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dumpwave-filter.patch
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dumpwave-filter.patch
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diff --git a/Changelog b/Changelog
index f135fa56b6..1eb337796b 100644
--- a/Changelog
+++ b/Changelog
@@ -13,6 +13,7 @@ version <next>:
- GIF parser
- vividas demuxer
- hymt decoder
+- dumpwave filter
version 4.1:
diff --git a/doc/filters.texi b/doc/filters.texi
index 98858c5356..66c2961fd8 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2912,6 +2912,29 @@ Set window length in seconds used to split audio into segments of equal length.
Default is 3 seconds.
@end table
+@section dumpwave
+Dump RMS envelope to a file.
+Convert samples to decibels and calculates RMS (Root-Mean-Square) audio power in 0 to 1.0 floats.
+
+@table @option
+@item w, width
+Number of data values.
+The default value is @var{1800}
+
+@item n, nb_samples
+Samples count per value per channel, default 128
+
+@item f, file
+Path to a file ``-'' is a shorthand
+for standard output.
+@end table
+
+For example, to generate RMS envelope for 44.1 kHz 6 seconds length audio
+with dimensions @var{1800x140}, samples count @code{44100*6/1800=147} and store it to @var{/tmp/out.csv}, you might use:
+@example
+dumpwave=w=1800:n=147:f=/tmp/out.csv
+@end example
+
@section dynaudnorm
Dynamic Audio Normalizer.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 6e2658186d..c40e61fcbe 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -99,6 +99,7 @@ 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-$(CONFIG_DUMPWAVE_FILTER) += af_dumpwave.o
OBJS-$(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_dumpwave.c b/libavfilter/af_dumpwave.c
new file mode 100644
index 0000000000..19917ed22f
--- /dev/null
+++ b/libavfilter/af_dumpwave.c
@@ -0,0 +1,234 @@
+/*
+ * Copyright (c) 2019 Dmytro Humeniuk
+ *
+ * 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
+ * waveform audio filter – dump RMS amplitude to a file
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "libavutil/parseutils.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct DumpWaveContext {
+ const AVClass *class; /**< class for AVOptions */
+ int width; /**< number of data values */
+ int idx; /**< index of current value */
+ char *filename; /**< output filename */
+ float *values; /**< scaling factors */
+ int64_t nb_samples; /**< samples per value per channel */
+ int64_t count; /**< current number of samples counted */
+ int64_t frame_size; /**< samples per value */
+ double sum; /**< sum of the squared samples per value */
+ double max_value; /**< keep track of max value */
+ FILE *dump_fp;
+} DumpWaveContext;
+
+#define OFFSET(x) offsetof(DumpWaveContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption dumpwave_options[] = {
+ { "w", "set number of data values", OFFSET(width), AV_OPT_TYPE_INT, {.i64 = 1800}, 1, INT_MAX, FLAGS },
+ { "width", "set number of data values", OFFSET(width), AV_OPT_TYPE_INT, {.i64 = 1800}, 1, INT_MAX, FLAGS },
+ { "n", "set number of samples per value per channel", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 128}, 1, INT64_MAX, FLAGS },
+ { "nb_samples", "set number of samples per value per channel", OFFSET(nb_samples), AV_OPT_TYPE_INT64, {.i64 = 128}, 1, INT64_MAX, FLAGS },
+ { "f", "set dump file", OFFSET(filename), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, FLAGS },
+ { "file", "set dump file", OFFSET(filename), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(dumpwave);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ DumpWaveContext *dumpwave = ctx->priv;
+
+ dumpwave->sum = dumpwave->idx = dumpwave->count = 0;
+
+ if (!dumpwave->filename) {
+ av_log(ctx, AV_LOG_ERROR, "No output file provided\n");
+ return AVERROR(EINVAL);
+ } else if (!strcmp("-", dumpwave->filename)) {
+ dumpwave->dump_fp = stdout;
+ } else {
+ dumpwave->dump_fp = fopen(dumpwave->filename, "w");
+ if (!dumpwave->dump_fp) {
+ int err = AVERROR(errno);
+ char buf[128];
+ av_strerror(err, buf, sizeof(buf));
+ av_log(ctx, AV_LOG_ERROR, "Could not open file %s: %s\n",
+ dumpwave->filename, buf);
+ return err;
+ }
+ }
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ DumpWaveContext *dumpwave = ctx->priv;
+ fclose(dumpwave->dump_fp);
+ av_freep(&dumpwave->values);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE
+ };
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ int ret;
+
+ if (!(formats = ff_make_format_list(sample_fmts)))
+ return AVERROR(ENOMEM);
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ return ff_set_common_formats(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ DumpWaveContext *dumpwave = inlink->dst->priv;
+ dumpwave->frame_size = dumpwave->nb_samples * inlink->channels;
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ DumpWaveContext *dumpwave = ctx->priv;
+ dumpwave->values = av_malloc(dumpwave->width * sizeof(float));
+ if (!dumpwave->values)
+ return AVERROR(ENOMEM);
+ memset(dumpwave->values, 0, dumpwave->width * sizeof(float));
+
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink) {
+ AVFilterContext *ctx = outlink->src;
