FFmpeg
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 
45 
47 {
48  DCShiftContext *s = ctx->priv;
49 
50  s->limiterthreshold = 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_counts();
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 
85 {
86  AVFilterContext *ctx = inlink->dst;
87  AVFilterLink *outlink = ctx->outputs[0];
88  AVFrame *out;
89  DCShiftContext *s = ctx->priv;
90  int i, j;
91  double dcshift = s->dcshift;
92 
93  if (av_frame_is_writable(in)) {
94  out = in;
95  } else {
96  out = ff_get_audio_buffer(outlink, in->nb_samples);
97  if (!out) {
98  av_frame_free(&in);
99  return AVERROR(ENOMEM);
100  }
101  av_frame_copy_props(out, in);
102  }
103 
104  if (s->limitergain > 0) {
105  for (i = 0; i < inlink->channels; i++) {
106  const int32_t *src = (int32_t *)in->extended_data[i];
107  int32_t *dst = (int32_t *)out->extended_data[i];
108 
109  for (j = 0; j < in->nb_samples; j++) {
110  double d;
111 
112  d = src[j];
113 
114  if (d > s->limiterthreshold && dcshift > 0) {
115  d = (d - s->limiterthreshold) * s->limitergain /
116  (INT32_MAX - s->limiterthreshold) +
118  } else if (d < -s->limiterthreshold && dcshift < 0) {
119  d = (d + s->limiterthreshold) * s->limitergain /
120  (INT32_MAX - s->limiterthreshold) -
122  } else {
123  d = dcshift * INT32_MAX + d;
124  }
125 
126  dst[j] = av_clipl_int32(d);
127  }
128  }
129  } else {
130  for (i = 0; i < inlink->channels; i++) {
131  const int32_t *src = (int32_t *)in->extended_data[i];
132  int32_t *dst = (int32_t *)out->extended_data[i];
133 
134  for (j = 0; j < in->nb_samples; j++) {
135  double d = dcshift * (INT32_MAX + 1.) + src[j];
136 
137  dst[j] = av_clipl_int32(d);
138  }
139  }
140  }
141 
142  if (out != in)
143  av_frame_free(&in);
144  return ff_filter_frame(outlink, out);
145 }
146 static const AVFilterPad dcshift_inputs[] = {
147  {
148  .name = "default",
149  .type = AVMEDIA_TYPE_AUDIO,
150  .filter_frame = filter_frame,
151  },
152  { NULL }
153 };
154 
155 static const AVFilterPad dcshift_outputs[] = {
156  {
157  .name = "default",
158  .type = AVMEDIA_TYPE_AUDIO,
159  },
160  { NULL }
161 };
162 
164  .name = "dcshift",
165  .description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
166  .query_formats = query_formats,
167  .priv_size = sizeof(DCShiftContext),
168  .priv_class = &dcshift_class,
169  .init = init,
170  .inputs = dcshift_inputs,
171  .outputs = dcshift_outputs,
173 };
#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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
Main libavfilter public API header.
double dcshift
Definition: af_dcshift.c:30
AVFILTER_DEFINE_CLASS(dcshift)
#define src
Definition: vp8dsp.c:254
double limiterthreshold
Definition: af_dcshift.c:31
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:125
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
AVOptions.
#define OFFSET(x)
Definition: af_dcshift.c:35
A filter pad used for either input or output.
Definition: internal.h:54
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
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
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
AVFilter ff_af_dcshift
Definition: af_dcshift.c:163
void * priv
private data for use by the filter
Definition: avfilter.h:353
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:68
static const AVFilterPad dcshift_inputs[]
Definition: af_dcshift.c:146
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
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
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
static const AVFilterPad dcshift_outputs[]
Definition: af_dcshift.c:155
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
#define flags(name, subs,...)
Definition: cbs_av1.c:561
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 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
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
#define A
Definition: af_dcshift.c:36
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
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
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