[FFmpeg-devel] [PATCH 2/2] lavfi: add audio silencedetect filter.
Nicolas George
nicolas.george at normalesup.org
Thu Jan 5 20:29:35 CET 2012
Le sextidi 16 nivôse, an CCXX, Clément Bœsch a écrit :
> ---
> Changelog | 1 +
> doc/filters.texi | 23 ++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_silencedetect.c | 161 ++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> libavfilter/avfilter.h | 4 +-
> 6 files changed, 189 insertions(+), 2 deletions(-)
> create mode 100644 libavfilter/af_silencedetect.c
>
> diff --git a/Changelog b/Changelog
> index 8b59ac8..80da97b 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -19,6 +19,7 @@ version next:
> - Avid 1:1 10-bit RGB Packer decoder
> - v308 Quicktime Uncompressed 4:4:4 encoder and decoder
> - yuv4 libquicktime packed 4:2:0 encoder and decoder
> +- silencedetect audio filter
>
>
> version 0.9:
> diff --git a/doc/filters.texi b/doc/filters.texi
> index de73e3f..5349fd8 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -358,6 +358,29 @@ Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
> that should be preferred (see "-ac" option) unless you have very specific
> needs.
>
> + at section silencedetect
> +
> +Detect silence in an audio stream.
And do what with it? A few more words may be nice.
> +
> + at table @option
> + at item d, duration
> +Set silence duration until notification (default is 2 seconds).
> +
> + at item noise, n
> +Set noise tolerance. Can be specified in dB or amplitude ratio. Default is
> +-60dB, or 0.001.
> + at end table
> +
> +Detect 5 seconds of silence with -50dB noise tolerance:
> + at example
> +silencedetect=n=-50dB:d=5
> + at end example
> +
> +Detect silence with 0.0001 noise tolerance:
> + at example
> +silencedetect=noise=0.0001
> + at end example
An example with a complete filtergraph may be nice, especially if the use
requires something more complex than just inserting the filter.
> +
> @section volume
>
> Adjust the input audio volume.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 0d8f120..bb5748b 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -35,6 +35,7 @@ OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
> OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
> OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
> OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
> +OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
>
> OBJS-$(CONFIG_ABUFFER_FILTER) += asrc_abuffer.o
> OBJS-$(CONFIG_AEVALSRC_FILTER) += asrc_aevalsrc.o
> diff --git a/libavfilter/af_silencedetect.c b/libavfilter/af_silencedetect.c
> new file mode 100644
> index 0000000..8d68b63
> --- /dev/null
> +++ b/libavfilter/af_silencedetect.c
> @@ -0,0 +1,161 @@
> +/*
> + * This file is part of FFmpeg.
> + *
The copyright line is missing.
> + * 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
> + * Audio silence detector
> + */
> +
> +#include "libavutil/opt.h"
> +#include "avfilter.h"
> +
> +typedef struct {
> + const AVClass *class;
> + char *noise_str;
Too bad we can not let the options system parse the option instead of
copying it.
> + double noise;
> + int duration;
> + int nb_null_samples;
int64_t? If the silence if very long, it could overflow.
> + int silence;
> +} SilenceDetectContext;
> +
> +#define OFFSET(x) offsetof(SilenceDetectContext, x)
> +static const AVOption silencedetect_options[] = {
> + { "n", "set noise tolerance", OFFSET(noise_str), AV_OPT_TYPE_STRING, {.str="-60dB"}, CHAR_MIN, CHAR_MAX },
> + { "noise", "set noise tolerance", OFFSET(noise_str), AV_OPT_TYPE_STRING, {.str="-60dB"}, CHAR_MIN, CHAR_MAX },
> + { "d", "set minimum duration in seconds", OFFSET(duration), AV_OPT_TYPE_INT, {.dbl=2}, 0, INT_MAX},
> + { "duration", "set minimum duration in seconds", OFFSET(duration), AV_OPT_TYPE_INT, {.dbl=2}, 0, INT_MAX},
> + { NULL },
> +};
> +
> +static const char *silencedetect_get_name(void *ctx)
> +{
> + return "silencedetect";
> +}
> +
> +static const AVClass silencedetect_class = {
> + .class_name = "SilenceDetectContext",
> + .item_name = silencedetect_get_name,
> + .option = silencedetect_options,
> +};
> +
> +static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
> +{
> + int ret;
> + char *tail;
> + SilenceDetectContext *silence = ctx->priv;
> +
> + silence->class = &silencedetect_class;
> + av_opt_set_defaults(silence);
> +
> + if ((ret = av_set_options_string(silence, args, "=", ":")) < 0) {
> + av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
> + return ret;
> + }
> +
> + silence->noise = strtod(silence->noise_str, &tail);
> + if (strcmp(tail, "dB") == 0)
Some people do not like == 0 around here.
