ffmpeg | branch: master | Paul B Mahol <one...@gmail.com> | Tue Mar 6 19:47:29 2018 +0100| [8fb0e51bd198c996b8932735e8002f0952ef1d06] | committer: Paul B Mahol
avfilter: add drmeter audio filter Signed-off-by: Paul B Mahol <one...@gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8fb0e51bd198c996b8932735e8002f0952ef1d06 --- Changelog | 1 + doc/filters.texi | 15 +++ libavfilter/Makefile | 1 + libavfilter/af_drmeter.c | 233 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + libavfilter/version.h | 2 +- 6 files changed, 252 insertions(+), 1 deletion(-) diff --git a/Changelog b/Changelog index d8630731fa..c1b9df46bc 100644 --- a/Changelog +++ b/Changelog @@ -45,6 +45,7 @@ version <next>: - Moved nvidia codec headers into an external repository. They can be found at http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git - native SBC encoder and decoder +- drmeter audio filter version 3.4: diff --git a/doc/filters.texi b/doc/filters.texi index 7151d4c748..bd43a7ac6e 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2538,6 +2538,21 @@ Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping. @end table +@section drmeter +Measure audio dynamic range. + +DR values of 14 and higher is found in very dynamic material. DR of 8 to 13 +is found in transition material. And anything less that 8 have very poor dynamics +and is very compressed. + +The filter accepts the following options: + +@table @option +@item length +Set window length in seconds used to split audio into segments of equal length. +Default is 3 seconds. +@end table + @section dynaudnorm Dynamic Audio Normalizer. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 6a6083618d..fc16512e2c 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -87,6 +87,7 @@ OBJS-$(CONFIG_COMPENSATIONDELAY_FILTER) += af_compensationdelay.o OBJS-$(CONFIG_CROSSFEED_FILTER) += af_crossfeed.o OBJS-$(CONFIG_CRYSTALIZER_FILTER) += af_crystalizer.o OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o +OBJS-$(CONFIG_DRMETER_FILTER) += af_drmeter.o OBJS-$(CONFIG_DYNAUDNORM_FILTER) += af_dynaudnorm.o OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o diff --git a/libavfilter/af_drmeter.c b/libavfilter/af_drmeter.c new file mode 100644 index 0000000000..ecccb65186 --- /dev/null +++ b/libavfilter/af_drmeter.c @@ -0,0 +1,233 @@ +/* + * Copyright (c) 2018 Paul B Mahol + * + * 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 + */ + +#include <float.h> + +#include "libavutil/ffmath.h" +#include "libavutil/opt.h" +#include "audio.h" +#include "avfilter.h" +#include "internal.h" + +typedef struct ChannelStats { + uint64_t nb_samples; + uint64_t blknum; + float peak; + float sum; + uint32_t peaks[10001]; + uint32_t rms[10001]; +} ChannelStats; + +typedef struct DRMeterContext { + const AVClass *class; + ChannelStats *chstats; + int nb_channels; + uint64_t tc_samples; + double time_constant; +} DRMeterContext; + +#define OFFSET(x) offsetof(DRMeterContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption drmeter_options[] = { + { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=3}, .01, 10, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(drmeter); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT, + AV_SAMPLE_FMT_NONE + }; + int ret; + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_output(AVFilterLink *outlink) +{ + DRMeterContext *s = outlink->src->priv; + + s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels); + if (!s->chstats) + return AVERROR(ENOMEM); + s->nb_channels = outlink->channels; + s->tc_samples = s->time_constant * outlink->sample_rate + .5; + + return 0; +} + +static void finish_block(ChannelStats *p) +{ + int peak_bin, rms_bin; + float peak, rms; + + rms = sqrt(2 * p->sum / p->nb_samples); + peak = p->peak; + rms_bin = av_clip(rms * 10000, 0, 10000); + peak_bin = av_clip(peak * 10000, 0, 10000); + p->rms[rms_bin]++; + p->peaks[peak_bin]++; + + p->peak = 0; + p->sum = 0; + p->nb_samples = 0; + p->blknum++; +} + +static void update_stat(DRMeterContext *s, ChannelStats *p, float sample) +{ + if (p->nb_samples >= s->tc_samples) { + finish_block(p); + } + + p->peak = FFMAX(FFABS(sample), p->peak); + p->sum += sample * sample; + p->nb_samples++; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *buf) +{ + DRMeterContext *s = inlink->dst->priv; + const int channels = s->nb_channels; + int i, c; + + switch (inlink->format) { + case AV_SAMPLE_FMT_FLTP: + for (c = 0; c < channels; c++) { + ChannelStats *p = &s->chstats[c]; + const float *src = (const float *)buf->extended_data[c]; + + for (i = 0; i < buf->nb_samples; i++, src++) + update_stat(s, p, *src); + } + break; + case AV_SAMPLE_FMT_FLT: { + const float *src = (const float *)buf->extended_data[0]; + + for (i = 0; i < buf->nb_samples; i++) { + for (c = 0; c < channels; c++, src++) + update_stat(s, &s->chstats[c], *src); + }} + break; + } + + return ff_filter_frame(inlink->dst->outputs[0], buf); +} + +#define SQR(a) ((a)*(a)) + +static void print_stats(AVFilterContext *ctx) +{ + DRMeterContext *s = ctx->priv; + float dr = 0; + int ch; + + for (ch = 0; ch < s->nb_channels; ch++) { + ChannelStats *p = &s->chstats[ch]; + float chdr, secondpeak, rmssum = 0; + int i, j, first = 0; + + finish_block(p); + + for (i = 0; i <= 10000; i++) { + if (p->peaks[10000 - i]) { + if (first) + break; + first = 1; + } + } + + secondpeak = (10000 - i) / 10000.; + + for (i = 10000, j = 0; i >= 0 && j < 0.2 * p->blknum; i--) { + if (p->rms[i]) { + rmssum += SQR(i / 10000.) * p->rms[i]; + j += p->rms[i]; + } + } + + chdr = 20 * log10(secondpeak / sqrt(rmssum / (0.2 * p->blknum))); + dr += chdr; + av_log(ctx, AV_LOG_INFO, "Channel %d: DR: %.1f\n", ch + 1, chdr); + } + + av_log(ctx, AV_LOG_INFO, "Overall DR: %.1f\n", dr / s->nb_channels); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + DRMeterContext *s = ctx->priv; + + if (s->nb_channels) + print_stats(ctx); + av_freep(&s->chstats); +} + +static const AVFilterPad drmeter_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad drmeter_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +AVFilter ff_af_drmeter = { + .name = "drmeter", + .description = NULL_IF_CONFIG_SMALL("Measure audio dynamic range."), + .query_formats = query_formats, + .priv_size = sizeof(DRMeterContext), + .priv_class = &drmeter_class, + .uninit = uninit, + .inputs = drmeter_inputs, + .outputs = drmeter_outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 9adb1090b7..cc423af738 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -98,6 +98,7 @@ static void register_all(void) REGISTER_FILTER(CROSSFEED, crossfeed, af); REGISTER_FILTER(CRYSTALIZER, crystalizer, af); REGISTER_FILTER(DCSHIFT, dcshift, af); + REGISTER_FILTER(DRMETER, drmeter, af); REGISTER_FILTER(DYNAUDNORM, dynaudnorm, af); REGISTER_FILTER(EARWAX, earwax, af); REGISTER_FILTER(EBUR128, ebur128, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index ca096962bb..babb4187b4 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 7 -#define LIBAVFILTER_VERSION_MINOR 12 +#define LIBAVFILTER_VERSION_MINOR 13 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog
