FFmpeg  4.0
af_crystalizer.c
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1 /*
2  * Copyright (c) 2016 The FFmpeg Project
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 
22 #include "libavutil/opt.h"
23 #include "avfilter.h"
24 #include "audio.h"
25 #include "formats.h"
26 
27 typedef struct CrystalizerContext {
28  const AVClass *class;
29  float mult;
30  int clip;
32  void (*filter)(void **dst, void **prv, const void **src,
33  int nb_samples, int channels, float mult, int clip);
35 
36 #define OFFSET(x) offsetof(CrystalizerContext, x)
37 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
38 
39 static const AVOption crystalizer_options[] = {
40  { "i", "set intensity", OFFSET(mult), AV_OPT_TYPE_FLOAT, {.dbl=2.0}, 0, 10, A },
41  { "c", "enable clipping", OFFSET(clip), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
42  { NULL }
43 };
44 
45 AVFILTER_DEFINE_CLASS(crystalizer);
46 
48 {
51  static const enum AVSampleFormat sample_fmts[] = {
55  };
56  int ret;
57 
58  formats = ff_make_format_list(sample_fmts);
59  if (!formats)
60  return AVERROR(ENOMEM);
61  ret = ff_set_common_formats(ctx, formats);
62  if (ret < 0)
63  return ret;
64 
65  layouts = ff_all_channel_counts();
66  if (!layouts)
67  return AVERROR(ENOMEM);
68 
69  ret = ff_set_common_channel_layouts(ctx, layouts);
70  if (ret < 0)
71  return ret;
72 
73  formats = ff_all_samplerates();
74  return ff_set_common_samplerates(ctx, formats);
75 }
76 
77 static void filter_flt(void **d, void **p, const void **s,
78  int nb_samples, int channels,
79  float mult, int clip)
80 {
81  const float *src = s[0];
82  float *dst = d[0];
83  float *prv = p[0];
84  int n, c;
85 
86  for (n = 0; n < nb_samples; n++) {
87  for (c = 0; c < channels; c++) {
88  float current = src[c];
89 
90  dst[c] = current + (current - prv[c]) * mult;
91  prv[c] = current;
92  if (clip) {
93  dst[c] = av_clipf(dst[c], -1, 1);
94  }
95  }
96 
97  dst += c;
98  src += c;
99  }
100 }
101 
102 static void filter_dbl(void **d, void **p, const void **s,
103  int nb_samples, int channels,
104  float mult, int clip)
105 {
106  const double *src = s[0];
107  double *dst = d[0];
108  double *prv = p[0];
109  int n, c;
110 
111  for (n = 0; n < nb_samples; n++) {
112  for (c = 0; c < channels; c++) {
113  double current = src[c];
114 
115  dst[c] = current + (current - prv[c]) * mult;
116  prv[c] = current;
117  if (clip) {
118  dst[c] = av_clipd(dst[c], -1, 1);
119  }
120  }
121 
122  dst += c;
123  src += c;
124  }
125 }
126 
127 static void filter_fltp(void **d, void **p, const void **s,
128  int nb_samples, int channels,
129  float mult, int clip)
130 {
131  int n, c;
132 
133  for (c = 0; c < channels; c++) {
134  const float *src = s[c];
135  float *dst = d[c];
136  float *prv = p[c];
137 
138  for (n = 0; n < nb_samples; n++) {
139  float current = src[n];
140 
141  dst[n] = current + (current - prv[0]) * mult;
142  prv[0] = current;
143  if (clip) {
144  dst[n] = av_clipf(dst[n], -1, 1);
145  }
146  }
147  }
148 }
149 
150 static void filter_dblp(void **d, void **p, const void **s,
151  int nb_samples, int channels,
152  float mult, int clip)
153 {
154  int n, c;
155 
156  for (c = 0; c < channels; c++) {
157  const double *src = s[c];
158  double *dst = d[c];
159  double *prv = p[c];
160 
161  for (n = 0; n < nb_samples; n++) {
162  double current = src[n];
163 
164  dst[n] = current + (current - prv[0]) * mult;
165  prv[0] = current;
166  if (clip) {
167  dst[n] = av_clipd(dst[n], -1, 1);
168  }
169  }
170  }
171 }
172 
173 static int config_input(AVFilterLink *inlink)
174 {
175  AVFilterContext *ctx = inlink->dst;
176  CrystalizerContext *s = ctx->priv;
177 
178  switch (inlink->format) {
179  case AV_SAMPLE_FMT_FLT: s->filter = filter_flt; break;
180  case AV_SAMPLE_FMT_DBL: s->filter = filter_dbl; break;
181  case AV_SAMPLE_FMT_FLTP: s->filter = filter_fltp; break;
182  case AV_SAMPLE_FMT_DBLP: s->filter = filter_dblp; break;
183  }
184 
185  return 0;
186 }
187 
188 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
189 {
190  AVFilterContext *ctx = inlink->dst;
