Hi,
>Can you provide some context for this? Please add docs as well.
filter converts samples to decibels and calculates RMS
(Root-Mean-Square) audio power scaled to desired values,
it supposed to generate something like SoundCloud does
http://plnkr.co/edit/WeppGcEPQiOyKDsA0C4Y
I added docs.
>Can't really test this because it segfaults for me: ./ffmpeg -f lavfi
>-i anoisesrc -af dumpwave -f null -
it expected some config - fixed
>Just set the sample format to something like AV_SAMPLE_FMT_DBL in
>query_format. There's no need for this big switch statement.
it handles samples in different way, you may look at the astats filter
2018-01-08 23:58 GMT+01:00 Kyle Swanson <k...@ylo.ph>:
> Hi,
>
> Can you provide some context for this? Please add docs as well.
> Can't really test this because it segfaults for me: ./ffmpeg -f lavfi
> -i anoisesrc -af dumpwave -f null -
>
> On Sun, Jan 7, 2018 at 4:36 PM, <dmitry.gumen...@gmail.com> wrote:
>>
>> From: Dmytro Humeniuk <dmitry.gumen...@gmail.com>
>>
>> Signed-off-by: Dmytro Humeniuk <dmitry.gumen...@gmail.com>
>> ---
>> Changelog | 1 +
>> libavfilter/Makefile | 1 +
>> libavfilter/af_dumpwave.c | 273
>> +++++++++++++++++++++++++++++++++++++++++
>> libavfilter/allfilters.c | 1 +
>> libavfilter/version.h | 4 +-
>> tests/fate/filter-audio.mak | 5 +
>> tests/ref/fate/filter-dumpwave | 1 +
>> 7 files changed, 284 insertions(+), 2 deletions(-)
>> create mode 100644 libavfilter/af_dumpwave.c
>> create mode 100644 tests/ref/fate/filter-dumpwave
>>
>> diff --git a/Changelog b/Changelog
>> index 61075b3392..40fd624449 100644
>> --- a/Changelog
>> +++ b/Changelog
>> @@ -38,6 +38,7 @@ version <next>:
>> - 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/libavfilter/Makefile b/libavfilter/Makefile
>> index 256dfabd66..020d90ee01 100644
>> --- a/libavfilter/Makefile
>> +++ b/libavfilter/Makefile
>> @@ -122,6 +122,7 @@ OBJS-$(CONFIG_TREMOLO_FILTER) +=
>> af_tremolo.o
>> OBJS-$(CONFIG_VIBRATO_FILTER) += af_vibrato.o
>> generate_wave_table.o
>> OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
>> OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
>> +OBJS-$(CONFIG_DUMPWAVE_FILTER) += af_dumpwave.o
>>
>> OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o
>> OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.o
>> diff --git a/libavfilter/af_dumpwave.c b/libavfilter/af_dumpwave.c
>> new file mode 100644
>> index 0000000000..493b5b7ff2
>> --- /dev/null
>> +++ b/libavfilter/af_dumpwave.c
>> @@ -0,0 +1,273 @@
>> +/*
>> + * 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 like SoundCloud
>> does
>> + */
>> +
>> +#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;
>> + int w, h;
>> + AVRational rate;
>> + int col;
>> + char *json;
>> + char *str;
>> + double *values;
>> + int64_t c, n, max_samples;
>> + double sum; /* sum of the squared samples per segment */
>> +} DumpWaveContext;
>> +
>> +#define OFFSET(x) offsetof(DumpWaveContext, x)
>> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
>> +
>> +static const AVOption dumpwave_options[] = {
>> + { "s", "set dump size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str =
>> "600x240"}, 0, 0, FLAGS },
>> + { "c", "set number of samples per item", OFFSET(c), AV_OPT_TYPE_INT64,
>> {.i64 = 0}, 0, INT64_MAX, FLAGS },
>
> Use more descriptive parameter names. Also, per item? What is an item?
