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lavfi: drop af_volume_stefano.c in favor of af_volume_justin.c 

Justin's version has more features but is otherwise equivalent from the
point of view of the syntax.
tags/n1.1
Stefano Sabatini 13 years ago
parent
759e7a237f
commit
10db70d5e9
6 changed files with 16 additions and 263 deletions
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  1. +11
    -55
      doc/filters.texi
  2. +1
    -2
      libavfilter/Makefile
  3. +2
    -2
      libavfilter/af_volume.c
  4. +0
    -201
      libavfilter/af_volume_stefano.c
  5. +0
    -1
      libavfilter/allfilters.c
  6. +2
    -2
      libavfilter/version.h

+ 11
- 55
doc/filters.texi View File

@@ -829,56 +829,6 @@ out
Convert the audio sample format, sample rate and channel layout. This filter is
not meant to be used directly.

@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 decreased.

Output values are clipped to the maximum value.

If @var{vol} is expressed as a decimal number, 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

@section volumedetect

Detect the volume of the input video.
@@ -919,7 +869,7 @@ There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
In other words, raising the volume by +4 dB does not cause any clipping,
raising it by +5 dB causes clipping for 6 samples, etc.

@section volume_justin
@section volume

Adjust the input audio volume.

@@ -966,15 +916,21 @@ precision of the volume scaling.
@item
Halve the input audio volume:
@example
volume_justin=volume=0.5
volume_justin=volume=1/2
volume_justin=volume=-6.0206dB
volume=volume=0.5
volume=volume=1/2
volume=volume=-6.0206dB
@end example

In all the above example the named key for @option{volume} can be
omitted, for example like in:
@example
volume=0.5
@end example

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



+ 1
- 2
libavfilter/Makefile View File

@@ -71,8 +71,7 @@ OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume_stefano.o
OBJS-$(CONFIG_VOLUME_JUSTIN_FILTER) += af_volume_justin.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o

OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o


libavfilter/af_volume_justin.c → libavfilter/af_volume.c View File

@@ -299,8 +299,8 @@ static const AVFilterPad avfilter_af_volume_outputs[] = {
{ NULL }
};

AVFilter avfilter_af_volume_justin = {
.name = "volume_justin",
AVFilter avfilter_af_volume = {
.name = "volume",
.description = NULL_IF_CONFIG_SMALL("Change input volume."),
.query_formats = query_formats,
.priv_size = sizeof(VolumeContext),

+ 0
- 201
libavfilter/af_volume_stefano.c View File

@@ -1,201 +0,0 @@
/*
* 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/channel_layout.h"
#include "libavutil/eval.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"

typedef struct {
double volume;
int volume_i;
} VolumeContext;

static av_cold int init(AVFilterContext *ctx, const char *args)
{
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 */
d = pow(10, 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_VERBOSE, "volume=%f\n", vol->volume);
return 0;
}

static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts;
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
};

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

formats = ff_make_format_list(sample_fmts);
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 int filter_frame(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;
}
}
}
return ff_filter_frame(outlink, insamples);
}

static const AVFilterPad volume_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ | AV_PERM_WRITE,
},
{ NULL },
};

static const AVFilterPad volume_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ 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 = volume_inputs,
.outputs = volume_outputs,
};

+ 0
- 1
libavfilter/allfilters.c View File

@@ -64,7 +64,6 @@ void avfilter_register_all(void)
REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
REGISTER_FILTER (VOLUME, volume, af);
REGISTER_FILTER (VOLUME_JUSTIN, volume_justin, af);
REGISTER_FILTER (VOLUMEDETECT,volumedetect,af);

REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);


+ 2
- 2
libavfilter/version.h View File

@@ -29,8 +29,8 @@
#include "libavutil/avutil.h"

#define LIBAVFILTER_VERSION_MAJOR 3
#define LIBAVFILTER_VERSION_MINOR 25
#define LIBAVFILTER_VERSION_MICRO 102
#define LIBAVFILTER_VERSION_MINOR 26
#define LIBAVFILTER_VERSION_MICRO 100

#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \


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