FFmpeg  4.4.4
af_asubboost.c
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Lesser General Public
6  * License as published by the Free Software Foundation; either
7  * version 2.1 of the License, or (at your option) any later version.
8  *
9  * FFmpeg is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12  * Lesser General Public License for more details.
13  *
14  * You should have received a copy of the GNU Lesser General Public
15  * License along with FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
20 #include "libavutil/ffmath.h"
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "audio.h"
24 #include "formats.h"
25 
26 typedef struct ASubBoostContext {
27  const AVClass *class;
28 
29  double dry_gain;
30  double wet_gain;
31  double feedback;
32  double decay;
33  double delay;
34  double cutoff;
35  double slope;
36 
37  double a0, a1, a2;
38  double b0, b1, b2;
39 
40  int *write_pos;
42 
46 
48 {
51  static const enum AVSampleFormat sample_fmts[] = {
54  };
55  int ret;
56 
58  if (!formats)
59  return AVERROR(ENOMEM);
61  if (ret < 0)
62  return ret;
63 
65  if (!layouts)
66  return AVERROR(ENOMEM);
67 
69  if (ret < 0)
70  return ret;
71 
74 }
75 
77 {
78  ASubBoostContext *s = ctx->priv;
79  AVFilterLink *inlink = ctx->inputs[0];
80  double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
81  double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
82 
83  s->a0 = 1 + alpha;
84  s->a1 = -2 * cos(w0);
85  s->a2 = 1 - alpha;
86  s->b0 = (1 - cos(w0)) / 2;
87  s->b1 = 1 - cos(w0);
88  s->b2 = (1 - cos(w0)) / 2;
89 
90  s->a1 /= s->a0;
91  s->a2 /= s->a0;
92  s->b0 /= s->a0;
93  s->b1 /= s->a0;
94  s->b2 /= s->a0;
95 
96  s->buffer_samples = inlink->sample_rate * s->delay / 1000;
97 
98  return 0;
99 }
100 
101 static int config_input(AVFilterLink *inlink)
102 {
103  AVFilterContext *ctx = inlink->dst;
104  ASubBoostContext *s = ctx->priv;
105 
106  s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
107  s->w = ff_get_audio_buffer(inlink, 2);
108  s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
109  if (!s->buffer || !s->w || !s->write_pos)
110  return AVERROR(ENOMEM);
111 
112  return get_coeffs(ctx);
113 }
114 
115 typedef struct ThreadData {
116  AVFrame *in, *out;
117 } ThreadData;
118 
119 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
120 {
121  ASubBoostContext *s = ctx->priv;
122  ThreadData *td = arg;
123  AVFrame *out = td->out;
124  AVFrame *in = td->in;
125  const double mix = ctx->is_disabled ? 0. : 1.;
126  const double wet = ctx->is_disabled ? 1. : s->wet_gain;
127  const double dry = ctx->is_disabled ? 1. : s->dry_gain;
128  const double feedback = s->feedback, decay = s->decay;
129  const double b0 = s->b0;
130  const double b1 = s->b1;
131  const double b2 = s->b2;
132  const double a1 = -s->a1;
133  const double a2 = -s->a2;
134  const int start = (in->channels * jobnr) / nb_jobs;
135  const int end = (in->channels * (jobnr+1)) / nb_jobs;
136  const int buffer_samples = s->buffer_samples;
137 
138  for (int ch = start; ch < end; ch++) {
139  const double *src = (const double *)in->extended_data[ch];
140  double *dst = (double *)out->extended_data[ch];
141  double *buffer = (double *)s->buffer->extended_data[ch];
142  double *w = (double *)s->w->extended_data[ch];
143  int write_pos = s->write_pos[ch];
144 
145  for (int n = 0; n < in->nb_samples; n++) {
146  double out_sample;
147 
148  out_sample = src[n] * b0 + w[0];
149  w[0] = b1 * src[n] + w[1] + a1 * out_sample;
150  w[1] = b2 * src[n] + a2 * out_sample;
151 
152  buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
153  dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;
154 
155  if (++write_pos >= buffer_samples)
156  write_pos = 0;
157  }
158 
159  s->write_pos[ch] = write_pos;
160  }
161 
162  return 0;
163 }
164 
165 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
166 {
167  AVFilterContext *ctx = inlink->dst;
168  AVFilterLink *outlink = ctx->outputs[0];
169  ThreadData td;
170  AVFrame *out;
171 
172  if (av_frame_is_writable(in)) {
173  out = in;
174  } else {
175  out = ff_get_audio_buffer(outlink, in->nb_samples);
176  if (!out) {
177  av_frame_free(&in);
178  return AVERROR(ENOMEM);
179  }
181  }
182 
183  td.in = in; td.out = out;
184  ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
186 
187  if (out != in)
188  av_frame_free(&in);
189  return ff_filter_frame(outlink, out);
190 }
191 
193 {
194  ASubBoostContext *s = ctx->priv;
195 
196  av_frame_free(&s->buffer);
197  av_frame_free(&s->w);
198  av_freep(&s->write_pos);
199 }
200 
201 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
202  char *res, int res_len, int flags)
203 {
204  int ret;
205 
206  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
207  if (ret < 0)
208  return ret;
209 
210  return get_coeffs(ctx);
211 }
212 
