Re: [FFmpeg-devel] [PATCH v2] avfilter: add dumpwave filter.

2018-01-12 Thread Дмитрий Гуменюк 
Hi
One more thing regarding asetnsamples->metadata I think it would be hard to set 
up 2 chains of them for getting 2 different “resolutions" at the same time
> On 12 Jan 2018, at 08:36, Kyle Swanson  wrote:
> 
> Hi,
> 
> Make sure you go back and read our comments. There were several things you
> didn't address in your most recent patch.
> 
> Thanks,
> Kyle
> ___
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From d15553e52b9723651bb368264d86d0c5ec414f7a Mon Sep 17 00:00:00 2001
From: Dmytro Humeniuk 
Date: Fri, 12 Jan 2018 23:48:23 +0100
Subject: [PATCH] avfilter: add dumpwave filter.

Signed-off-by: Dmytro Humeniuk 
---
 Changelog|   1 +
 doc/filters.texi |  22 
 libavfilter/Makefile |   1 +
 libavfilter/af_dumpwave.c| 233 +++
 libavfilter/allfilters.c |   1 +
 libavfilter/version.h|   4 +-
 tests/fate/filter-audio.mak  |  13 ++
 tests/ref/fate/filter-dumpwave   |   1 +
 tests/ref/fate/filter-dumpwave-24bit |   1 +
 tests/ref/fate/filter-dumpwave-fltp  |   1 +
 10 files changed, 276 insertions(+), 2 deletions(-)
 create mode 100644 libavfilter/af_dumpwave.c
 create mode 100644 tests/ref/fate/filter-dumpwave
 create mode 100644 tests/ref/fate/filter-dumpwave-24bit
 create mode 100644 tests/ref/fate/filter-dumpwave-fltp

diff --git a/Changelog b/Changelog
index 61075b3392..40fd624449 100644
--- a/Changelog
+++ b/Changelog
@@ -38,6 +38,7 @@ version :
 - Removed the ffserver program
 - Removed the ffmenc and ffmdec muxer and demuxer
 - VideoToolbox HEVC encoder and hwaccel
+- dumpwave audio filter
 
 
 version 3.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index bd93e0ab84..bd49a8ec91 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2538,6 +2538,28 @@ 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 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
+@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 ef4729dd3f..2ffbc9497a 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-$(CONFIG_DUMPWAVE_FILTER)   += af_dumpwave.o
 OBJS-$(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 00..3d1653d8de
--- /dev/null
+++ b/libavfilter/af_dumpwave.c
@@ -0,0 +1,233 @@
+/*
+ * Copyright (c) 2017 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 file
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "libavutil/parseutils.h"
+#include "libavformat/avio.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

Re: [FFmpeg-devel] [PATCH v2] avfilter: add dumpwave filter.

2018-01-12 Thread Dmitry Gumenyuk 

> On 12 Jan 2018, at 08:36, Kyle Swanson  wrote:
> 
> Hi,
> 
> Make sure you go back and read our comments. There were several things you
> didn't address in your most recent patch.
Sorry, seems I missed few emails. Sure
> Thanks,
> Kyle
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Re: [FFmpeg-devel] [PATCH v2] avfilter: add dumpwave filter.

2018-01-11 Thread Kyle Swanson 
Hi,

Make sure you go back and read our comments. There were several things you
didn't address in your most recent patch.

Thanks,
Kyle
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Re: [FFmpeg-devel] [PATCH v2] avfilter: add dumpwave filter.

2018-01-11 Thread Michael Niedermayer 
On Thu, Jan 11, 2018 at 10:11:47PM +0100, Dmytro Humeniuk wrote:
[...]

> +static const AVOption dumpwave_options[] = {
> +{ "s", "set width and height", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str 
> = "1800x140"}, 0, 0, FLAGS },
> +{ "size", "set width and height", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, 
> {.str = "1800x140"}, 0, 0, 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 json dump file", OFFSET(json_filename), AV_OPT_TYPE_STRING, 
> { .str = NULL }, 0, 0, FLAGS },
> +{ "json", "set json dump file", OFFSET(json_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->i = dumpwave->n = 0;
> +
> +if (!dumpwave->json_filename) {
> +dumpwave->dump_fp = stdout;

stdout is not safe as a output for filters and a bad default
2 filters writing to stdout would interfere
it would interfere with a user application using stdout



