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lavfi: add volume filter
This commit is contained in:
Stefano Sabatini
parent
1fc7077115
commit
618ac71354
@ -73,6 +73,8 @@ easier to use. The changes are:
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- Video Decoder Acceleration (VDA) HWAccel module.
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- replacement Indeo 3 decoder
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- new ffmpeg option: -map_channel
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- volume audio filter added
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version 0.8:
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@ -224,6 +224,56 @@ expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5}
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@var{c6} @var{c7}]"
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@end table
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@section volume
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Adjust the input audio volume.
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The filter accepts exactly one parameter @var{vol}, which expresses
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how the audio volume will be increased or decresed.
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Output values are clipped to the maximum value.
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If @var{vol} is expressed as a decimal number, and the output audio
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volume is given by the relation:
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@example
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@var{output_volume} = @var{vol} * @var{input_volume}
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@end example
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If @var{vol} is expressed as a decimal number followed by the string
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"dB", the value represents the requested change in decibels of the
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input audio power, and the output audio volume is given by the
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relation:
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@example
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@var{output_volume} = 10^(@var{vol}/20) * @var{input_volume}
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@end example
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Otherwise @var{vol} is considered an expression and its evaluated
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value is used for computing the output audio volume according to the
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first relation.
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Default value for @var{vol} is 1.0.
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@subsection Examples
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@itemize
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@item
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Half the input audio volume:
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@example
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volume=0.5
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@end example
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The above example is equivalent to:
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@example
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volume=1/2
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@end example
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@item
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Decrease input audio power by 12 decibels:
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@example
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volume=-12dB
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@end example
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@end itemize
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@c man end AUDIO FILTERS
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@chapter Audio Sources
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@ -28,6 +28,7 @@ OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
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OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
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OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
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OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
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OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
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OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
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191
libavfilter/af_volume.c
Normal file
191
libavfilter/af_volume.c
Normal file
@ -0,0 +1,191 @@
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/*
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* Copyright (c) 2011 Stefano Sabatini
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* audio volume filter
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* based on ffmpeg.c code
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/eval.h"
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#include "avfilter.h"
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typedef struct {
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double volume;
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int volume_i;
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} VolumeContext;
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static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
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{
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VolumeContext *vol = ctx->priv;
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char *tail;
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int ret = 0;
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vol->volume = 1.0;
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if (args) {
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/* parse the number as a decimal number */
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double d = strtod(args, &tail);
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if (*tail) {
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if (!strcmp(tail, "dB")) {
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/* consider the argument an adjustement in decibels */
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if (!strcmp(tail, "dB")) {
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d = exp10(d/20);
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}
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} else {
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/* parse the argument as an expression */
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ret = av_expr_parse_and_eval(&d, args, NULL, NULL,
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NULL, NULL, NULL, NULL,
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NULL, 0, ctx);
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}
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}
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if (ret < 0) {
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av_log(ctx, AV_LOG_ERROR,
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"Invalid volume argument '%s'\n", args);
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return AVERROR(EINVAL);
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}
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if (d < 0 || d > 65536) { /* 65536 = INT_MIN / (128 * 256) */
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av_log(ctx, AV_LOG_ERROR,
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"Negative or too big volume value %f\n", d);
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return AVERROR(EINVAL);
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}
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vol->volume = d;
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}
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vol->volume_i = (int)(vol->volume * 256 + 0.5);
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av_log(ctx, AV_LOG_INFO, "volume=%f\n", vol->volume);
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_U8,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE
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};
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int packing_fmts[] = { AVFILTER_PACKED, -1 };
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formats = avfilter_make_all_channel_layouts();
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if (!formats)
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return AVERROR(ENOMEM);
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avfilter_set_common_channel_layouts(ctx, formats);
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formats = avfilter_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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avfilter_set_common_sample_formats(ctx, formats);
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formats = avfilter_make_format_list(packing_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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avfilter_set_common_packing_formats(ctx, formats);
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return 0;
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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VolumeContext *vol = inlink->dst->priv;
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AVFilterLink *outlink = inlink->dst->outputs[0];
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const int nb_samples = insamples->audio->nb_samples *
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av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
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const double volume = vol->volume;
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const int volume_i = vol->volume_i;
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int i;
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if (volume_i != 256) {
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switch (insamples->format) {
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case AV_SAMPLE_FMT_U8:
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{
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uint8_t *p = (void *)insamples->data[0];
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for (i = 0; i < nb_samples; i++) {
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int v = (((*p - 128) * volume_i + 128) >> 8) + 128;
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*p++ = av_clip_uint8(v);
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}
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break;
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}
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case AV_SAMPLE_FMT_S16:
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{
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int16_t *p = (void *)insamples->data[0];
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for (i = 0; i < nb_samples; i++) {
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int v = ((int64_t)*p * volume_i + 128) >> 8;
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*p++ = av_clip_int16(v);
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}
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break;
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}
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case AV_SAMPLE_FMT_S32:
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{
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int32_t *p = (void *)insamples->data[0];
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for (i = 0; i < nb_samples; i++) {
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int64_t v = (((int64_t)*p * volume_i + 128) >> 8);
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*p++ = av_clipl_int32(v);
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}
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break;
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}
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case AV_SAMPLE_FMT_FLT:
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{
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float *p = (void *)insamples->data[0];
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float scale = (float)volume;
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for (i = 0; i < nb_samples; i++) {
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*p++ *= scale;
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}
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break;
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}
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case AV_SAMPLE_FMT_DBL:
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{
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double *p = (void *)insamples->data[0];
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for (i = 0; i < nb_samples; i++) {
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*p *= volume;
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p++;
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}
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break;
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}
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}
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}
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avfilter_filter_samples(outlink, insamples);
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}
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AVFilter avfilter_af_volume = {
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.name = "volume",
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.description = NULL_IF_CONFIG_SMALL("Change input volume."),
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.query_formats = query_formats,
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.priv_size = sizeof(VolumeContext),
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.init = init,
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.inputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ|AV_PERM_WRITE},
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{ .name = NULL}},
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.outputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO, },
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{ .name = NULL}},
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};
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@ -39,6 +39,7 @@ void avfilter_register_all(void)
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REGISTER_FILTER (ANULL, anull, af);
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REGISTER_FILTER (ARESAMPLE, aresample, af);
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REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
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REGISTER_FILTER (VOLUME, volume, af);
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REGISTER_FILTER (ABUFFER, abuffer, asrc);
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REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);
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@ -29,8 +29,8 @@
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#include "libavutil/rational.h"
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#define LIBAVFILTER_VERSION_MAJOR 2
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#define LIBAVFILTER_VERSION_MINOR 45
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#define LIBAVFILTER_VERSION_MICRO 3
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#define LIBAVFILTER_VERSION_MINOR 46
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#define LIBAVFILTER_VERSION_MICRO 0
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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LIBAVFILTER_VERSION_MINOR, \
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