FFmpeg  2.1.1
af_volume.c
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1 /*
2  * Copyright (c) 2011 Stefano Sabatini
3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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 /**
23  * @file
24  * audio volume filter
25  */
26 
27 #include "libavutil/channel_layout.h"
28 #include "libavutil/common.h"
29 #include "libavutil/eval.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/opt.h"
32 #include "audio.h"
33 #include "avfilter.h"
34 #include "formats.h"
35 #include "internal.h"
36 #include "af_volume.h"
37 
38 static const char *precision_str[] = {
39  "fixed", "float", "double"
40 };
41 
42 #define OFFSET(x) offsetof(VolumeContext, x)
43 #define A AV_OPT_FLAG_AUDIO_PARAM
44 #define F AV_OPT_FLAG_FILTERING_PARAM
45 
46 static const AVOption volume_options[] = {
47  { "volume", "set volume adjustment",
48  OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A|F },
49  { "precision", "select mathematical precision",
50  OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
51  { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" },
52  { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" },
53  { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
54  { NULL }
55 };
56 
57 AVFILTER_DEFINE_CLASS(volume);
58 
59 static av_cold int init(AVFilterContext *ctx)
60 {
61  VolumeContext *vol = ctx->priv;
62 
63  if (vol->precision == PRECISION_FIXED) {
64  vol->volume_i = (int)(vol->volume * 256 + 0.5);
65  vol->volume = vol->volume_i / 256.0;
66  av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
67  vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
68  } else {
69  av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
70  vol->volume, 20.0*log(vol->volume)/M_LN10,
71  precision_str[vol->precision]);
72  }
73 
74  return 0;
75 }
76 
78 {
79  VolumeContext *vol = ctx->priv;
80  AVFilterFormats *formats = NULL;
82  static const enum AVSampleFormat sample_fmts[][7] = {
83  [PRECISION_FIXED] = {
91  },
92  [PRECISION_FLOAT] = {
96  },
97  [PRECISION_DOUBLE] = {
101  }
102  };
103 
104  layouts = ff_all_channel_layouts();
105  if (!layouts)
106  return AVERROR(ENOMEM);
107  ff_set_common_channel_layouts(ctx, layouts);
108 
109  formats = ff_make_format_list(sample_fmts[vol->precision]);
110  if (!formats)
111  return AVERROR(ENOMEM);
112  ff_set_common_formats(ctx, formats);
113 
114  formats = ff_all_samplerates();
115  if (!formats)
116  return AVERROR(ENOMEM);
117  ff_set_common_samplerates(ctx, formats);
118 
119  return 0;
120 }
121 
122 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
123  int nb_samples, int volume)
124 {
125  int i;
126  for (i = 0; i < nb_samples; i++)
127  dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
128 }
129 
130 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
131  int nb_samples, int volume)
132 {
133  int i;
134  for (i = 0; i < nb_samples; i++)
135  dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
136 }
137 
138 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
139  int nb_samples, int volume)
140 {
141  int i;
142  int16_t *smp_dst = (int16_t *)dst;
143  const int16_t *smp_src = (const int16_t *)src;
144  for (i = 0; i < nb_samples; i++)
145  smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
146 }
147 
148 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
149  int nb_samples, int volume)
150 {
151  int i;
152  int16_t *smp_dst = (int16_t *)dst;
153  const int16_t *smp_src = (const int16_t *)src;
154  for (i = 0; i < nb_samples; i++)
155  smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
156 }
157 
158 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
159  int nb_samples, int volume)
160 {
161  int i;
162  int32_t *smp_dst = (int32_t *)dst;
163  const int32_t *smp_src = (const int32_t *)src;
164  for (i = 0; i < nb_samples; i++)
165  smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
166 }
167 
169 {
170  vol->samples_align = 1;
171 
172  switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
173  case AV_SAMPLE_FMT_U8:
174  if (vol->volume_i < 0x1000000)
176  else
178  break;
179  case AV_SAMPLE_FMT_S16:
180  if (vol->volume_i < 0x10000)
182  else
184  break;
185  case AV_SAMPLE_FMT_S32:
187  break;
188  case AV_SAMPLE_FMT_FLT:
189  avpriv_float_dsp_init(&vol->fdsp, 0);
190  vol->samples_align = 4;
191  break;
192  case AV_SAMPLE_FMT_DBL:
193  avpriv_float_dsp_init(&vol->fdsp, 0);
194  vol->samples_align = 8;
195  break;
196  }
197 
198  if (ARCH_X86)
199  ff_volume_init_x86(vol);
200 }
201 
202 static int config_output(AVFilterLink *outlink)
203 {
204  AVFilterContext *ctx = outlink->src;
205  VolumeContext *vol = ctx->priv;
206  AVFilterLink *inlink = ctx->inputs[0];
207 
208  vol->sample_fmt = inlink->format;
210  vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
211 
212  volume_init(vol);
213 
214  return 0;
215 }
216 
217 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
218 {
219  VolumeContext *vol = inlink->dst->priv;
220  AVFilterLink *outlink = inlink->dst->outputs[0];
221  int nb_samples = buf->nb_samples;
222  AVFrame *out_buf;
223 
224  if (vol->volume == 1.0 || vol->volume_i == 256)
225  return ff_filter_frame(outlink, buf);
226 
227  /* do volume scaling in-place if input buffer is writable */
228  if (av_frame_is_writable(buf)) {
229  out_buf = buf;
230  } else {
231  out_buf = ff_get_audio_buffer(inlink, nb_samples);
232  if (!out_buf)
233  return AVERROR(ENOMEM);
234  av_frame_copy_props(out_buf, buf);
235  }
236 
237  if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
238  int p, plane_samples;
239 
241  plane_samples = FFALIGN(nb_samples, vol->samples_align);
242  else
243  plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
244 
245  if (vol->precision == PRECISION_FIXED) {
246  for (p = 0; p < vol->planes; p++) {
247  vol->scale_samples(out_buf->extended_data[p],
248  buf->extended_data[p], plane_samples,
249  vol->volume_i);
250  }
252  for (p = 0; p < vol->planes; p++) {
253  vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
254  (const float *)buf->extended_data[p],
255  vol->volume, plane_samples);
256  }
