outbuf->extended_data,
259  outbuf->nb_samples, outbuf->channels);
260 
261  if (inbuf != outbuf)
262  av_frame_free(&inbuf);
263 
264  return ff_filter_frame(outlink, outbuf);
265 }
266 
268 {
269  AudioPhaserContext *s = ctx->priv;
270 
271  av_freep(&s->delay_buffer);
273 }
274 
275 static const AVFilterPad aphaser_inputs[] = {
276  {
277  .name = "default",
278  .type = AVMEDIA_TYPE_AUDIO,
279  .filter_frame = filter_frame,
280  },
281  { NULL }
282 };
283 
284 static const AVFilterPad aphaser_outputs[] = {
285  {
286  .name = "default",
287  .type = AVMEDIA_TYPE_AUDIO,
288  .config_props = config_output,
289  },
290  { NULL }
291 };
292 
294  .name = "aphaser",
295  .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
296  .query_formats = query_formats,
297  .priv_size = sizeof(AudioPhaserContext),
298  .init = init,
299  .uninit = uninit,
300  .inputs = aphaser_inputs,
301  .outputs = aphaser_outputs,
302  .priv_class = &aphaser_class,
303 };
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:550
void(* phaser)(struct AudioPhaserContext *s, uint8_t *const *src, uint8_t **dst, int nb_samples, int channels)
Definition: af_aphaser.c:50
static const AVOption aphaser_options[]
Definition: af_aphaser.c:58
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
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
double, planar
Definition: samplefmt.h:70
GLint GLenum type
Definition: opengl_enc.c:104
#define src
Definition: vp8dsp.c:254
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
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
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
uint8_t
#define av_cold
Definition: attributes.h:82
AVOptions.
static const AVFilterPad aphaser_outputs[]
Definition: af_aphaser.c:284
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aphaser.c:267
signed 32 bits
Definition: samplefmt.h:62
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
#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:569
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 PHASER(name, type)
Definition: af_aphaser.c:157
#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 FLAGS
Definition: af_aphaser.c:56
simple assert() macros that are a bit more flexible than ISO C assert().
static int config_output(AVFilterLink *outlink)
Definition: af_aphaser.c:203
static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
Definition: af_aphaser.c:241
int channels
number of audio channels, only used for audio.
Definition: frame.h:601
signed 32 bits, planar
Definition: samplefmt.h:68
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
static av_cold int init(AVFilterContext *ctx)
Definition: af_aphaser.c:74
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
void ff_generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt, void *table, int table_size, double min, double max, double phase)
A list of supported channel layouts.
Definition: formats.h:85
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
AVFILTER_DEFINE_CLASS(aphaser)
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
double * delay_buffer
Definition: af_aphaser.c:43
#define OFFSET(x)
Definition: af_aphaser.c:55
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
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
AVFilter ff_af_aphaser
Definition: af_aphaser.c:293
int modulation_buffer_length
Definition: af_aphaser.c:45
signed 16 bits
Definition: samplefmt.h:61
static int query_formats(AVFilterContext *ctx)
Definition: af_aphaser.c:86
A list of supported formats for one end of a filter link.
Definition: formats.h:64
int32_t * modulation_buffer
Definition: af_aphaser.c:46
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
static const AVFilterPad aphaser_inputs[]
Definition: af_aphaser.c:275
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
#define M_PI
Definition: mathematics.h:52
#define av_malloc_array(a, b)
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:409
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
#define PHASER_PLANAR(name, type)
Definition: af_aphaser.c:121
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654