FFmpeg
af_aphaser.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 /**
22  * @file
23  * phaser audio filter
24  */
25 
26 #include "libavutil/avassert.h"
27 #include "libavutil/mem.h"
28 #include "libavutil/opt.h"
29 #include "audio.h"
30 #include "avfilter.h"
31 #include "filters.h"
32 #include "generate_wave_table.h"
33 
34 typedef struct AudioPhaserContext {
35  const AVClass *class;
36  double in_gain, out_gain;
37  double delay;
38  double decay;
39  double speed;
40 
41  int type;
42 
43  int delay_buffer_length;
44  double *delay_buffer;
45 
46  int modulation_buffer_length;
47  int32_t *modulation_buffer;
48 
49  int delay_pos, modulation_pos;
50 
51  void (*phaser)(struct AudioPhaserContext *s,
52  uint8_t * const *src, uint8_t **dst,
53  int nb_samples, int channels);
54 } AudioPhaserContext;
55 
56 #define OFFSET(x) offsetof(AudioPhaserContext, x)
57 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58 
59 static const AVOption aphaser_options[] = {
60  { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
61  { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
62  { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
63  { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
64  { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
65  { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, .unit = "type" },
66  { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, .unit = "type" },
67  { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, .unit = "type" },
68  { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, .unit = "type" },
69  { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, .unit = "type" },
70  { NULL }
71 };
72 
73 AVFILTER_DEFINE_CLASS(aphaser);
74 
75 static av_cold int init(AVFilterContext *ctx)
76 {
77  AudioPhaserContext *s = ctx->priv;
78 
79  if (s->in_gain > (1 - s->decay * s->decay))
80  av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
81  if (s->in_gain / (1 - s->decay) > 1 / s->out_gain)
82  av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
83 
84  return 0;
85 }
86 
87 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
88 
89 #define PHASER_PLANAR(name, type) \
90 static void phaser_## name ##p(AudioPhaserContext *s, \
91  uint8_t * const *ssrc, uint8_t **ddst, \
92  int nb_samples, int channels) \
93 { \
94  int i, c, delay_pos, modulation_pos; \
95  \
96  av_assert0(channels > 0); \
97  for (c = 0; c < channels; c++) { \
98  type *src = (type *)ssrc[c]; \
99  type *dst = (type *)ddst[c]; \
100  double *buffer = s->delay_buffer + \
101  c * s->delay_buffer_length; \
102  \
103  delay_pos = s->delay_pos; \
104  modulation_pos = s->modulation_pos; \
105  \
106  for (i = 0; i < nb_samples; i++, src++, dst++) { \
107  double v = *src * s->in_gain + buffer[ \
108  MOD(delay_pos + s->modulation_buffer[ \
109  modulation_pos], \
110  s->delay_buffer_length)] * s->decay; \
111  \
112  modulation_pos = MOD(modulation_pos + 1, \
113  s->modulation_buffer_length); \
114  delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
115  buffer[delay_pos] = v; \
116  \
117  *dst = v * s->out_gain; \
118  } \
119  } \
120  \
121  s->delay_pos = delay_pos; \
122  s->modulation_pos = modulation_pos; \
123 }
124 
125 #define PHASER(name, type) \
126 static void phaser_## name (AudioPhaserContext *s, \
127  uint8_t * const *ssrc, uint8_t **ddst, \
128  int nb_samples, int channels) \
129 { \
130  int i, c, delay_pos, modulation_pos; \
131  type *src = (type *)ssrc[0]; \
132  type *dst = (type *)ddst[0]; \
133  double *buffer = s->delay_buffer; \
134  \
135  delay_pos = s->delay_pos; \
136  modulation_pos = s->modulation_pos; \
137  \
138  for (i = 0; i < nb_samples; i++) { \
139  int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
140  s->delay_buffer_length) * channels; \
141  int npos; \
142  \
143  delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
144  npos = delay_pos * channels; \
145  for (c = 0; c < channels; c++, src++, dst++) { \
146  double v = *src * s->in_gain + buffer[pos + c] * s->decay; \
147  \
148  buffer[npos + c] = v; \
149  \
150  *dst = v * s->out_gain; \
151  } \
152  \
153  modulation_pos = MOD(modulation_pos + 1, \
154  s->modulation_buffer_length); \
155  } \
156  \
157  s->delay_pos = delay_pos; \
158  s->modulation_pos = modulation_pos; \
159 }
160 
161 PHASER_PLANAR(dbl, double)
162 PHASER_PLANAR(flt, float)
163 PHASER_PLANAR(s16, int16_t)
164 PHASER_PLANAR(s32, int32_t)
165 
166 PHASER(dbl, double)
167 PHASER(flt, float)
168 PHASER(s16, int16_t)
169 PHASER(s32, int32_t)
170 
171 static int config_output(AVFilterLink *outlink)
172 {
173  AudioPhaserContext *s = outlink->src->priv;
174  AVFilterLink *inlink = outlink->src->inputs[0];
175 
176  s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
177  if (s->delay_buffer_length <= 0) {
178  av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
179  return AVERROR(EINVAL);
180  }
181  s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->ch_layout.nb_channels);
182  s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
183  s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
184 
185  if (!s->modulation_buffer || !s->delay_buffer)
186  return AVERROR(ENOMEM);
187 
188  ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
189  s->modulation_buffer, s->modulation_buffer_length,
190  1., s->delay_buffer_length, M_PI / 2.0);
191 
192  s->delay_pos = s->modulation_pos = 0;
193 
194  switch (inlink->format) {
195  case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break;
196  case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
197  case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break;
198  case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
199  case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break;
200  case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
201  case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break;
202  case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
203  default: av_assert0(0);
204  }
205 
206  return 0;
207 }
208 
209 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
210 {
211  AudioPhaserContext *s = inlink->dst->priv;
212  AVFilterLink *outlink = inlink->dst->outputs[0];
213  AVFrame *outbuf;
214 
215  if (av_frame_is_writable(inbuf)) {
