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 @@ > 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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