FFmpeg  4.4.5
af_earwax.c
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2011 Mina Nagy Zaki
3  * Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
4  * This source code is freely redistributable and may be used for any purpose.
5  * This copyright notice must be maintained. Edward Beingessner And Sundry
6  * Contributors are not responsible for the consequences of using this
7  * software.
8  *
9  * This file is part of FFmpeg.
10  *
11  * FFmpeg is free software; you can redistribute it and/or
12  * modify it under the terms of the GNU Lesser General Public
13  * License as published by the Free Software Foundation; either
14  * version 2.1 of the License, or (at your option) any later version.
15  *
16  * FFmpeg is distributed in the hope that it will be useful,
17  * but WITHOUT ANY WARRANTY; without even the implied warranty of
18  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19  * Lesser General Public License for more details.
20  *
21  * You should have received a copy of the GNU Lesser General Public
22  * License along with FFmpeg; if not, write to the Free Software
23  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24  */
25 
26 /**
27  * @file
28  * Stereo Widening Effect. Adds audio cues to move stereo image in
29  * front of the listener. Adapted from the libsox earwax effect.
30  */
31 
33 #include "avfilter.h"
34 #include "audio.h"
35 #include "formats.h"
36 
37 #define NUMTAPS 32
38 
39 static const int8_t filt[NUMTAPS * 2] = {
40 /* 30° 330° */
41  4, -6, /* 32 tap stereo FIR filter. */
42  4, -11, /* One side filters as if the */
43  -1, -5, /* signal was from 30 degrees */
44  3, 3, /* from the ear, the other as */
45  -2, 5, /* if 330 degrees. */
46  -5, 0,
47  9, 1,
48  6, 3, /* Input */
49  -4, -1, /* Left Right */
50  -5, -3, /* __________ __________ */
51  -2, -5, /* | | | | */
52  -7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
53  6, -7, /* / |__________| |__________| \ */
54  30, -29, /* / \ / \ */
55  12, -3, /* / X \ */
56  -11, 4, /* / / \ \ */
57  -3, 7, /* ____V_____ __________V V__________ _____V____ */
58  -20, 23, /* | | | | | | | | */
59  2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
60  1, -6, /* |__________| |__________| |__________| |__________| */
61  -14, -5, /* \ ___ / \ ___ / */
62  15, -18, /* \ / \ / _____ \ / \ / */
63  6, 7, /* `->| + |<--' / \ `-->| + |<-' */
64  15, -10, /* \___/ _/ \_ \___/ */
65  -14, 22, /* \ / \ / \ / */
66  -7, -2, /* `--->| | | |<---' */
67  -4, 9, /* \_/ \_/ */
68  6, -12, /* */
69  6, -6, /* Headphones */
70  0, -11,
71  0, -5,
72  4, 0};
73 
74 typedef struct EarwaxContext {
75  int16_t filter[2][NUMTAPS];
76  int16_t taps[4][NUMTAPS * 2];
77 
80 
82 {
83  static const int sample_rates[] = { 44100, -1 };
84  int ret;
85 
88 
89  if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
90  (ret = ff_set_common_formats (ctx , formats )) < 0 ||
92  (ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
94  return ret;
95 
96  return 0;
97 }
98 
99 //FIXME: replace with DSPContext.scalarproduct_int16
100 static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
101  const int16_t *filt, int16_t *out)
102 {
103  int32_t sample;
104  int16_t j;
105 
106  while (in < endin) {
107  sample = 0;
108  for (j = 0; j < NUMTAPS; j++)
109  sample += in[j] * filt[j];
110  *out = av_clip_int16(sample >> 7);
111  out++;
112  in++;
113  }
114 
115  return out;
116 }
117 
118 static int config_input(AVFilterLink *inlink)
119 {
120  EarwaxContext *s = inlink->dst->priv;
121 
122  for (int i = 0; i < NUMTAPS; i++) {
123  s->filter[0][i] = filt[i * 2];
124  s->filter[1][i] = filt[i * 2 + 1];
125  }
126 
127  return 0;
128 }
129 
131  int input_ch, int output_ch,
132  int filter_ch, int tap_ch)
133 {
134  EarwaxContext *s = ctx->priv;
135  int16_t *taps, *endin, *dst, *src;
136  int len;
137 
138  taps = s->taps[tap_ch];
139  dst = (int16_t *)s->frame[input_ch]->data[output_ch];
140  src = (int16_t *)in->data[input_ch];
141 
142  len = FFMIN(NUMTAPS, in->nb_samples);
143  // copy part of new input and process with saved input
144  memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
145  dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
146 
147  // process current input
148  if (in->nb_samples >= NUMTAPS) {
149  endin = src + in->nb_samples - NUMTAPS;
150  scalarproduct(src, endin, s->filter[filter_ch], dst);
151 
152  // save part of input for next round
153  memcpy(taps, endin, NUMTAPS * sizeof(*taps));
154  } else {
155  memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