216  outbuf = inbuf;
217  } else {
218  outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples);
219  if (!outbuf) {
220  av_frame_free(&inbuf);
221  return AVERROR(ENOMEM);
222  }
223  av_frame_copy_props(outbuf, inbuf);
224  }
225 
226  s->phaser(s, inbuf->extended_data, outbuf->extended_data,
227  outbuf->nb_samples, outbuf->ch_layout.nb_channels);
228 
229  if (inbuf != outbuf)
230  av_frame_free(&inbuf);
231 
232  return ff_filter_frame(outlink, outbuf);
233 }
234 
235 static av_cold void uninit(AVFilterContext *ctx)
236 {
237  AudioPhaserContext *s = ctx->priv;
238 
239  av_freep(&s->delay_buffer);
240  av_freep(&s->modulation_buffer);
241 }
242 
243 static const AVFilterPad aphaser_inputs[] = {
244  {
245  .name = "default",
246  .type = AVMEDIA_TYPE_AUDIO,
247  .filter_frame = filter_frame,
248  },
249 };
250 
251 static const AVFilterPad aphaser_outputs[] = {
252  {
253  .name = "default",
254  .type = AVMEDIA_TYPE_AUDIO,
255  .config_props = config_output,
256  },
257 };
258 
259 const AVFilter ff_af_aphaser = {
260  .name = "aphaser",
261  .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
262  .priv_size = sizeof(AudioPhaserContext),
263  .init = init,
264  .uninit = uninit,
265  FILTER_INPUTS(aphaser_inputs),
266  FILTER_OUTPUTS(aphaser_outputs),
267  FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
268  AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
269  AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
270  AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P),
271  .priv_class = &aphaser_class,
272 };
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
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
AudioPhaserContext::phaser
void(* phaser)(struct AudioPhaserContext *s, uint8_t *const *src, uint8_t **dst, int nb_samples, int channels)
Definition: af_aphaser.c:51
AudioPhaserContext::decay
double decay
Definition: af_aphaser.c:38
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1061
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: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
AVOption
AVOption.
Definition: opt.h:429
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:65
aphaser_options
static const AVOption aphaser_options[]
Definition: af_aphaser.c:59
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:328
AudioPhaserContext
Definition: af_aphaser.c:34
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
Definition: af_aphaser.c:209
WAVE_TRI
@ WAVE_TRI
Definition: generate_wave_table.h:26
AudioPhaserContext::delay_buffer_length
int delay_buffer_length
Definition: af_aphaser.c:43
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:790
type
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
Definition: writing_filters.txt:86
AVFilterPad
A filter pad used for either input or output.
Definition: filters.h:38
avassert.h
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:209
av_cold
#define av_cold
Definition: attributes.h:90
FILTER_SAMPLEFMTS
#define FILTER_SAMPLEFMTS(...)
Definition: filters.h:250
s
#define s(width, name)
Definition: cbs_vp9.c:198
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Underlying C type is double.
Definition: opt.h:267
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:49
channels
channels
Definition: aptx.h:31
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: filters.h:263
WAVE_SIN
@ WAVE_SIN
Definition: generate_wave_table.h:25
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
WAVE_NB
@ WAVE_NB
Definition: generate_wave_table.h:27
PHASER_PLANAR
#define PHASER_PLANAR(name, type)
Definition: af_aphaser.c:89
AudioPhaserContext::delay
double delay
Definition: af_aphaser.c:37
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
AudioPhaserContext::delay_pos
int delay_pos
Definition: af_aphaser.c:49
FLAGS
#define FLAGS
Definition: af_aphaser.c:57
AudioPhaserContext::modulation_pos
int modulation_pos
Definition: af_aphaser.c:49
aphaser_outputs
static const AVFilterPad aphaser_outputs[]
Definition: af_aphaser.c:251
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
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aphaser.c:235
ff_af_aphaser
const AVFilter ff_af_aphaser
Definition: af_aphaser.c:259
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:661
AudioPhaserContext::type
int type
Definition: af_aphaser.c:41
AudioPhaserContext::out_gain
double out_gain
Definition: af_aphaser.c:36
M_PI
#define M_PI
Definition: mathematics.h:67
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:64
PHASER
#define PHASER(name, type)
Definition: af_aphaser.c:125
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:469
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:450
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:31
AudioPhaserContext::delay_buffer
double * delay_buffer
Definition: af_aphaser.c:44
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(aphaser)
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
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
AudioPhaserContext::modulation_buffer
int32_t * modulation_buffer
Definition: af_aphaser.c:47
aphaser_inputs
static const AVFilterPad aphaser_inputs[]
Definition: af_aphaser.c:243
generate_wave_table.h
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Underlying C type is int.
Definition: opt.h:259
avfilter.h
AudioPhaserContext::speed
double speed
Definition: af_aphaser.c:39
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
AudioPhaserContext::modulation_buffer_length
int modulation_buffer_length
Definition: af_aphaser.c:46
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
mem.h
audio.h
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_aphaser.c:75
int32_t
int32_t
Definition: audioconvert.c:56
AudioPhaserContext::in_gain
double in_gain
Definition: af_aphaser.c:36
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:61
OFFSET
#define OFFSET(x)
Definition: af_aphaser.c:56
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:59
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Special option type for declaring named constants.
Definition: opt.h:299
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_aphaser.c:171
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
src
#define src
Definition: vp8dsp.c:248
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