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lavfi: add volume filter 

Based on the volume filter in FFmpeg written by Stefano Sabatini
<stefasab@gmail.com>.
tags/n1.1
Justin Ruggles 12 years ago
parent
9d5c62ba5b
commit
b384e031da
7 changed files with 424 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +53
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +314
    -0
      libavfilter/af_volume.c
  5. +53
    -0
      libavfilter/af_volume.h
  6. +1
    -0
      libavfilter/allfilters.c
  7. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

@@ -4,6 +4,7 @@ releases are sorted from youngest to oldest.
version <next>:
- ashowinfo audio filter
- 24-bit FLAC encoding
- audio volume filter


version 9_beta2:


+ 53
- 0
doc/filters.texi View File

@@ -359,6 +359,59 @@ not meant to be used directly, it is inserted automatically by libavfilter
whenever conversion is needed. Use the @var{aformat} filter to force a specific
conversion.

@section volume

Adjust the input audio volume.

The filter accepts the following named parameters:
@table @option

@item volume
Expresses how the audio volume will be increased or decreased.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:
@example
@var{output_volume} = @var{volume} * @var{input_volume}
@end example

Default value for @var{volume} is 1.0.

@item precision
Mathematical precision.

This determines which input sample formats will be allowed, which affects the
precision of the volume scaling.

@table @option
@item fixed
8-bit fixed-point; limits input sample format to U8, S16, and S32.
@item float
32-bit floating-point; limits input sample format to FLT. (default)
@item double
64-bit floating-point; limits input sample format to DBL.
@end table
@end table

@subsection Examples

@itemize
@item
Halve the input audio volume:
@example
volume=volume=0.5
volume=volume=1/2
volume=volume=-6.0206dB
@end example

@item
Increase input audio power by 6 decibels using fixed-point precision:
@example
volume=volume=6dB:precision=fixed
@end example
@end itemize

@c man end AUDIO FILTERS

@chapter Audio Sources


+ 1
- 0
libavfilter/Makefile View File

@@ -35,6 +35,7 @@ OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o

OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o



+ 314
- 0
libavfilter/af_volume.c View File

@@ -0,0 +1,314 @@
/*
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

/**
* @file
* audio volume filter
*/

#include "libavutil/audioconvert.h"
#include "libavutil/common.h"
#include "libavutil/eval.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#include "af_volume.h"

static const char *precision_str[] = {
"fixed", "float", "double"
};

#define OFFSET(x) offsetof(VolumeContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM

static const AVOption options[] = {
{ "volume", "Volume adjustment.",
OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A },
{ "precision", "Mathematical precision.",
OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" },
{ "fixed", "8-bit fixed-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A, "precision" },
{ "float", "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A, "precision" },
{ "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" },
{ NULL },
};

static const AVClass volume_class = {
.class_name = "volume filter",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};

static av_cold int init(AVFilterContext *ctx, const char *args)
{
VolumeContext *vol = ctx->priv;
int ret;

vol->class = &volume_class;
av_opt_set_defaults(vol);

if ((ret = av_set_options_string(vol, args, "=", ":")) < 0) {
av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
return ret;
}

if (vol->precision == PRECISION_FIXED) {
vol->volume_i = (int)(vol->volume * 256 + 0.5);
vol->volume = vol->volume_i / 256.0;
av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n",
vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10);
} else {
av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n",
vol->volume, 20.0*log(vol->volume)/M_LN10,
precision_str[vol->precision]);
}

av_opt_free(vol);
return ret;
}

static int query_formats(AVFilterContext *ctx)
{
VolumeContext *vol = ctx->priv;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[][7] = {
/* PRECISION_FIXED */
{
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_NONE
},
/* PRECISION_FLOAT */
{
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
},
/* PRECISION_DOUBLE */
{
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
}
};

layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ff_set_common_channel_layouts(ctx, layouts);

formats = ff_make_format_list(sample_fmts[vol->precision]);
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);

formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_samplerates(ctx, formats);

return 0;
}

static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
int nb_samples, int volume)
{
int i;
for (i = 0; i < nb_samples; i++)
dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
}

static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
int nb_samples, int volume)
{
int i;
for (i = 0; i < nb_samples; i++)
dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
}

static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
int nb_samples, int volume)
{
int i;
int16_t *smp_dst = (int16_t *)dst;
const int16_t *smp_src = (const int16_t *)src;
for (i = 0; i < nb_samples; i++)
smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
}