156  }
157 }
158 
160  int output_ch, int f0, int f1, int i0, int i1)
161 {
162  EarwaxContext *s = ctx->priv;
163  const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
164  const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
165  int16_t *dst = (int16_t *)out->data[output_ch];
166 
167  for (int n = 0; n < out->nb_samples; n++)
168  dst[n] = av_clip_int16(srcl[n] + srcr[n]);
169 }
170 
171 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
172 {
173  AVFilterContext *ctx = inlink->dst;
174  EarwaxContext *s = ctx->priv;
175  AVFilterLink *outlink = ctx->outputs[0];
176  AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
177 
178  for (int ch = 0; ch < 2; ch++) {
179  if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
180  av_frame_free(&s->frame[ch]);
181  s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
182  if (!s->frame[ch]) {
183  av_frame_free(&in);
184  av_frame_free(&out);
185  return AVERROR(ENOMEM);
186  }
187  }
188  }
189 
190  if (!out) {
191  av_frame_free(&in);
192  return AVERROR(ENOMEM);
193  }
195 
196  convolve(ctx, in, 0, 0, 0, 0);
197  convolve(ctx, in, 0, 1, 1, 1);
198  convolve(ctx, in, 1, 0, 0, 2);
199  convolve(ctx, in, 1, 1, 1, 3);
200 
201  mix(ctx, out, 0, 0, 1, 1, 0);
202  mix(ctx, out, 1, 0, 1, 0, 1);
203 
204  av_frame_free(&in);
205  return ff_filter_frame(outlink, out);
206 }
207 
209 {
210  EarwaxContext *s = ctx->priv;
211 
212  av_frame_free(&s->frame[0]);
213  av_frame_free(&s->frame[1]);
214 }
215 
216 static const AVFilterPad earwax_inputs[] = {
217  {
218  .name = "default",
219  .type = AVMEDIA_TYPE_AUDIO,
220  .filter_frame = filter_frame,
221  .config_props = config_input,
222  },
223  { NULL }
224 };
225 
226 static const AVFilterPad earwax_outputs[] = {
227  {
228  .name = "default",
229  .type = AVMEDIA_TYPE_AUDIO,
230  },
231  { NULL }
232 };
233 
235  .name = "earwax",
236  .description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
237  .query_formats = query_formats,
238  .priv_size = sizeof(EarwaxContext),
239  .uninit = uninit,
242 };
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
static const int8_t filt[NUMTAPS *2]
Definition: af_earwax.c:39
AVFilter ff_af_earwax
Definition: af_earwax.c:234
static const AVFilterPad earwax_inputs[]
Definition: af_earwax.c:216
static const AVFilterPad earwax_outputs[]
Definition: af_earwax.c:226
static void convolve(AVFilterContext *ctx, AVFrame *in, int input_ch, int output_ch, int filter_ch, int tap_ch)
Definition: af_earwax.c:130
static int query_formats(AVFilterContext *ctx)
Definition: af_earwax.c:81
static int16_t * scalarproduct(const int16_t *in, const int16_t *endin, const int16_t *filt, int16_t *out)
Definition: af_earwax.c:100
static int config_input(AVFilterLink *inlink)
Definition: af_earwax.c:118
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_earwax.c:171
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_earwax.c:208
#define NUMTAPS
Definition: af_earwax.c:37
static void mix(AVFilterContext *ctx, AVFrame *out, int output_ch, int f0, int f1, int i0, int i1)
Definition: af_earwax.c:159
#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
int32_t
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1096
Main libavfilter public API header.
#define s(width, name)
Definition: cbs_vp9.c:257
uint64_t layout
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:105
#define av_clip_int16
Definition: common.h:137
#define NULL
Definition: coverity.c:32
sample_rates
#define sample
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:338
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
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:332
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
#define AV_CH_LAYOUT_STEREO
#define AVERROR(e)
Definition: error.h:43
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
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
for(j=16;j >0;--j)
int i
Definition: input.c:407
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
formats
Definition: signature.h:48
A list of supported channel layouts.
Definition: formats.h:86
An instance of a filter.
Definition: avfilter.h:341
void * priv
private data for use by the filter
Definition: avfilter.h:356
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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVFrame * frame[2]
Definition: af_earwax.c:78
int16_t filter[2][NUMTAPS]
Definition: af_earwax.c:75
int16_t taps[4][NUMTAPS *2]
Definition: af_earwax.c:76
#define src
Definition: vp8dsp.c:255
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
int len