FFmpeg
af_acontrast.c
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1 /*
2  * Copyright (c) 2008 Rob Sykes
3  * Copyright (c) 2017 Paul B Mahol
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
23 #include "libavutil/opt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "formats.h"
27 
28 typedef struct AudioContrastContext {
29  const AVClass *class;
30  float contrast;
31  void (*filter)(void **dst, const void **src,
32  int nb_samples, int channels, float contrast);
34 
35 #define OFFSET(x) offsetof(AudioContrastContext, x)
36 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
37 
38 static const AVOption acontrast_options[] = {
39  { "contrast", "set contrast", OFFSET(contrast), AV_OPT_TYPE_FLOAT, {.dbl=33}, 0, 100, A },
40  { NULL }
41 };
42 
43 AVFILTER_DEFINE_CLASS(acontrast);
44 
46 {
49  static const enum AVSampleFormat sample_fmts[] = {
53  };
54  int ret;
55 
57  if (!formats)
58  return AVERROR(ENOMEM);
60  if (ret < 0)
61  return ret;
62 
64  if (!layouts)
65  return AVERROR(ENOMEM);
66 
68  if (ret < 0)
69  return ret;
70 
73 }
74 
75 static void filter_flt(void **d, const void **s,
76  int nb_samples, int channels,
77  float contrast)
78 {
79  const float *src = s[0];
80  float *dst = d[0];
81  int n, c;
82 
83  for (n = 0; n < nb_samples; n++) {
84  for (c = 0; c < channels; c++) {
85  float d = src[c] * M_PI_2;
86 
87  dst[c] = sinf(d + contrast * sinf(d * 4));
88  }
89 
90  dst += c;
91  src += c;
92  }
93 }
94 
95 static void filter_dbl(void **d, const void **s,
96  int nb_samples, int channels,
97  float contrast)
98 {
99  const double *src = s[0];
100  double *dst = d[0];
101  int n, c;
102 
103  for (n = 0; n < nb_samples; n++) {
104  for (c = 0; c < channels; c++) {
105  double d = src[c] * M_PI_2;
106 
107  dst[c] = sin(d + contrast * sin(d * 4));
108  }
109 
110  dst += c;
111  src += c;
112  }
113 }
114 
115 static void filter_fltp(void **d, const void **s,
116  int nb_samples, int channels,
117  float contrast)
118 {
119  int n, c;
120 
121  for (c = 0; c < channels; c++) {
122  const float *src = s[c];
123  float *dst = d[c];
124 
125  for (n = 0; n < nb_samples; n++) {
126  float d = src[n] * M_PI_2;
127 
128  dst[n] = sinf(d + contrast * sinf(d * 4));
129  }
130  }
131 }
132 
133 static void filter_dblp(void **d, const void **s,
134  int nb_samples, int channels,
135  float contrast)
136 {
137  int n, c;
138 
139  for (c = 0; c < channels; c++) {
140  const double *src = s[c];
141  double *dst = d[c];
142 
143  for (n = 0; n < nb_samples; n++) {
144  double d = src[n] * M_PI_2;
145 
146  dst[n] = sin(d + contrast * sin(d * 4));
147  }
148  }
149 }
150 
152 {
153  AVFilterContext *ctx = inlink->dst;
154  AudioContrastContext *s = ctx->priv;
155 
156  switch (inlink->format) {
157  case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break;
158  case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break;
159  case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
160  case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
161  }
162 
163  return 0;
164 }
165 
167 {
168  AVFilterContext *ctx = inlink->dst;
169  AVFilterLink *outlink = ctx->outputs[0];
170  AudioContrastContext *s = ctx->priv;
171  AVFrame *out;
172 
173  if (av_frame_is_writable(in)) {
174  out = in;
175  } else {
176  out = ff_get_audio_buffer(outlink, in->nb_samples);
177  if (!out) {
178  av_frame_free(&in);
179  return AVERROR(ENOMEM);
180  }
182  }
183 
184  s->filter((void **)out->extended_data, (const void **)in->extended_data,
185  in->nb_samples, in->channels, s->contrast / 750);
186 
187  if (out != in)
188  av_frame_free(&in);
189 
190  return ff_filter_frame(outlink, out);
191 }
192 
193 static const AVFilterPad inputs[] = {
194  {
195  .name = "default",
196  .type = AVMEDIA_TYPE_AUDIO,
197  .filter_frame = filter_frame,
198  .config_props = config_input,
199  },
200  { NULL }
201 };
202 
203 static const AVFilterPad outputs[] = {
204  {
205  .name = "default",
206  .type = AVMEDIA_TYPE_AUDIO,
207  },
208  { NULL }
209 };
210 
212  .name = "acontrast",
213  .description = NULL_IF_CONFIG_SMALL("Simple audio dynamic range compression/expansion filter."),
214  .query_formats = query_formats,
215  .priv_size = sizeof(AudioContrastContext),
216  .priv_class = &acontrast_class,
217  .inputs = inputs,
218  .outputs = outputs,
219 };
formats
formats
Definition: signature.h:48
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:86
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
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
ff_make_format_list
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
out
FILE * out
Definition: movenc.c:54
n
int n
Definition: avisynth_c.h:760
ff_set_common_channel_layouts
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:549
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:686
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
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:202
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:410
AudioConvert::channels
int channels
Definition: audio_convert.c:54
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
M_PI_2
#define M_PI_2
Definition: mathematics.h:55
AVOption
AVOption.
Definition: opt.h:246
channels
channels
Definition: aptx.c:30
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(acontrast)
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
AudioContrastContext::contrast
float contrast
Definition: af_acontrast.c:30
acontrast_options
static const AVOption acontrast_options[]
Definition: af_acontrast.c:38
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
formats.h
AudioContrastContext
Definition: af_acontrast.c:28
src
#define src
Definition: vp8dsp.c:254
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
ff_set_common_formats
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:568
s
#define s(width, name)
Definition: cbs_vp9.c:257
filter_dbl
static void filter_dbl(void **d, const void **s, int nb_samples, int channels, float contrast)
Definition: af_acontrast.c:95
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
ctx
AVFormatContext * ctx
Definition: movenc.c:48
ff_af_acontrast
AVFilter ff_af_acontrast
Definition: af_acontrast.c:211
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
AudioContrastContext::filter
void(* filter)(void **dst, const void **src, int nb_samples, int channels, float contrast)
Definition: af_acontrast.c:31
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:654
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_acontrast.c:151
sinf
#define sinf(x)
Definition: libm.h:419
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
OFFSET
#define OFFSET(x)
Definition: af_acontrast.c:35
filter_flt
static void filter_flt(void **d, const void **s, int nb_samples, int channels, float contrast)
Definition: af_acontrast.c:75
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:188
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_acontrast.c:45
A
#define A
Definition: af_acontrast.c:36
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:226
in
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
Definition: audio_convert.c:326
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
AVFilter
Filter definition.
Definition: avfilter.h:144
ret
ret
Definition: filter_design.txt:187
filter_fltp
static void filter_fltp(void **d, const void **s, int nb_samples, int channels, float contrast)
Definition: af_acontrast.c:115
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
channel_layout.h
avfilter.h
inputs
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_acontrast.c:166
audio.h
filter_dblp
static void filter_dblp(void **d, const void **s, int nb_samples, int channels, float contrast)
Definition: af_acontrast.c:133
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63