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https://github.com/FFmpeg/FFmpeg.git
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avfilter: add drmeter audio filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
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@ -45,6 +45,7 @@ version <next>:
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- Moved nvidia codec headers into an external repository.
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They can be found at http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git
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- native SBC encoder and decoder
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- drmeter audio filter
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version 3.4:
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@ -2538,6 +2538,21 @@ Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
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used to prevent clipping.
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@end table
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@section drmeter
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Measure audio dynamic range.
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DR values of 14 and higher is found in very dynamic material. DR of 8 to 13
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is found in transition material. And anything less that 8 have very poor dynamics
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and is very compressed.
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The filter accepts the following options:
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@table @option
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@item length
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Set window length in seconds used to split audio into segments of equal length.
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Default is 3 seconds.
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@end table
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@section dynaudnorm
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Dynamic Audio Normalizer.
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@ -87,6 +87,7 @@ OBJS-$(CONFIG_COMPENSATIONDELAY_FILTER) += af_compensationdelay.o
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OBJS-$(CONFIG_CROSSFEED_FILTER) += af_crossfeed.o
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OBJS-$(CONFIG_CRYSTALIZER_FILTER) += af_crystalizer.o
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OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
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OBJS-$(CONFIG_DRMETER_FILTER) += af_drmeter.o
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OBJS-$(CONFIG_DYNAUDNORM_FILTER) += af_dynaudnorm.o
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OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
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OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
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233
libavfilter/af_drmeter.c
Normal file
233
libavfilter/af_drmeter.c
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@ -0,0 +1,233 @@
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/*
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* Copyright (c) 2018 Paul B Mahol
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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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#include <float.h>
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#include "libavutil/ffmath.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct ChannelStats {
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uint64_t nb_samples;
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uint64_t blknum;
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float peak;
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float sum;
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uint32_t peaks[10001];
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uint32_t rms[10001];
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} ChannelStats;
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typedef struct DRMeterContext {
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const AVClass *class;
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ChannelStats *chstats;
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int nb_channels;
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uint64_t tc_samples;
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double time_constant;
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} DRMeterContext;
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#define OFFSET(x) offsetof(DRMeterContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption drmeter_options[] = {
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{ "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=3}, .01, 10, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(drmeter);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_output(AVFilterLink *outlink)
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{
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DRMeterContext *s = outlink->src->priv;
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s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);
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if (!s->chstats)
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return AVERROR(ENOMEM);
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s->nb_channels = outlink->channels;
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s->tc_samples = s->time_constant * outlink->sample_rate + .5;
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return 0;
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}
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static void finish_block(ChannelStats *p)
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{
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int peak_bin, rms_bin;
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float peak, rms;
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rms = sqrt(2 * p->sum / p->nb_samples);
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peak = p->peak;
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rms_bin = av_clip(rms * 10000, 0, 10000);
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peak_bin = av_clip(peak * 10000, 0, 10000);
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p->rms[rms_bin]++;
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p->peaks[peak_bin]++;
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p->peak = 0;
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p->sum = 0;
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p->nb_samples = 0;
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p->blknum++;
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}
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static void update_stat(DRMeterContext *s, ChannelStats *p, float sample)
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{
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if (p->nb_samples >= s->tc_samples) {
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finish_block(p);
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}
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p->peak = FFMAX(FFABS(sample), p->peak);
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p->sum += sample * sample;
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p->nb_samples++;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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{
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DRMeterContext *s = inlink->dst->priv;
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const int channels = s->nb_channels;
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int i, c;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLTP:
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for (c = 0; c < channels; c++) {
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ChannelStats *p = &s->chstats[c];
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const float *src = (const float *)buf->extended_data[c];
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for (i = 0; i < buf->nb_samples; i++, src++)
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update_stat(s, p, *src);
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}
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break;
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case AV_SAMPLE_FMT_FLT: {
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const float *src = (const float *)buf->extended_data[0];
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for (i = 0; i < buf->nb_samples; i++) {
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for (c = 0; c < channels; c++, src++)
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update_stat(s, &s->chstats[c], *src);
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}}
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break;
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}
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return ff_filter_frame(inlink->dst->outputs[0], buf);
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}
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#define SQR(a) ((a)*(a))
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static void print_stats(AVFilterContext *ctx)
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{
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DRMeterContext *s = ctx->priv;
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float dr = 0;
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int ch;
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for (ch = 0; ch < s->nb_channels; ch++) {
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ChannelStats *p = &s->chstats[ch];
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float chdr, secondpeak, rmssum = 0;
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int i, j, first = 0;
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finish_block(p);
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for (i = 0; i <= 10000; i++) {
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if (p->peaks[10000 - i]) {
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if (first)
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break;
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first = 1;
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}
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}
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secondpeak = (10000 - i) / 10000.;
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for (i = 10000, j = 0; i >= 0 && j < 0.2 * p->blknum; i--) {
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if (p->rms[i]) {
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rmssum += SQR(i / 10000.) * p->rms[i];
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j += p->rms[i];
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}
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}
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chdr = 20 * log10(secondpeak / sqrt(rmssum / (0.2 * p->blknum)));
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dr += chdr;
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av_log(ctx, AV_LOG_INFO, "Channel %d: DR: %.1f\n", ch + 1, chdr);
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}
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av_log(ctx, AV_LOG_INFO, "Overall DR: %.1f\n", dr / s->nb_channels);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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DRMeterContext *s = ctx->priv;
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if (s->nb_channels)
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print_stats(ctx);
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av_freep(&s->chstats);
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}
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static const AVFilterPad drmeter_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad drmeter_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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{ NULL }
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};
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AVFilter ff_af_drmeter = {
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.name = "drmeter",
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.description = NULL_IF_CONFIG_SMALL("Measure audio dynamic range."),
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.query_formats = query_formats,
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.priv_size = sizeof(DRMeterContext),
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.priv_class = &drmeter_class,
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.uninit = uninit,
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.inputs = drmeter_inputs,
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.outputs = drmeter_outputs,
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};
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@ -98,6 +98,7 @@ static void register_all(void)
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REGISTER_FILTER(CROSSFEED, crossfeed, af);
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REGISTER_FILTER(CRYSTALIZER, crystalizer, af);
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REGISTER_FILTER(DCSHIFT, dcshift, af);
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REGISTER_FILTER(DRMETER, drmeter, af);
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REGISTER_FILTER(DYNAUDNORM, dynaudnorm, af);
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REGISTER_FILTER(EARWAX, earwax, af);
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REGISTER_FILTER(EBUR128, ebur128, af);
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 7
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#define LIBAVFILTER_VERSION_MINOR 12
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#define LIBAVFILTER_VERSION_MINOR 13
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#define LIBAVFILTER_VERSION_MICRO 100
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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