FFmpeg
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 
43  AVFrame *i, *o;
46 
48 {
51  static const enum AVSampleFormat sample_fmts[] = {
54  };
55  int ret;
56 
57  formats = ff_make_format_list(sample_fmts);
58  if (!formats)
59  return AVERROR(ENOMEM);
60  ret = ff_set_common_formats(ctx, formats);
61  if (ret < 0)
62  return ret;
63 
64  layouts = ff_all_channel_counts();
65  if (!layouts)
66  return AVERROR(ENOMEM);
67 
68  ret = ff_set_common_channel_layouts(ctx, layouts);
69  if (ret < 0)
70  return ret;
71 
72  formats = ff_all_samplerates();
73  return ff_set_common_samplerates(ctx, formats);
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 
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->i = ff_get_audio_buffer(inlink, 2);
108  s->o = ff_get_audio_buffer(inlink, 2);
109  if (!s->buffer || !s->i || !s->o)
110  return AVERROR(ENOMEM);
111 
112  return get_coeffs(ctx);
113 }
114 
116 {
117  AVFilterContext *ctx = inlink->dst;
118  AVFilterLink *outlink = ctx->outputs[0];
119  ASubBoostContext *s = ctx->priv;
120  const float wet = s->wet_gain, dry = s->dry_gain, feedback = s->feedback, decay = s->decay;
121  int write_pos;
122  AVFrame *out;
123 
124  if (av_frame_is_writable(in)) {
125  out = in;
126  } else {
127  out = ff_get_audio_buffer(outlink, in->nb_samples);
128  if (!out) {
129  av_frame_free(&in);
130  return AVERROR(ENOMEM);
131  }
132  av_frame_copy_props(out, in);
133  }
134 
135  for (int ch = 0; ch < in->channels; ch++) {
136  const double *src = (const double *)in->extended_data[ch];
137  double *dst = (double *)out->extended_data[ch];
138  double *buffer = (double *)s->buffer->extended_data[ch];
139  double *ix = (double *)s->i->extended_data[ch];
140  double *ox = (double *)s->o->extended_data[ch];
141 
142  write_pos = s->write_pos;
143  for (int n = 0; n < in->nb_samples; n++) {
144  double out_sample;
145 
146  out_sample = src[n] * s->b0 + ix[0] * s->b1 + ix[1] * s->b2 - ox[0] * s->a1 - ox[1] * s->a2;
147  ix[1] = ix[0];
148  ix[0] = src[n];
149  ox[1] = ox[0];
150  ox[0] = out_sample;
151 
152  buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
153  dst[n] = src[n] * dry + buffer[write_pos] * wet;
154 
155  if (++write_pos >= s->buffer_samples)
156  write_pos = 0;
157  }
158  }
159 
160  s->write_pos = write_pos;
161 
162  if (out != in)
163  av_frame_free(&in);
164  return ff_filter_frame(outlink, out);
165 }
166 
168 {
169  ASubBoostContext *s = ctx->priv;
170 
171  av_frame_free(&s->buffer);
172  av_frame_free(&s->i);
173  av_frame_free(&s->o);
174 }
175 
176 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
177  char *res, int res_len, int flags)
178 {
179  int ret;
180 
181  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
182  if (ret < 0)
183  return ret;
184 
185  return get_coeffs(ctx);
186 }
187 
188 #define OFFSET(x) offsetof(ASubBoostContext, x)
189 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
190 
191 static const AVOption asubboost_options[] = {
192  { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
193  { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.8}, 0, 1, FLAGS },
194  { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
195  { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
196  { "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
197  { "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
198  { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
199  { NULL }
200 };
201 
202 AVFILTER_DEFINE_CLASS(asubboost);
203 
204 static const AVFilterPad inputs[] = {
205  {
206  .name = "default",
207  .type = AVMEDIA_TYPE_AUDIO,
208  .filter_frame = filter_frame,
209  .config_props = config_input,
210  },
211  { NULL }
212 };
213 
214 static const AVFilterPad outputs[] = {
215  {
216  .name = "default",
217  .type = AVMEDIA_TYPE_AUDIO,
218  },
219  { NULL }
220 };
221 
223  .name = "asubboost",
224  .description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
225  .query_formats = query_formats,
226  .priv_size = sizeof(ASubBoostContext),
227  .priv_class = &asubboost_class,
228  .uninit = uninit,
229  .inputs = inputs,
230  .outputs = outputs,
232 };
#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:581
This structure describes decoded (raw) audio or video data.
Definition: frame.h:308
AVOption.
Definition: opt.h:248
Main libavfilter public API header.
double, planar
Definition: samplefmt.h:70
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
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1091
#define av_cold
Definition: attributes.h:88
AVOptions.
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_asubboost.c:176
static int config_input(AVFilterLink *inlink)
Definition: af_asubboost.c:101
static int query_formats(AVFilterContext *ctx)
Definition: af_asubboost.c:47
A filter pad used for either input or output.
Definition: internal.h:54
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_asubboost.c:115
#define src
Definition: vp8dsp.c:254
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:600
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:203
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
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:885
void * priv
private data for use by the filter
Definition: avfilter.h:353
static const AVFilterPad outputs[]
Definition: af_asubboost.c:214
AVFILTER_DEFINE_CLASS(asubboost)
int channels
number of audio channels, only used for audio.
Definition: frame.h:614
audio channel layout utility functions
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
A list of supported channel layouts.
Definition: formats.h:85
static const AVOption asubboost_options[]
Definition: af_asubboost.c:191
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:595
static const int16_t alpha[]
Definition: ilbcdata.h:55
static int get_coeffs(AVFilterContext *ctx)
Definition: af_asubboost.c:76
#define FLAGS
Definition: af_asubboost.c:189
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
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
AVFilter ff_af_asubboost
Definition: af_asubboost.c:222
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
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){int planes=out->planar?out->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:425
#define flags(name, subs,...)
Definition: cbs_av1.c:560
AVFrame * buffer
Definition: af_asubboost.c:44
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
internal math functions header
#define OFFSET(x)
Definition: af_asubboost.c:188
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_asubboost.c:167
static const AVFilterPad inputs[]
Definition: af_asubboost.c:204
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:723
FILE * out
Definition: movenc.c:54
#define M_PI
Definition: mathematics.h:52
formats
Definition: signature.h:48
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:440
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:355
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:588
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659
GLuint buffer
Definition: opengl_enc.c:101