[FFmpeg-cvslog] avfilter: add drmeter audio filter
Paul B Mahol
git at videolan.org
Sun Mar 11 14:44:42 EET 2018
ffmpeg | branch: master | Paul B Mahol <onemda at 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 <onemda at 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
+ at 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:
+
+ at table @option
+ at item length
+Set window length in seconds used to split audio into segments of equal length.
+Default is 3 seconds.
+ at 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, \
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