257  } else {
258  for (p = 0; p < vol->planes; p++) {
259  vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
260  (const double *)buf->extended_data[p],
261  vol->volume, plane_samples);
262  }
263  }
264  }
265 
266  if (buf != out_buf)
267  av_frame_free(&buf);
268 
269  return ff_filter_frame(outlink, out_buf);
270 }
271 
273  {
274  .name = "default",
275  .type = AVMEDIA_TYPE_AUDIO,
276  .filter_frame = filter_frame,
277  },
278  { NULL }
279 };
280 
282  {
283  .name = "default",
284  .type = AVMEDIA_TYPE_AUDIO,
285  .config_props = config_output,
286  },
287  { NULL }
288 };
289 
291  .name = "volume",
292  .description = NULL_IF_CONFIG_SMALL("Change input volume."),
293  .query_formats = query_formats,
294  .priv_size = sizeof(VolumeContext),
295  .priv_class = &volume_class,
296  .init = init,
297  .inputs = avfilter_af_volume_inputs,
298  .outputs = avfilter_af_volume_outputs,
300 };
This structure describes decoded (raw) audio or video data.
Definition: frame.h:96
AVOption.
Definition: opt.h:253
const char * name
Filter name.
Definition: avfilter.h:468
void * priv
private data for use by the filter
Definition: avfilter.h:648
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:111
enum PrecisionType precision
Definition: af_volume.h:41
static av_cold void volume_init(VolumeContext *vol)
Definition: af_volume.c:168
AVFloatDSPContext fdsp
Definition: af_volume.h:40
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:140
static av_cold int init(AVFilterContext *ctx)
Definition: af_volume.c:59
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
static enum AVSampleFormat formats[]
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:382
#define av_cold
Definition: avcodec.h:653
unsigned 8 bits
Definition: samplefmt.h:51
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:294
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic &quot;enable&quot; expression option that can be used to enable or disable a fil...
Definition: avfilter.h:445
const char * name
Pad name.
Definition: internal.h:66
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1118
uint8_t
signed 32 bits, planar
Definition: samplefmt.h:59
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only &quot;metadata&quot; fields from src to dst.
Definition: frame.c:446
signed 32 bits
Definition: samplefmt.h:53
int samples_align
Definition: af_volume.h:50
void(* scale_samples)(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.h:48
#define AV_LOG_VERBOSE
Detailed information.
Definition: avcodec.h:4163
void 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:531
#define A(x)
Definition: vp56_arith.h:28
signed 16 bits
Definition: samplefmt.h:52
static void scale_samples_s32(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:158
A filter pad used for either input or output.
Definition: internal.h:60
audio volume filter
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 ARCH_X86
Definition: config.h:35
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:151
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:123
void(* vector_dmul_scalar)(double *dst, const double *src, double mul, int len)
Multiply a vector of double by a scalar double.
Definition: float_dsp.h:84
unsigned 8 bits, planar
Definition: samplefmt.h:57
static int query_formats(AVFilterContext *ctx)
Definition: af_volume.c:77
#define F
Definition: af_volume.c:44
static void scale_samples_s16(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:138
static void scale_samples_u8_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:130
int32_t
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:69
static const AVFilterPad avfilter_af_volume_outputs[]
Definition: af_volume.c:281
static void scale_samples_u8(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:122
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (w...
Definition: formats.c:388
A list of supported channel layouts.
Definition: formats.h:85
Main libavfilter public API header.
double volume
Definition: af_volume.h:42
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:642
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:157
enum AVSampleFormat sample_fmt
Definition: af_volume.h:46
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
AVS_Value src
Definition: avisynth_c.h:523
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition: samplefmt.c:73
void * buf
Definition: avisynth_c.h:594
Filter definition.
Definition: avfilter.h:464
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:102
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:635
static const AVOption volume_options[]
Definition: af_volume.c:46
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:382
static void scale_samples_s16_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:148
static int flags
Definition: cpu.c:45
static const char * precision_str[]
Definition: af_volume.c:38
av_cold void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
Initialize a float DSP context.
Definition: float_dsp.c:118
void ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:519
void ff_volume_init_x86(VolumeContext *vol)
static const AVFilterPad avfilter_af_volume_inputs[]
Definition: af_volume.c:272
#define M_LN10
Definition: mathematics.h:37
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_volume.c:217
#define FFALIGN(x, a)
Definition: avcodec.h:930
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:118
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:301
#define OFFSET(x)
Definition: af_volume.c:42
A list of supported formats for one end of a filter link.
Definition: formats.h:64
#define AVERROR(e)
float, planar
Definition: samplefmt.h:60
An instance of a filter.
Definition: avfilter.h:627
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:700
signed 16 bits, planar
Definition: samplefmt.h:58
void 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:512
internal API functions
double, planar
Definition: samplefmt.h:61
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:150
AVFilter avfilter_af_volume
Definition: af_volume.c:290
static int config_output(AVFilterLink *outlink)
Definition: af_volume.c:202