ffmpeg/libavfilter/af_volume.c
Michael Niedermayer 052f4f859c Merge commit 'a5e8c41c28f907d98d2a739db08f7aef4cbfcf3a'
* commit 'a5e8c41c28f907d98d2a739db08f7aef4cbfcf3a':
  lavfi: remove 'opaque' parameter from AVFilter.init()
  mov: do not try to read total disc/track number if data atom is too short.
  avconv: fix -force_key_frames
  dxva2_h264: fix signaling of mbaff frames
  x86: fft: elf64: fix PIC build

Conflicts:
	ffmpeg.c
	libavcodec/v210dec.h
	libavfilter/asrc_anullsrc.c
	libavfilter/buffersrc.c
	libavfilter/src_movie.c
	libavfilter/vf_drawtext.c
	libavfilter/vf_fade.c
	libavfilter/vf_overlay.c
	libavfilter/vsrc_color.c
	libavfilter/vsrc_testsrc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-26 23:57:07 +02:00

192 lines
5.7 KiB
C

/*
* 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 "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_INFO, "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 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;
}
}
}
ff_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 = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ|AV_PERM_WRITE},
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};