>
>> + { "json", "set dump file", OFFSET(json), AV_OPT_TYPE_STRING, { .str =
>> NULL }, 0, 0, FLAGS },
>> + { NULL }
>> +};
>> +
>> +AVFILTER_DEFINE_CLASS(dumpwave);
>> +
>> +static int config_output(AVFilterLink *outlink)
>> +{
>> + DumpWaveContext *dumpwave = outlink->src->priv;
>> + const int ch_width = dumpwave->w;
>> + dumpwave->values = av_malloc(ch_width * sizeof(double));
>> + dumpwave->str = av_malloc(ch_width*sizeof(int));
>> + dumpwave->max_samples = dumpwave->c * outlink->channels;
>> +
>> + return 0;
>> +}
>> +
>> +static av_cold void uninit(AVFilterContext *ctx)
>> +{
>> + DumpWaveContext *dumpwave = ctx->priv;
>> + const int ch_height = dumpwave->h;
>> + const int ch_width = dumpwave->w;
>> + char *result = dumpwave->str;
>> + FILE *dump_fp = NULL;
>> +
>> + if (dumpwave->json && !(dump_fp = av_fopen_utf8(dumpwave->json, "w")))
>> + av_log(ctx, AV_LOG_WARNING, "dump failed.\n");
>> +
>> + if (dump_fp) {
>> + fprintf(dump_fp, "{\"width\":%d,\"height\":%d,\"samples\":[%s]}",
>> ch_width, ch_height, result);
>> + fclose(dump_fp);
>> + }
>> + av_freep(&dumpwave->str);
>> + av_freep(&dumpwave->values);
>> +}
>> +
>> +static int dumpwave_request_frame(AVFilterLink *outlink)
>> +{
>> + AVFilterContext *ctx = outlink->src;
>> + DumpWaveContext *dumpwave = ctx->priv;
>> + const int ch_height = dumpwave->h;
>> + const int ch_width = dumpwave->w;
>> + const double *values = dumpwave->values;
>> + char *result = dumpwave->str;
>> +
>> + AVFilterLink *inlink = ctx->inputs[0];
>> + int ret, val;
>> +
>> + ret = ff_request_frame(inlink);
>> + if (ret == AVERROR_EOF) {
>> +
>> + char *pos = result;
>> +
>> + for(int i = 0; i < ch_width; i++) {
>> + val = ch_height * values[i];
>> + if (val < 0 ) val = 0;
>> + pos += sprintf(pos, "%d,", val);
>> + }
>> + pos[-1] = '\0'; //removing trailing comma
>> + }
>> +
>> + return ret;
>> +}
>> +
>> +static inline void calculate(DumpWaveContext *dumpwave, double smpl)
>> +{
>> + double *sum = &dumpwave->sum;
>> + double *values = dumpwave->values;
>> + const int64_t max_samples = dumpwave->max_samples;
>> + int *col = &dumpwave->col;
>> + int64_t *n = &dumpwave->n;
>> +
>> + if (smpl != 0)
>> + smpl = (20 * log10(fabs(smpl)) + 60) / 60;
>> +
>> + *sum += smpl*smpl;
>> +
>> + if ((*n)++ == max_samples) {
>> + values[(*col)++] = sqrt(*sum / max_samples);
>> + *sum = *n = 0;
>> + }
>> +}
>> +
>> +static int dumpwave_filter_frame(AVFilterLink *inlink, AVFrame *buf)
>> +{
>> + AVFilterContext *ctx = inlink->dst;
>> + DumpWaveContext *dumpwave = ctx->priv;
>> + const int channels = inlink->channels;
>> +
>> + int i, c;
>> +
>> + switch (inlink->format) {
>> + case AV_SAMPLE_FMT_DBLP:
>> + for (c = 0; c < channels; c++) {
>> + const double *src = (const double *)buf->extended_data[c];
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++)
>> + calculate(dumpwave, *src);
>> + }
>> + break;
>> + case AV_SAMPLE_FMT_DBL: {
>> + const double *src = (const double *)buf->extended_data[0];
>> +
>> + for (i = 0; i < buf->nb_samples; i++) {
>> + for (c = 0; c < channels; c++, src++)
>> + calculate(dumpwave, *src);
>> + }}
>> + break;
>> + case AV_SAMPLE_FMT_FLTP:
>> + for (c = 0; c < channels; c++) {
>> + const float *src = (const float *)buf->extended_data[c];
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++)
>> + calculate(dumpwave, *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++)
>> + calculate(dumpwave, *src);
>> + }}
>> + break;
>> + case AV_SAMPLE_FMT_S64P:
>> + for (c = 0; c < channels; c++) {
>> + const int64_t *src = (const int64_t *)buf->extended_data[c];
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++)
>> + calculate(dumpwave, *src / (double)INT64_MAX);
>> + }
>> + break;
>> + case AV_SAMPLE_FMT_S64: {
>> + const int64_t *src = (const int64_t *)buf->extended_data[0];
>> +
>> + for (i = 0; i < buf->nb_samples; i++) {