+ DumpWaveContext *dumpwave = ctx->priv;
+
+ AVFilterLink *inlink = ctx->inputs[0];
+
+ int ret = ff_request_frame(inlink);
+
+ if (ret == AVERROR_EOF)
+ for (int i = 0; i < dumpwave->width; i++)
+ fprintf(dumpwave->dump_fp, (i == dumpwave->width - 1) ? "%f\n" : "%f,",
+ av_clipf(dumpwave->values[i] / dumpwave->max_value, 0, 1.0));
+ return ret;
+}
+
+/**
+ * Convert sample to dB and calculate root mean squared value
+ */
+static inline void RMS(DumpWaveContext *dumpwave, const float sample)
+{
+ double value = 0.;
+ if (sample != 0)
+ value = (20. * log10(fabs(sample)) + 60.) / 60.;
+
+ dumpwave->sum += value * value;
+
+ if (dumpwave->count++ == dumpwave->frame_size) {
+ value = av_clipd(sqrt(dumpwave->sum / dumpwave->frame_size), 0, 1.0);
+ dumpwave->max_value = FFMAX(value, dumpwave->max_value);
+ dumpwave->values[dumpwave->idx++] = value;
+ dumpwave->sum = dumpwave->count = 0;
+ }
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ DumpWaveContext *dumpwave = ctx->priv;
+ int channels = frame->channels;
+ int c, i;
+
+ if (dumpwave->idx < dumpwave->width)
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ for (c = 0; c < channels; c++) {
+ const float *src = (const float *)frame->extended_data[c];
+
+ for (i = 0; i < frame->nb_samples; i++, src++)
+ RMS(dumpwave, *src);
+ }
+ break;
+ case AV_SAMPLE_FMT_FLT: {
+ const float *src = (const float *)frame->extended_data[0];
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ for (c = 0; c < channels; c++, src++)
+ RMS(dumpwave, *src);
+ }
+
+ } break;
+ }
+ return ff_filter_frame(inlink->dst->outputs[0], frame);
+}
+
+static const AVFilterPad dumpwave_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input
+ },
+ { NULL }
+};
+
+static const AVFilterPad dumpwave_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .config_props = config_output
+ },
+ { NULL }
+};
+
+AVFilter ff_af_dumpwave = {
+ .name = "dumpwave",
+ .description = NULL_IF_CONFIG_SMALL("Dump RMS amplitude to a file"),
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .priv_size = sizeof(DumpWaveContext),
+ .inputs = dumpwave_inputs,
+ .outputs = dumpwave_outputs,
+ .priv_class = &dumpwave_class,
+};
\ No newline at end of file
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index a600069500..0661d70126 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -91,6 +91,7 @@ extern AVFilter ff_af_crossfeed;
extern AVFilter ff_af_crystalizer;
extern AVFilter ff_af_dcshift;
extern AVFilter ff_af_drmeter;
+extern AVFilter ff_af_dumpwave;
extern AVFilter ff_af_dynaudnorm;
extern AVFilter ff_af_earwax;
extern AVFilter ff_af_ebur128;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index c6b4d981c9..6af31400eb 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 46
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MINOR 47
+#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
diff --git a/tests/fate-run.sh b/tests/fate-run.sh
index aece90a01d..9b758b9d14 100755
--- a/tests/fate-run.sh
+++ b/tests/fate-run.sh
@@ -264,6 +264,15 @@ refcmp_metadata(){
-f null /dev/null | awk -v ref=${ref} -v fuzz=${fuzz} -f ${base}/refcmp-metadata.awk -
}
+refcmp_dumpwave(){
+ src=$1
+ size=$2
+ fuzz=${3:-1}
+ ffmpeg $FLAGS $DEC_OPTS -i ${src} \
+ -af "dumpwave=n=${size}:f=-" \
+ -f null /dev/null | tr ',.' '\n=' | awk -v ref=${ref} -v fuzz=${fuzz} -f ${base}/refcmp-metadata.awk -
+}
+
video_filter(){
filters=$1
shift
diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
index dcc8d56c88..dc75174aad 100644
--- a/tests/fate/filter-audio.mak
+++ b/tests/fate/filter-audio.mak
@@ -109,6 +109,18 @@ fate-filter-dcshift: tests/data/asynth-44100-2.wav
fate-filter-dcshift: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-dcshift: CMD = framecrc -i $(SRC) -frames:a 20 -af dcshift=shift=0.25:limitergain=0.05
+FATE_AFILTER-$(call FILTERDEMDEC, DUMPWAVE, WAV, PCM_S16LE) += fate-filter-dumpwave
+fate-filter-dumpwave: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
+fate-filter-dumpwave: CMD = refcmp_dumpwave $(SRC) 147 0.000010
+
+FATE_AFILTER-$(call FILTERDEMDEC, DUMPWAVE, TRUEHD, TRUEHD) += fate-filter-dumpwave-fltp
+fate-filter-dumpwave-fltp: SRC = $(TARGET_SAMPLES)/audiomatch/tones_afconvert_44100_stereo_aac_he2.m4a
+fate-filter-dumpwave-fltp: CMD = refcmp_dumpwave $(SRC) 52 0.000015
+
+FATE_AFILTER-$(call FILTERDEMDEC, DUMPWAVE, FLAC, FLAC) += fate-filter-dumpwave-24bit
+fate-filter-dumpwave-24bit: SRC = $(TARGET_SAMPLES)/lossless-audio/master_audio_2.0_24bit.wma
+fate-filter-dumpwave-24bit: CMD = refcmp_dumpwave $(SRC) 51 0.000012
+
FATE_AFILTER-$(call FILTERDEMDECENCMUX, EARWAX, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-earwax
fate-filter-earwax: tests/data/asynth-44100-2.wav
fate-filter-earwax: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
diff --git a/tests/ref/fate/filter-dumpwave b/tests/ref/fate/filter-dumpwave
new file mode 100644
index 0000000000..9ff02a6428
--- /dev/null
+++ b/tests/ref/fate/filter-dumpwave
@@ -0,0 +1,1800 @@
+0=824713
+0=836494
+0=825428
+0=824060
+0=835604
+0=826313
+0=823901
+0=833144
+0=828954
+0=824104
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