FFmpeg
af_aderivative.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 
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[] = {
41  AV_SAMPLE_FMT_NONE
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 
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 
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:550
static const AVFilterPad aderivative_outputs[]
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
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:1093
#define av_cold
Definition: attributes.h:82
void(* filter)(void **dst, void **prv, const void **src, int nb_samples, int channels)
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:569
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
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:601
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 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)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
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
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:394
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
static int query_formats(AVFilterContext *ctx)
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
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:409
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
#define DERIVATIVE(name, type)
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
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:557
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654