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af_dcshift.c
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1 /*
2  * Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
3  * Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
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 
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "internal.h"
27 
28 typedef struct DCShiftContext {
29  const AVClass *class;
30  double dcshift;
32  double limitergain;
34 
35 #define OFFSET(x) offsetof(DCShiftContext, x)
36 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
37 
38 static const AVOption dcshift_options[] = {
39  { "shift", "set DC shift", OFFSET(dcshift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
40  { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
41  { NULL }
42 };
43 
44 AVFILTER_DEFINE_CLASS(dcshift);
45 
46 static av_cold int init(AVFilterContext *ctx)
47 {
48  DCShiftContext *s = ctx->priv;
49 
50  s->limiterthreshhold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
51 
52  return 0;
53 }
54 
56 {
59  static const enum AVSampleFormat sample_fmts[] = {
61  };
62  int ret;
63 
64  layouts = ff_all_channel_layouts();
65  if (!layouts)
66  return AVERROR(ENOMEM);
67  ret = ff_set_common_channel_layouts(ctx, layouts);
68  if (ret < 0)
69  return ret;
70 
71  formats = ff_make_format_list(sample_fmts);
72  if (!formats)
73  return AVERROR(ENOMEM);
74  ret = ff_set_common_formats(ctx, formats);
75  if (ret < 0)
76  return ret;
77 
78  formats = ff_all_samplerates();
79  if (!formats)
80  return AVERROR(ENOMEM);
81  return ff_set_common_samplerates(ctx, formats);
82 }
83 
84 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
85 {
86  AVFilterContext *ctx = inlink->dst;
87  AVFilterLink *outlink = ctx->outputs[0];
88  AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples);
89  DCShiftContext *s = ctx->priv;
90  int i, j;
91  double dcshift = s->dcshift;
92 
93  if (!out) {
94  av_frame_free(&in);
95  return AVERROR(ENOMEM);
96  }
97  av_frame_copy_props(out, in);
98 
99  if (s->limitergain > 0) {
100  for (i = 0; i < inlink->channels; i++) {
101  const int32_t *src = (int32_t *)in->extended_data[i];
102  int32_t *dst = (int32_t *)out->extended_data[i];
103 
104  for (j = 0; j < in->nb_samples; j++) {
105  double d;
106 
107  d = src[j];
108 
109  if (d > s->limiterthreshhold && dcshift > 0) {
110  d = (d - s->limiterthreshhold) * s->limitergain /
111  (INT32_MAX - s->limiterthreshhold) +
112  s->limiterthreshhold + dcshift;
113  } else if (d < -s->limiterthreshhold && dcshift < 0) {
114  d = (d + s->limiterthreshhold) * s->limitergain /
115  (INT32_MAX - s->limiterthreshhold) -
116  s->limiterthreshhold + dcshift;
117  } else {
118  d = dcshift * INT32_MAX + d;
119  }
120 
121  dst[j] = av_clipl_int32(d);
122  }
123  }
124  } else {
125  for (i = 0; i < inlink->channels; i++) {
126  const int32_t *src = (int32_t *)in->extended_data[i];
127  int32_t *dst = (int32_t *)out->extended_data[i];
128 
129  for (j = 0; j < in->nb_samples; j++) {
130  double d = dcshift * (INT32_MAX + 1.) + src[j];
131 
132  dst[j] = av_clipl_int32(d);
133  }
134  }
135  }
136 
137  av_frame_free(&in);
138  return ff_filter_frame(outlink, out);
139 }
140 static const AVFilterPad dcshift_inputs[] = {
141  {
142  .name = "default",
143  .type = AVMEDIA_TYPE_AUDIO,
144  .filter_frame = filter_frame,
145  },
146  { NULL }
147 };
148 
149 static const AVFilterPad dcshift_outputs[] = {
150  {
151  .name = "default",
152  .type = AVMEDIA_TYPE_AUDIO,
153  },
154  { NULL }
155 };
156 
158  .name = "dcshift",
159  .description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
160  .query_formats = query_formats,
161  .priv_size = sizeof(DCShiftContext),
162  .priv_class = &dcshift_class,
163  .init = init,
164  .inputs = dcshift_inputs,
165  .outputs = dcshift_outputs,
167 };
#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:523
const char * s
Definition: avisynth_c.h:631
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
AVOption.
Definition: opt.h:255
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:248
Main libavfilter public API header.
double dcshift
Definition: af_dcshift.c:30
static enum AVSampleFormat formats[]
AVFILTER_DEFINE_CLASS(dcshift)
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
static const AVOption dcshift_options[]
Definition: af_dcshift.c:38
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:451
const char * name
Pad name.
Definition: internal.h:67
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1145
#define av_cold
Definition: attributes.h:74
AVOptions.
double limiterthreshhold
Definition: af_dcshift.c:31
#define OFFSET(x)
Definition: af_dcshift.c:35
A filter pad used for either input or output.
Definition: internal.h:61
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:542
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:70
#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:148
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:175
AVFilter ff_af_dcshift
Definition: af_dcshift.c:157
void * priv
private data for use by the filter
Definition: avfilter.h:654
static int query_formats(AVFilterContext *ctx)
Definition: af_dcshift.c:55
double limitergain
Definition: af_dcshift.c:32
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_dcshift.c:84
signed 32 bits, planar
Definition: samplefmt.h:69
ret
Definition: avfilter.c:974
static const AVFilterPad dcshift_inputs[]
Definition: af_dcshift.c:140
int32_t
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (w...
Definition: formats.c:385
A list of supported channel layouts.
Definition: formats.h:85
AVS_Value src
Definition: avisynth_c.h:482
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
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:470
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:239
const char * name
Filter name.
Definition: avfilter.h:474
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:648
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
static const AVFilterPad dcshift_outputs[]
Definition: af_dcshift.c:149
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:379
static int flags
Definition: cpu.c:47
static av_cold int init(AVFilterContext *ctx)
Definition: af_dcshift.c:46
A list of supported formats for one end of a filter link.
Definition: formats.h:64
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-> out
An instance of a filter.
Definition: avfilter.h:633
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define A
Definition: af_dcshift.c:36
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:215
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:530
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:548