FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
af_aderivative.c
Go to the documentation of this file.
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 
19 #include "audio.h"
20 #include "avfilter.h"
21 #include "internal.h"
22 
23 typedef struct ADerivativeContext {
24  const AVClass *class;
26  void (*filter)(void **dst, void **prv, const void **src,
27  int nb_samples, int channels);
29 
31 {
34  static const enum AVSampleFormat derivative_sample_fmts[] = {
38  };
39  static const enum AVSampleFormat integral_sample_fmts[] = {
42  };
43  int ret;
44 
45  formats = ff_make_format_list(strcmp(ctx->filter->name, "aintegral") ?
46  derivative_sample_fmts : integral_sample_fmts);
47  if (!formats)
48  return AVERROR(ENOMEM);
49  ret = ff_set_common_formats(ctx, formats);
50  if (ret < 0)
51  return ret;
52 
53  layouts = ff_all_channel_counts();
54  if (!layouts)
55  return AVERROR(ENOMEM);
56 
57  ret = ff_set_common_channel_layouts(ctx, layouts);
58  if (ret < 0)
59  return ret;
60 
61  formats = ff_all_samplerates();
62  return ff_set_common_samplerates(ctx, formats);
63 }
64 
65 #define DERIVATIVE(name, type) \
66 static void aderivative_## name ##p(void **d, void **p, const void **s, \
67  int nb_samples, int channels) \
68 { \
69  int n, c; \
70  \
71  for (c = 0; c < channels; c++) { \
72  const type *src = s[c]; \
73  type *dst = d[c]; \
74  type *prv = p[c]; \
75  \
76  for (n = 0; n < nb_samples; n++) { \
77  const type current = src[n]; \
78  \
79  dst[n] = current - prv[0]; \
80  prv[0] = current; \
81  } \
82  } \
83 }
84 
85 DERIVATIVE(flt, float)
86 DERIVATIVE(dbl, double)
87 DERIVATIVE(s16, int16_t)
88 DERIVATIVE(s32, int32_t)
89 
90 #define INTEGRAL(name, type) \
91 static void aintegral_## name ##p(void **d, void **p, const void **s, \
92  int nb_samples, int channels) \
93 { \
94  int n, c; \
95  \
96  for (c = 0; c < channels; c++) { \
97  const type *src = s[c]; \
98  type *dst = d[c]; \
99  type *prv = p[c]; \
100  \
101  for (n = 0; n < nb_samples; n++) { \
102  const type current = src[n]; \
103  \
104  dst[n] = current + prv[0]; \
105  prv[0] = dst[n]; \
106  } \
107  } \
108 }
109 
110 INTEGRAL(flt, float)
111 INTEGRAL(dbl, double)
112 
113 static int config_input(AVFilterLink *inlink)
114 {
115  AVFilterContext *ctx = inlink->dst;
116  ADerivativeContext *s = ctx->priv;
117 
118  switch (inlink->format) {
119  case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break;
120  case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break;
121  case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break;
122  case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break;
123  }
124 
125  if (strcmp(ctx->filter->name, "aintegral"))
126  return 0;
127 
128  switch (inlink->format) {
129  case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break;
130  case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break;
131  }
132 
133  return 0;
134 }
135 
136 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
137 {
138  AVFilterContext *ctx = inlink->dst;
139  ADerivativeContext *s = ctx->priv;
140  AVFilterLink *outlink = ctx->outputs[0];
141  AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
142 
143  if (!out) {
144  av_frame_free(&in);
145  return AVERROR(ENOMEM);
146  }
147  av_frame_copy_props(out, in);
148 
149  if (!s->prev) {
150  s->prev = ff_get_audio_buffer(inlink, 1);
151  if (!s->prev) {
152  av_frame_free(&in);
153  return AVERROR(ENOMEM);
154  }
155  }
156 
157  s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
158  in->nb_samples, in->channels);
159 
160  av_frame_free(&in);
161  return ff_filter_frame(outlink, out);
162 }
163 
165 {
166  ADerivativeContext *s = ctx->priv;
167 
168  av_frame_free(&s->prev);
169 }
170 
171 static const AVFilterPad aderivative_inputs[] = {
172  {
173  .name = "default",
174  .type = AVMEDIA_TYPE_AUDIO,
175  .filter_frame = filter_frame,
176  .config_props = config_input,
177  },
178  { NULL }
179 };
180 
182  {
183  .name = "default",
184  .type = AVMEDIA_TYPE_AUDIO,
185  },
186  { NULL }
187 };
188 
190  .name = "aderivative",
191  .description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."),
192  .query_formats = query_formats,
193  .priv_size = sizeof(ADerivativeContext),
194  .uninit = uninit,
195  .inputs = aderivative_inputs,
196  .outputs = aderivative_outputs,
197 };
198 
200  .name = "aintegral",
201  .description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
202  .query_formats = query_formats,
203  .priv_size = sizeof(ADerivativeContext),
204  .uninit = uninit,
205  .inputs = aderivative_inputs,
206  .outputs = aderivative_outputs,
207 };
float, planar
Definition: samplefmt.h:69
#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:549
static const AVFilterPad aderivative_outputs[]
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
Main libavfilter public API header.
channels
Definition: aptx.c:30
double, planar
Definition: samplefmt.h:70
#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
static const AVFilterPad aderivative_inputs[]
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
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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 int config_input(AVFilterLink *inlink)
int channels
number of audio channels, only used for audio.
Definition: frame.h:531
signed 32 bits, planar
Definition: samplefmt.h:68
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
#define INTEGRAL(name, type)
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
AVFilter ff_af_aderivative
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
AVFilter ff_af_aintegral
static av_cold void uninit(AVFilterContext *ctx)
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_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)
void(* filter)(void **dst, void **prv, const void **src, int nb_samples, int channels)
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
FILE * out
Definition: movenc.c:54
signed 16 bits, planar
Definition: samplefmt.h:67
formats
Definition: signature.h:48
internal API functions
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:273
#define DERIVATIVE(name, type)
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
const AVFilter * filter
the AVFilter of which this is an instance
Definition: avfilter.h:341
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:654