static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
int nb_samples, int volume)
{
int i;
int16_t *smp_dst = (int16_t *)dst;
const int16_t *smp_src = (const int16_t *)src;
for (i = 0; i < nb_samples; i++)
smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
}

static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
int nb_samples, int volume)
{
int i;
int32_t *smp_dst = (int32_t *)dst;
const int32_t *smp_src = (const int32_t *)src;
for (i = 0; i < nb_samples; i++)
smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
}



static void volume_init(VolumeContext *vol)
{
vol->samples_align = 1;

switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
case AV_SAMPLE_FMT_U8:
if (vol->volume_i < 0x1000000)
vol->scale_samples = scale_samples_u8_small;
else
vol->scale_samples = scale_samples_u8;
break;
case AV_SAMPLE_FMT_S16:
if (vol->volume_i < 0x10000)
vol->scale_samples = scale_samples_s16_small;
else
vol->scale_samples = scale_samples_s16;
break;
case AV_SAMPLE_FMT_S32:
vol->scale_samples = scale_samples_s32;
break;
case AV_SAMPLE_FMT_FLT:
avpriv_float_dsp_init(&vol->fdsp, 0);
vol->samples_align = 4;
break;
case AV_SAMPLE_FMT_DBL:
avpriv_float_dsp_init(&vol->fdsp, 0);
vol->samples_align = 8;
break;
}
}

static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
VolumeContext *vol = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];

vol->sample_fmt = inlink->format;
vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;

volume_init(vol);

return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
int nb_samples = buf->audio->nb_samples;
AVFilterBufferRef *out_buf;

if (vol->volume == 1.0 || vol->volume_i == 256)
return ff_filter_frame(outlink, buf);

/* do volume scaling in-place if input buffer is writable */
if (buf->perms & AV_PERM_WRITE) {
out_buf = buf;
} else {
out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
out_buf->pts = buf->pts;
}

if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
int p, plane_samples;

if (av_sample_fmt_is_planar(buf->format))
plane_samples = FFALIGN(nb_samples, vol->samples_align);
else
plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);

if (vol->precision == PRECISION_FIXED) {
for (p = 0; p < vol->planes; p++) {
vol->scale_samples(out_buf->extended_data[p],
buf->extended_data[p], plane_samples,
vol->volume_i);
}
} else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
for (p = 0; p < vol->planes; p++) {
vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
(const float *)buf->extended_data[p],
vol->volume, plane_samples);
}
} else {
for (p = 0; p < vol->planes; p++) {
vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
(const double *)buf->extended_data[p],
vol->volume, plane_samples);
}
}
}

if (buf != out_buf)
avfilter_unref_buffer(buf);

return ff_filter_frame(outlink, out_buf);
}

static const AVFilterPad avfilter_af_volume_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};

static const AVFilterPad avfilter_af_volume_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
{ NULL }
};

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 = avfilter_af_volume_inputs,
.outputs = avfilter_af_volume_outputs,
};

+ 53
- 0
libavfilter/af_volume.h View File

@@ -0,0 +1,53 @@
/*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

/**
* @file
* audio volume filter
*/

#ifndef AVFILTER_AF_VOLUME_H
#define AVFILTER_AF_VOLUME_H

#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"

enum PrecisionType {
PRECISION_FIXED = 0,
PRECISION_FLOAT,
PRECISION_DOUBLE,
};

typedef struct VolumeContext {
const AVClass *class;
AVFloatDSPContext fdsp;
enum PrecisionType precision;
double volume;
int volume_i;
int channels;
int planes;
enum AVSampleFormat sample_fmt;

void (*scale_samples)(uint8_t *dst, const uint8_t *src, int nb_samples,
int volume);
int samples_align;
} VolumeContext;

#endif /* AVFILTER_AF_VOLUME_H */

+ 1
- 0
libavfilter/allfilters.c View File

@@ -46,6 +46,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (CHANNELSPLIT,channelsplit,af);
REGISTER_FILTER (JOIN, join, af);
REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (VOLUME, volume, af);

REGISTER_FILTER (ANULLSRC, anullsrc, asrc);



+ 1
- 1
libavfilter/version.h View File

@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"

#define LIBAVFILTER_VERSION_MAJOR 3
#define LIBAVFILTER_VERSION_MINOR 2
#define LIBAVFILTER_VERSION_MINOR 3
#define LIBAVFILTER_VERSION_MICRO 0

#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \


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