>> + for (c = 0; c < channels; c++, src++)
>> + calculate(dumpwave, *src / (double)INT64_MAX);
>> + }}
>> + break;
>> + case AV_SAMPLE_FMT_S32P:
>> + for (c = 0; c < channels; c++) {
>> + const int32_t *src = (const int32_t *)buf->extended_data[c];
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++)
>> + calculate(dumpwave, *src / (double)INT32_MAX);
>> + }
>> + break;
>> + case AV_SAMPLE_FMT_S32: {
>> + const int32_t *src = (const int32_t *)buf->extended_data[0];
>> +
>> + for (i = 0; i < buf->nb_samples; i++) {
>> + for (c = 0; c < channels; c++, src++)
>> + calculate(dumpwave, *src / (double)INT32_MAX);
>> + }}
>> + break;
>> + case AV_SAMPLE_FMT_S16P:
>> + for (c = 0; c < channels; c++) {
>> + const int16_t *src = (const int16_t *)buf->extended_data[c];
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++)
>> + calculate(dumpwave, *src / (double)INT16_MAX);
>> + }
>> + break;
>> + case AV_SAMPLE_FMT_S16: {
>> + const int16_t *src = (const int16_t *)buf->extended_data[0];
>> +
>> + for (i = 0; i < buf->nb_samples; i++) {
>> + for (c = 0; c < channels; c++, src++)
>> + calculate(dumpwave, *src / (double)INT16_MAX);
>> + }}
>> + break;
>> + case AV_SAMPLE_FMT_U8P:
>> + for (c = 0; c < channels; c++) {
>> + const int8_t *src = (const int8_t *)buf->extended_data[c];
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++)
>> + calculate(dumpwave, *src / (double)INT8_MAX);
>> + }
>> + break;
>> + case AV_SAMPLE_FMT_U8: {
>> + const int8_t *src = (const int8_t *)buf->extended_data[0];
>> +
>> + for (i = 0; i < buf->nb_samples; i++) {
>> + for (c = 0; c < channels; c++, src++)
>> + calculate(dumpwave, *src / (double)INT8_MAX);
>> + }}
>> + break;
>> + }
>
> Just set the sample format to something like AV_SAMPLE_FMT_DBL in
> query_format. There's no need for this big switch statement.
>
>> + return ff_filter_frame(ctx->outputs[0], buf);
>> +}
>> +
>> +static const AVFilterPad dumpwave_inputs[] = {
>> + {
>> + .name = "default",
>> + .type = AVMEDIA_TYPE_AUDIO,
>> + .filter_frame = dumpwave_filter_frame,
>> + },
>> + { NULL }
>> +};
>> +
>> +static const AVFilterPad dumpwave_outputs[] = {
>> + {
>> + .name = "default",
>> + .type = AVMEDIA_TYPE_AUDIO,
>> + .request_frame = dumpwave_request_frame,
>> + .config_props = config_output
>> + },
>> + { NULL }
>> +};
>> +
>> +AVFilter ff_af_dumpwave = {
>> + .name = "dumpwave",
>> + .description = NULL_IF_CONFIG_SMALL("Dumps RMS audio peaks to a json
>> file"),
>> + .uninit = uninit,
>> + .priv_size = sizeof(DumpWaveContext),
>> + .inputs = dumpwave_inputs,
>> + .outputs = dumpwave_outputs,
>> + .priv_class = &dumpwave_class,
>> +};
>> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
>> index 753ae968aa..887c80c706 100644
>> --- a/libavfilter/allfilters.c
>> +++ b/libavfilter/allfilters.c
>> @@ -133,6 +133,7 @@ static void register_all(void)
>> REGISTER_FILTER(VIBRATO, vibrato, af);
>> REGISTER_FILTER(VOLUME, volume, af);
>> REGISTER_FILTER(VOLUMEDETECT, volumedetect, af);
>> + REGISTER_FILTER(DUMPWAVE, dumpwave, af);
>>
>> REGISTER_FILTER(AEVALSRC, aevalsrc, asrc);
>> REGISTER_FILTER(ANOISESRC, anoisesrc, asrc);
>> diff --git a/libavfilter/version.h b/libavfilter/version.h
>> index 0f11721822..ca096962bb 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 11
>> -#define LIBAVFILTER_VERSION_MICRO 101
>> +#define LIBAVFILTER_VERSION_MINOR 12
>> +#define LIBAVFILTER_VERSION_MICRO 100
>>
>> #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
>> LIBAVFILTER_VERSION_MINOR, \
>> diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
>> index bd8b3d3c35..2da644a66f 100644
>> --- a/tests/fate/filter-audio.mak
>> +++ b/tests/fate/filter-audio.mak
>> @@ -347,3 +347,8 @@ fate-filter-formats: CMD = run libavfilter/tests/formats
>> FATE_SAMPLES_AVCONV += $(FATE_AFILTER_SAMPLES-yes)
>> FATE_FFMPEG += $(FATE_AFILTER-yes)
