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[] = {
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);
50  if (ret < 0)
51  return ret;
52 
54  if (!layouts)
55  return AVERROR(ENOMEM);
56 
58  if (ret < 0)
59  return ret;
60 
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  }
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,
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,
207 };
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
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
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
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
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
ADerivativeContext::filter
void(* filter)(void **dst, void **prv, const void **src, int nb_samples, int channels)
Definition: af_aderivative.c:26
channels
channels
Definition: aptx.c:30
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
src
#define src
Definition: vp8dsp.c:254
aderivative_inputs
static const AVFilterPad aderivative_inputs[]
Definition: af_aderivative.c:171
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
av_cold
#define av_cold
Definition: attributes.h:84
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_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_aderivative.c:136
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_aderivative.c:113
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
ctx
AVFormatContext * ctx
Definition: movenc.c:48
int32_t
int32_t
Definition: audio_convert.c:194
ff_af_aintegral
AVFilter ff_af_aintegral
Definition: af_aderivative.c:199
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
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
ff_af_aderivative
AVFilter ff_af_aderivative
Definition: af_aderivative.c:189
ADerivativeContext::prev
AVFrame * prev
Definition: af_aderivative.c:25
INTEGRAL
#define INTEGRAL(name, type)
Definition: af_aderivative.c:90
inputs
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
Definition: filter_design.txt:243
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_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
internal.h
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
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_aderivative.c:30
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aderivative.c:164
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
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
avfilter.h
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
audio.h
aderivative_outputs
static const AVFilterPad aderivative_outputs[]
Definition: af_aderivative.c:181
DERIVATIVE
#define DERIVATIVE(name, type)
Definition: af_aderivative.c:65
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
ADerivativeContext
Definition: af_aderivative.c:23