> + silence->noise = pow(10, silence->noise/20);
> +
> + return 0;
> +}
> +
> +static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
> +{
> + int i;
> + SilenceDetectContext *silence = inlink->dst->priv;
> + const int nb_samples = insamples->audio->nb_samples *
> + av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
> +
> + // FIXME: sample rate change will break this
> + int nbr_samples_notify = inlink->sample_rate * silence->duration;
It seems your code below counts samples per channel, but nbr_samples_notify
is computed without taking the number of channels into account.
Oh, and nbr looks like a French abbreviation, not an English one.
> +
> + switch (insamples->format) {
> + case AV_SAMPLE_FMT_DBL: {
> + double *p = (double *)insamples->data[0];
> + for (i = 0; i < nb_samples; i++, p++) {
> + if (*p < silence->noise && *p > -silence->noise) {
> + if (!silence->silence) {
> + silence->nb_null_samples++;
> + if (silence->nb_null_samples == nbr_samples_notify
> + && !silence->silence) {
!silence->silence seems duplicated. And possibly even useless: since you
check on silence->nb_null_samples == nbr_samples_notify and not >=, and you
increment each time, it can only be true once.
> + av_log(silence, AV_LOG_INFO, "Silence detected around %f sec\n",
> + insamples->pts * av_q2d(inlink->time_base));
I believe insamples->pts can be NOPTS?
> + silence->silence = 1;
> + }
> + }
> + } else {
> + if (silence->silence && silence->nb_null_samples)
Here again, the test could be made simpler: if silence->nb_null_samples >=
nbr_samples_notify.
> + av_log(silence, AV_LOG_INFO, "Silence ended (%f sec)\n",
> + insamples->pts * av_q2d(inlink->time_base));
> + silence->nb_null_samples = silence->silence = 0;
> + }
> + }
> + break;
> + }
> + }
> +
> + avfilter_filter_samples(inlink->dst->outputs[0], insamples);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats = NULL;
> + enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_DBL,
> + AV_SAMPLE_FMT_NONE
> + };
> + int packing_fmts[] = { AVFILTER_PACKED, -1 };
> +
> + formats = avfilter_make_all_channel_layouts();
> + if (!formats)
> + return AVERROR(ENOMEM);
> + avfilter_set_common_channel_layouts(ctx, formats);
> +
> + formats = avfilter_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + avfilter_set_common_sample_formats(ctx, formats);
> +
> + formats = avfilter_make_format_list(packing_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + avfilter_set_common_packing_formats(ctx, formats);
> +
> + return 0;
> +}
> +
> +AVFilter avfilter_af_silencedetect = {
> + .name = "silencedetect",
> + .description = NULL_IF_CONFIG_SMALL("Detect silence."),
> + .priv_size = sizeof(SilenceDetectContext),
> + .init = init,
> + .query_formats = query_formats,
> +
> + .inputs = (const AVFilterPad[]) {
> + { .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .get_audio_buffer = avfilter_null_get_audio_buffer,
> + .filter_samples = filter_samples, },
> + { .name = NULL}
Is it on purpose that most filters lack the space between the NULL and the
closing brace?
> + },
> + .outputs = (const AVFilterPad[]) {
> + { .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO, },
> + { .name = NULL}
> + },
Did you consider making it a sink rather than a filter?
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 621568e..a863a93 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -45,6 +45,7 @@ void avfilter_register_all(void)
> REGISTER_FILTER (EARWAX, earwax, af);
> REGISTER_FILTER (PAN, pan, af);
> REGISTER_FILTER (VOLUME, volume, af);
> + REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
>
> REGISTER_FILTER (ABUFFER, abuffer, asrc);
> REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);
> diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
> index d88d3ab..9c79597 100644
> --- a/libavfilter/avfilter.h
> +++ b/libavfilter/avfilter.h
> @@ -30,8 +30,8 @@
> #include "libavcodec/avcodec.h"
>
> #define LIBAVFILTER_VERSION_MAJOR 2
> -#define LIBAVFILTER_VERSION_MINOR 57
> -#define LIBAVFILTER_VERSION_MICRO 101
> +#define LIBAVFILTER_VERSION_MINOR 58
> +#define LIBAVFILTER_VERSION_MICRO 100
>
> #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
> LIBAVFILTER_VERSION_MINOR, \
Regards,
--
Nicolas George
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