213 #define OFFSET(x) offsetof(ASubBoostContext, x)
214 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
215 
216 static const AVOption asubboost_options[] = {
217  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
218  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
219  { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
220  { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9}, 0, 1, FLAGS },
221  { "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
222  { "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
223  { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
224  { NULL }
225 };
226 
228 
229 static const AVFilterPad inputs[] = {
230  {
231  .name = "default",
232  .type = AVMEDIA_TYPE_AUDIO,
233  .filter_frame = filter_frame,
234  .config_props = config_input,
235  },
236  { NULL }
237 };
238 
239 static const AVFilterPad outputs[] = {
240  {
241  .name = "default",
242  .type = AVMEDIA_TYPE_AUDIO,
243  },
244  { NULL }
245 };
246 
248  .name = "asubboost",
249  .description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
250  .query_formats = query_formats,
251  .priv_size = sizeof(ASubBoostContext),
252  .priv_class = &asubboost_class,
253  .uninit = uninit,
254  .inputs = inputs,
255  .outputs = outputs,
259 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
AVFILTER_DEFINE_CLASS(asubboost)
static int get_coeffs(AVFilterContext *ctx)
Definition: af_asubboost.c:76
static int query_formats(AVFilterContext *ctx)
Definition: af_asubboost.c:47
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_asubboost.c:119
static int config_input(AVFilterLink *inlink)
Definition: af_asubboost.c:101
#define FLAGS
Definition: af_asubboost.c:214
static const AVFilterPad inputs[]
Definition: af_asubboost.c:229
static const AVOption asubboost_options[]
Definition: af_asubboost.c:216
static const AVFilterPad outputs[]
Definition: af_asubboost.c:239
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_asubboost.c:165
AVFilter ff_af_asubboost
Definition: af_asubboost.c:247
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_asubboost.c:201
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_asubboost.c:192
#define OFFSET(x)
Definition: af_asubboost.c:213
#define av_cold
Definition: attributes.h:88
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
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1096
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:882
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:802
Main libavfilter public API header.
#define flags(name, subs,...)
Definition: cbs_av1.c:561
#define s(width, name)
Definition: cbs_vp9.c:257
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:105
#define NULL
Definition: coverity.c:32
internal math functions header
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:436
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
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
Definition: avfilter.h:134
#define AVERROR(e)
Definition: error.h:43
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:658
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:245
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
static const int16_t alpha[]
Definition: ilbcdata.h:55
static int mix(int c0, int c1)
Definition: 4xm.c:715
const char * arg
Definition: jacosubdec.c:66
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
uint8_t w
Definition: llviddspenc.c:39
#define M_PI
Definition: mathematics.h:52
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVOptions.
#define a2
Definition: regdef.h:48
#define td
Definition: regdef.h:70
#define a1
Definition: regdef.h:47
static char buffer[20]
Definition: seek.c:32
formats
Definition: signature.h:48
AVFrame * buffer
Definition: af_asubboost.c:44
Describe the class of an AVClass context structure.
Definition: log.h:67
A list of supported channel layouts.
Definition: formats.h:86
An instance of a filter.
Definition: avfilter.h:341
A list of supported formats for one end of a filter link.
Definition: formats.h:65
A filter pad used for either input or output.
Definition: internal.h:54
const char * name
Pad name.
Definition: internal.h:60
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
AVFormatInternal * internal
An opaque field for libavformat internal usage.
Definition: avformat.h:1699
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVOption.
Definition: opt.h:248
Used for passing data between threads.
Definition: dsddec.c:67
AVFrame * out
Definition: af_adeclick.c:502
AVFrame * in
Definition: af_adenorm.c:223
#define av_freep(p)
#define src
Definition: vp8dsp.c:255
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
static double b1(void *priv, double x, double y)
Definition: vf_xfade.c:1665
static double b2(void *priv, double x, double y)
Definition: vf_xfade.c:1666
static double b0(void *priv, double x, double y)
Definition: vf_xfade.c:1664