>> fate-afilter: $(FATE_AFILTER-yes) $(FATE_AFILTER_SAMPLES-yes)
>> +
>> +FATE_AFILTER-$(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 = ffmpeg -i $(SRC) -af
>> dumpwave=s=1800x140:c=147:json=tests/data/fate/filter-dumpwave.out -f null -
>> && cat tests/data/fate/filter-dumpwave.out
>> +
>> diff --git a/tests/ref/fate/filter-dumpwave b/tests/ref/fate/filter-dumpwave
>> new file mode 100644
>> index 0000000000..bd07098ef8
>> --- /dev/null
>> +++ b/tests/ref/fate/filter-dumpwave
>> @@ -0,0 +1 @@
>> +{"width":1800,"height":140,"samples":[103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,102,104,103,102,104,104,102,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,102,104,103,102,104,104,102,103,104,102,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,102,104,104,102,103,104,102,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,102,103,104,102,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,102,104,103,103,103,104,103,103,104,103,103,104,103,103,104,103,103,103,104,103,103,104,103,103,104,103,102,104,103,102,104,103,102,103,104,102,103,104,103,101,104,105,104,103,102,104,104,102,102,103,104,104,103,103,104,104,102,103,103,103,104,103,103,103,104,102,104,103,103,104,103,103,103,103,103,103,103,103,103,104,103,103,103,104,103,103,103,103,104,103,103,103,103,103,103,103,104,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,104,104,103,103,103,103,103,103,103,103,103,103,103,102,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,104,103,103,105,103,103,103,103,103,103,104,103,103,103,104,103,103,103,104,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,106,106,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,100,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,96,98,100,97,99,98,97,97,97,96,98,97,98,97,99,98,97,95,98,97,98,101,96,95,96,94,97,99,99,95,98,96,98,98,98,98,98,98,99,98,96,97,99,97,98,99,99,99,96,98,97,98,98,99,99,97,100,95,98,97,99,94,98,96,99,98,97,99,98,97,96,98,95,96,97,100,99,96,99,97,97,97,99,98,97,96,97,97,99,99,100,95,99,98,95,96,99,97,99,99,95,98,96,97,96,99,96,97,98,96,97,95,97,99,99,96,99,96,98,98,96,96,97,96,99,98,97,98,100,98,100,96,98,98,99,97,99,99,99,97,99,97,99,99,98,96,100,97,95,112,119,121,121,118,122,122,120,121,123,121,120,118,121,120,121,123,124,122,120,124,121,122,121,120,121,122,122,120,119,118,122,121,122,121,120,123,120,121,122,121,121,119,120,119,120,121,121,120,122,120,122,123,122,124,122,120,122,121,121,119,122,123,123,122,120,121,119,123,121,125,119,121,119,120,121,121,121,123,122,120,122,123,120,120,123,121,119,122,120,122,123,121,123,121,121,123,120,120,121,123,123,120,122,122,119,120,122,122,120,122,122,120,123,120,121,120,121,121,123,120,121,119,122,117,124,122,122,119,120,122,121,121,123,121,120,121,122,120,121,123,121,119,123,120,123,125,121,120,121,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,104,103,104,103,103,104,103,104,103,104,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,104,103,104,102,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,104,103,103,103,104,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,104,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,103,104,103,103,103,103,103,103,103,102,81,75,74,74,75,76,78,78,79,81,81,82,84,84,84,86,86,86,87,88,88,88,90,88,89,90,89,89,90,89,89,89,89,88,88,89,87,87,87,85,85,85,83,83,82,81,80,79,78,76,75,75,73,75,80,76,79,75,74,74,76,77,77,79,81,80,82,84,83,84,86,86,86,88,87,87,89,89,88,89,90,89,89,90,89,89,90,89,88,89,88,87,87,87,86,85,86,84,83,83,81,80,80,78,77,76,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>> --
>> 2.14.3 (Apple Git-98)
>>
>> _______________________________________________
>> ffmpeg-devel mailing list
>> ffmpeg-devel@ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
> Thanks,
> Kyle
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
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