191  AVFilterLink *outlink = ctx->outputs[0];
192  CrystalizerContext *s = ctx->priv;
193  AVFrame *out;
194 
195  if (!s->prev) {
196  s->prev = ff_get_audio_buffer(inlink, 1);
197  if (!s->prev) {
198  av_frame_free(&in);
199  return AVERROR(ENOMEM);
200  }
201  }
202 
203  if (av_frame_is_writable(in)) {
204  out = in;
205  } else {
206  out = ff_get_audio_buffer(outlink, in->nb_samples);
207  if (!out) {
208  av_frame_free(&in);
209  return AVERROR(ENOMEM);
210  }
211  av_frame_copy_props(out, in);
212  }
213 
214  s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
215  in->nb_samples, in->channels, s->mult, s->clip);
216 
217  if (out != in)
218  av_frame_free(&in);
219 
220  return ff_filter_frame(outlink, out);
221 }
222 
224 {
225  CrystalizerContext *s = ctx->priv;
226 
227  av_frame_free(&s->prev);
228 }
229 
230 static const AVFilterPad inputs[] = {
231  {
232  .name = "default",
233  .type = AVMEDIA_TYPE_AUDIO,
234  .filter_frame = filter_frame,
235  .config_props = config_input,
236  },
237  { NULL }
238 };
239 
240 static const AVFilterPad outputs[] = {
241  {
242  .name = "default",
243  .type = AVMEDIA_TYPE_AUDIO,
244  },
245  { NULL }
246 };
247 
249  .name = "crystalizer",
250  .description = NULL_IF_CONFIG_SMALL("Simple expand audio dynamic range filter."),
251  .query_formats = query_formats,
252  .priv_size = sizeof(CrystalizerContext),
253  .priv_class = &crystalizer_class,
254  .uninit = uninit,
255  .inputs = inputs,
256  .outputs = outputs,
257 };
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
void(* filter)(void **dst, void **prv, const void **src, int nb_samples, int channels, float mult, int clip)
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
const char * s
Definition: avisynth_c.h:768
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
AVOption.
Definition: opt.h:246
Main libavfilter public API header.
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
channels
Definition: aptx.c:30
#define OFFSET(x)
double, planar
Definition: samplefmt.h:70
AVFILTER_DEFINE_CLASS(crystalizer)
static void filter_dblp(void **d, void **p, const void **s, int nb_samples, int channels, float mult, int clip)
#define src
Definition: vp8dsp.c:254
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define av_cold
Definition: attributes.h:82
AVOptions.
A filter pad used for either input or output.
Definition: internal.h:54
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
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
#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:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
static void filter_dbl(void **d, void **p, const void **s, int nb_samples, int channels, float mult, int clip)
int channels
number of audio channels, only used for audio.
Definition: frame.h:523
audio channel layout utility functions
static void filter_fltp(void **d, void **p, const void **s, int nb_samples, int channels, float mult, int clip)
AVFormatContext * ctx
Definition: movenc.c:48
int n
Definition: avisynth_c.h:684
#define A
A list of supported channel layouts.
Definition: formats.h:85
static void filter_flt(void **d, void **p, const void **s, int nb_samples, int channels, float mult, int clip)
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
typedef void(RENAME(mix_any_func_type))
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:592
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)\", 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
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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad outputs[]
static double c[64]
static const AVOption crystalizer_options[]
static const AVFilterPad inputs[]
static int config_input(AVFilterLink *inlink)
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:701
FILE * out
Definition: movenc.c:54
formats
Definition: signature.h:48
AVFilter ff_af_crystalizer
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:265
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
static av_cold void uninit(AVFilterContext *ctx)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:652