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FFmpeg/libavfilter/af_volume.c

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/*
* 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"
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#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 */
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d = pow(10, d/20);
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} 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;
}
}
}
Merge remote-tracking branch 'qatar/master' * qatar/master: (25 commits) rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC ape: Use unsigned integer maths arm: dsputil: fix overreads in put/avg_pixels functions h264: K&R formatting cosmetics for header files (part II/II) h264: K&R formatting cosmetics for header files (part I/II) rtmp: Implement check bandwidth notification. rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player. rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin. rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream. cmdutils: Add fallback case to switch in check_stream_specifier(). sctp: be consistent with socket option level configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags. vcr1enc: drop pointless empty encode_init() wrapper function vcr1: drop pointless write-only AVCodecContext member from VCR1Context vcr1: group encoder code together to save #ifdefs vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments mov: make one comment slightly more specific lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX lavfi: move audio-related functions to a separate file. lavfi: remove some audio-related function from public API. ... Conflicts: cmdutils.c libavcodec/h264.h libavcodec/h264_mvpred.h libavcodec/vcr1.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/defaults.c libavfilter/internal.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-10 23:41:29 +03:00
ff_filter_samples(outlink, insamples);
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}
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",
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.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ|AV_PERM_WRITE},
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};