> +} else {
> +dumpwave->dump_fp = fopen(dumpwave->json_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 json file %s: %s\n",
> +   dumpwave->json_filename, buf);
> +return err;
> +}
> +}
> +return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> +DumpWaveContext *dumpwave = ctx->priv;
> +fclose(dumpwave->dump_fp);
> +av_freep(>str);
> +av_freep(>values);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> +static const enum AVSampleFormat sample_fmts[] = {
> +AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8P,
> +AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
> +AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
> +AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64P,
> +AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
> +AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
> +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_output(AVFilterLink *outlink)
> +{
> +AVFilterContext *ctx = outlink->src;
> +DumpWaveContext *dumpwave = ctx->priv;
> +const int width = dumpwave->w;
> +dumpwave->values = av_realloc(NULL, width * sizeof(double));
> +dumpwave->str = av_realloc(NULL, width * sizeof(int));
> +dumpwave->max_samples = dumpwave->nb_samples * outlink->channels;
> +
> +return 0;
> +}
> +
> +static int dumpwave_request_frame(AVFilterLink *outlink)
> +{
> +AVFilterContext *ctx = outlink->src;
> +DumpWaveContext *dumpwave = ctx->priv;
> +const int width = dumpwave->w;
> +const int height = dumpwave->h;
> +char *p, *result = dumpwave->str;
> +
> +AVFilterLink *inlink = ctx->inputs[0];
> +int ret;
> +
> +ret = ff_request_frame(inlink);
> +
> +if (ret == AVERROR_EOF) {
> +p = result;
> +
> +for(int i = 0; i < width; i++)
> +p += sprintf(p, "%d,", av_clip(dumpwave->h * 
> dumpwave->values[i], 0, dumpwave->h));

sprintf is generally not safe as it does not check if there is enough space

[...]

-- 
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

No great genius has ever existed without some touch of madness. -- Aristotle


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[FFmpeg-devel] [PATCH v2] avfilter: add dumpwave filter.

2018-01-11 Thread Dmytro Humeniuk 
Signed-off-by: Dmytro Humeniuk 
---
 Changelog|   1 +
 doc/filters.texi |  23 +++
 libavfilter/Makefile |   1 +
 libavfilter/af_dumpwave.c| 324 +++
 libavfilter/allfilters.c |   1 +
 libavfilter/version.h|   4 +-
 tests/fate/filter-audio.mak  |  13 ++
 tests/ref/fate/filter-dumpwave   |   1 +
 tests/ref/fate/filter-dumpwave-24bit |   1 +
 tests/ref/fate/filter-dumpwave-fltp  |   1 +
 10 files changed, 368 insertions(+), 2 deletions(-)
 create mode 100644 libavfilter/af_dumpwave.c
 create mode 100644 tests/ref/fate/filter-dumpwave
 create mode 100644 tests/ref/fate/filter-dumpwave-24bit
 create mode 100644 tests/ref/fate/filter-dumpwave-fltp

diff --git a/Changelog b/Changelog
index 61075b3392..40fd624449 100644
--- a/Changelog
+++ b/Changelog
@@ -38,6 +38,7 @@ version :
 - Removed the ffserver program
 - Removed the ffmenc and ffmdec muxer and demuxer
 - VideoToolbox HEVC encoder and hwaccel
+- dumpwave audio filter
 
 
 version 3.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index bd93e0ab84..1f7e4f5380 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2538,6 +2538,29 @@ 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 dumpwave
+Dumps RMS amplitude to JSON file.
+Converts samples to decibels and calculates RMS (Root-Mean-Square) audio power 
scaled to desired values.
+
+@table @option
+@item s, size
+Dimensions @code{WxH}.
+@code{W} - number of data values in json, values will be scaled according to 
@code{H}.
+The default value is @var{1800x140}
+
+@item n, nb_samples
+Samples count per value per channel, default 128
+
+@item f, json
+Path to json file
+@end table
+
+For example, to generate RMS amplitude 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.json}, you might use:
+@example
+dumpwave=s=1800x140:n=147:json=/tmp/out.json
+@end example
+
 @section dynaudnorm
 Dynamic Audio Normalizer.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index ef4729dd3f..2ffbc9497a 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-$(CONFIG_DUMPWAVE_FILTER)   += af_dumpwave.o
 OBJS-$(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 00..486f3a0d79
--- /dev/null
+++ b/libavfilter/af_dumpwave.c
@@ -0,0 +1,324 @@
+/*
+ * Copyright (c) 2017 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 – dumps RMS amplitude to JSON 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 w;  /**< number of data values in json */
+int h;  /**< values will be scaled according to provided 
here */
+int i;  /**< index of current value */
+char *json_filename;/**< json filename */
+char *str;  /**< comma separated values */
+double *values; /**< scaling factors */
+int64_t nb_samples; /**< samples per value per channel */
+int64_t n;  /**< current number of samples counted */
+int64_t max_samples;/**< samples per value */
+double sum; /**< sum of the squared samples per value */
+FILE *dump_fp;
+} DumpWaveContext;
+
+#define