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lavfi: add volume filter

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This commit is contained in:
Stefano Sabatini 2011-11-01 21:42:14 +01:00
parent 1fc7077115
commit 618ac71354
6 changed files with 247 additions and 2 deletions
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2
Changelog
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@ -73,6 +73,8 @@ easier to use. The changes are:
- Video Decoder Acceleration (VDA) HWAccel module.
- replacement Indeo 3 decoder
- new ffmpeg option: -map_channel
- volume audio filter added
version 0.8:

50
doc/filters.texi
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@ -224,6 +224,56 @@ expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5}
@var{c6} @var{c7}]"
@end table
@section volume
Adjust the input audio volume.
The filter accepts exactly one parameter @var{vol}, which expresses
how the audio volume will be increased or decresed.
Output values are clipped to the maximum value.
If @var{vol} is expressed as a decimal number, and the output audio
volume is given by the relation:
@example
@var{output_volume} = @var{vol} * @var{input_volume}
@end example
If @var{vol} is expressed as a decimal number followed by the string
"dB", the value represents the requested change in decibels of the
input audio power, and the output audio volume is given by the
relation:
@example
@var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
@end example
Otherwise @var{vol} is considered an expression and its evaluated
value is used for computing the output audio volume according to the
first relation.
Default value for @var{vol} is 1.0.
@subsection Examples
@itemize
@item
Half the input audio volume:
@example
volume=0.5
@end example
The above example is equivalent to:
@example
volume=1/2
@end example
@item
Decrease input audio power by 12 decibels:
@example
volume=-12dB
@end example
@end itemize
@c man end AUDIO FILTERS
@chapter Audio Sources

1
libavfilter/Makefile
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@ -28,6 +28,7 @@ OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o

191
libavfilter/af_volume.c Normal file
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@ -0,0 +1,191 @@
/*
* Copyright (c) 2011 Stefano Sabatini
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio volume filter
* based on ffmpeg.c code
*/
#include "libavutil/audioconvert.h"
#include "libavutil/eval.h"
#include "avfilter.h"
typedef struct {
double volume;
int volume_i;
} VolumeContext;
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
{
VolumeContext *vol = ctx->priv;
char *tail;
int ret = 0;
vol->volume = 1.0;
if (args) {
/* parse the number as a decimal number */
double d = strtod(args, &tail);
if (*tail) {
if (!strcmp(tail, "dB")) {
/* consider the argument an adjustement in decibels */
if (!strcmp(tail, "dB")) {
d = exp10(d/20);
}
} else {
/* parse the argument as an expression */
ret = av_expr_parse_and_eval(&d, args, NULL, NULL,
NULL, NULL, NULL, NULL,
NULL, 0, ctx);
}
}
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR,
"Invalid volume argument '%s'\n", args);
return AVERROR(EINVAL);
}
if (d < 0 || d > 65536) { /* 65536 = INT_MIN / (128 * 256) */
av_log(ctx, AV_LOG_ERROR,
"Negative or too big volume value %f\n", d);
return AVERROR(EINVAL);
}
vol->volume = d;
}
vol->volume_i = (int)(vol->volume * 256 + 0.5);
av_log(ctx, AV_LOG_INFO, "volume=%f\n", vol->volume);
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int packing_fmts[] = { AVFILTER_PACKED, -1 };
formats = avfilter_make_all_channel_layouts();
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_channel_layouts(ctx, formats);
formats = avfilter_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_sample_formats(ctx, formats);
formats = avfilter_make_format_list(packing_fmts);
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_packing_formats(ctx, formats);
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
const int nb_samples = insamples->audio->nb_samples *
av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
const double volume = vol->volume;
const int volume_i = vol->volume_i;
int i;
if (volume_i != 256) {
switch (insamples->format) {
case AV_SAMPLE_FMT_U8:
{
uint8_t *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
int v = (((*p - 128) * volume_i + 128) >> 8) + 128;
*p++ = av_clip_uint8(v);
}
break;
}
case AV_SAMPLE_FMT_S16:
{
int16_t *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
int v = ((int64_t)*p * volume_i + 128) >> 8;
*p++ = av_clip_int16(v);
}
break;
}
case AV_SAMPLE_FMT_S32:
{
int32_t *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
int64_t v = (((int64_t)*p * volume_i + 128) >> 8);
*p++ = av_clipl_int32(v);
}
break;
}
case AV_SAMPLE_FMT_FLT:
{
float *p = (void *)insamples->data[0];
float scale = (float)volume;
for (i = 0; i < nb_samples; i++) {
*p++ *= scale;
}
break;
}
case AV_SAMPLE_FMT_DBL:
{
double *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
*p *= volume;
p++;
}
break;
}
}
}
avfilter_filter_samples(outlink, insamples);
}
AVFilter avfilter_af_volume = {
.name = "volume",
.description = NULL_IF_CONFIG_SMALL("Change input volume."),
.query_formats = query_formats,
.priv_size = sizeof(VolumeContext),
.init = init,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ|AV_PERM_WRITE},
{ .name = NULL}},
.outputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};

1
libavfilter/allfilters.c
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@ -39,6 +39,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
REGISTER_FILTER (VOLUME, volume, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);

4
libavfilter/avfilter.h
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@ -29,8 +29,8 @@
#include "libavutil/rational.h"
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 45
#define LIBAVFILTER_VERSION_MICRO 3
#define LIBAVFILTER_VERSION_MINOR 46
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \