[FFmpeg-devel] [PATCH] add silenceremove filter
Paul B Mahol
onemda at gmail.com
Mon Sep 1 12:34:34 CEST 2014
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 69 ++++++
libavfilter/Makefile | 1 +
libavfilter/af_silenceremove.c | 480 +++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 551 insertions(+)
create mode 100644 libavfilter/af_silenceremove.c
diff --git a/doc/filters.texi b/doc/filters.texi
index cca15fc..0c6de18 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1881,6 +1881,75 @@ ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
@end example
@end itemize
+ at section silenceremove
+
+Remove silence from the beginning, middle or end of the audio.
+
+The filter accepts the following options:
+
+ at table @option
+ at item start_periods
+This value is used to indicate if audio should be trimmed at beginning of
+the audio. A value of zero indicates no silence should be trimmed from the
+beginning. When specifying a non-zero value, it trims audio up until it
+finds non-silence. Normally, when trimming silence from beginning of audio
+the @var{start_periods} will be @code{1} but it can be increased to higher
+values to trim all audio up to specific count of non-silence periods.
+Default value is @code{0}.
+
+ at item start_duration
+Specify the amount of time that non-silence must be detected before it stops
+trimming audio. By increasing the duration, bursts of noises can be treated
+as silence and trimmed off. Default value is @code{0}.
+
+ at item start_threshold
+This indicates what sample value should be treated as silence. For digital
+audio, a value of @code{0} may be fine but for audio recorded from analog,
+you may wish to increase the value to account for background noise.
+Can be specified in dB (in case "dB" is appended to the specified value)
+or amplitude ratio. Default value is @code{0}.
+
+ at item stop_periods
+Set the count for trimming silence from the end of audio.
+To remove silence from the middle of a file, specify a @var{stop_periods}
+that is negative. This value is then threated as a positive value and is
+used to indicate the effect should restart processing as specified by
+ at var{start_periods}, making it suitable for removing periods of silence
+in the middle of the audio.
+Default value is @code{0}.
+
+ at item stop_duration
+Specify a duration of silence that must exist before audio is not copied any
+more. By specifying a higher duration, silence that is wanted can be left in
+the audio.
+Default value is @code{0}.
+
+ at item stop_threshold
+This is the same as @option{start_threshold} but for trimming silence from
+the end of audio.
+Can be specified in dB (in case "dB" is appended to the specified value)
+or amplitude ratio. Default value is @code{0}.
+
+ at item leave_silence
+This indicate that @var{stop_duration} length of audio should be left intact
+at the beginning of each period of silence.
+For example, if you want to remove long pauses between words but do not want
+to remove the pauses completely. Default value is @code{0}.
+
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+The following example shows how this filter can be used to start a recording
+that does not contain the delay at the start which usually occurs between
+pressing the record button and the start of the performance:
+ at example
+silenceremove=1:5:0.02
+ at end example
+ at end itemize
+
@section treble
Boost or cut treble (upper) frequencies of the audio using a two-pole
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index ce71ce1..3241b76 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -78,6 +78,7 @@ OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
+OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
diff --git a/libavfilter/af_silenceremove.c b/libavfilter/af_silenceremove.c
new file mode 100644
index 0000000..1e73ad0
--- /dev/null
+++ b/libavfilter/af_silenceremove.c
@@ -0,0 +1,480 @@
+/*
+ * Copyright (c) 2001 Heikki Leinonen
+ * Copyright (c) 2001 Chris Bagwell
+ * Copyright (c) 2003 Donnie Smith
+ * Copyright (c) 2014 Paul B Mahol
+ *
+ * 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
+ */
+
+#include <float.h> /* DBL_MAX */
+
+#include "libavutil/opt.h"
+#include "libavutil/timestamp.h"
+#include "audio.h"
+#include "formats.h"
+#include "avfilter.h"
+#include "internal.h"
+
+enum SilenceMode {
+ SILENCE_TRIM,
+ SILENCE_TRIM_FLUSH,
+ SILENCE_COPY,
+ SILENCE_COPY_FLUSH,
+ SILENCE_STOP
+};
+
+typedef struct SilenceRemoveContext {
+ const AVClass *class;
+
+ enum SilenceMode mode;
+
+ int start_periods;
+ int64_t start_duration;
+ double start_threshold;
+
+ int stop_periods;
+ int64_t stop_duration;
+ double stop_threshold;
+
+ double *start_holdoff;
+ size_t start_holdoff_offset;
+ size_t start_holdoff_end;
+ int start_found_periods;
+
+ double *stop_holdoff;
+ size_t stop_holdoff_offset;
+ size_t stop_holdoff_end;
+ int stop_found_periods;
+
+ double *window;
+ double *window_current;
+ double *window_end;
+ int window_size;
+ double rms_sum;
+
+ int leave_silence;
+ int restart;
+ int64_t next_pts;
+} SilenceRemoveContext;
+
+#define OFFSET(x) offsetof(SilenceRemoveContext, x)
+#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption silenceremove_options[] = {
+ { "start_periods", NULL, OFFSET(start_periods), AV_OPT_TYPE_INT, {.i64=0}, 0, 9000, FLAGS },
+ { "start_duration", NULL, OFFSET(start_duration), AV_OPT_TYPE_DURATION, {.i64=0}, 0, 9000, FLAGS },
+ { "start_threshold", NULL, OFFSET(start_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, FLAGS },
+ { "stop_periods", NULL, OFFSET(stop_periods), AV_OPT_TYPE_INT, {.i64=0}, -9000, 9000, FLAGS },
+ { "stop_duration", NULL, OFFSET(stop_duration), AV_OPT_TYPE_DURATION, {.i64=0}, 0, 9000, FLAGS },
+ { "stop_threshold", NULL, OFFSET(stop_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, FLAGS },
+ { "leave_silence", NULL, OFFSET(leave_silence), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(silenceremove);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ SilenceRemoveContext *s = ctx->priv;
+
+ if (s->stop_periods < 0) {
+ s->stop_periods = -s->stop_periods;
+ s->restart = 1;
+ }
+
+ return 0;
+}
+
+static void clear_rms(SilenceRemoveContext *s)
+{
+ memset(s->window, 0, s->window_size * sizeof(*s->window));
+
+ s->window_current = s->window;
+ s->window_end = s->window + s->window_size;
+ s->rms_sum = 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SilenceRemoveContext *s = ctx->priv;
+
+ s->window_size = (inlink->sample_rate / 50) * inlink->channels;
+ s->window = av_malloc_array(s->window_size, sizeof(*s->window));
+ if (!s->window)
+ return AVERROR(ENOMEM);
+
+ clear_rms(s);
+
+ s->start_duration = av_rescale(s->start_duration, inlink->sample_rate,
+ AV_TIME_BASE);
+ s->stop_duration = av_rescale(s->stop_duration, inlink->sample_rate,
+ AV_TIME_BASE);
+
+ if (s->start_duration) {
+ s->start_holdoff = av_malloc_array(s->start_duration,
+ sizeof(*s->start_holdoff) *
+ inlink->channels);
+ if (!s->start_holdoff)
+ return AVERROR(ENOMEM);
+ }
+
+ s->start_holdoff_offset = 0;
+ s->start_holdoff_end = 0;
+ s->start_found_periods = 0;
+
+ if (s->stop_duration) {
+ s->stop_holdoff = av_malloc_array(s->stop_duration,
+ sizeof(*s->stop_holdoff) *
+ inlink->channels);
+ if (!s->stop_holdoff)
+ return AVERROR(ENOMEM);
+ }
+
+ s->stop_holdoff_offset = 0;
+ s->stop_holdoff_end = 0;
+ s->stop_found_periods = 0;
+
+ if (s->start_periods)
+ s->mode = SILENCE_TRIM;
+ else
+ s->mode = SILENCE_COPY;
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
+
+ return 0;
+}
+
+static double compute_rms(SilenceRemoveContext *s, double sample)
+{
+ double new_sum;
+
+ new_sum = s->rms_sum;
+ new_sum -= *s->window_current;
+ new_sum += sample * sample;
+
+ return sqrt(new_sum / s->window_size);
+}
+
+static void update_rms(SilenceRemoveContext *s, double sample)
+{
+ s->rms_sum -= *s->window_current;
+ *s->window_current = sample * sample;
+ s->rms_sum += *s->window_current;
+
+ s->window_current++;
+ if (s->window_current >= s->window_end)
+ s->window_current = s->window;
+}
+
+static void flush(AVFrame *out, AVFilterLink *outlink,
+ int *nb_samples_written, int *ret)
+{
+ if (*nb_samples_written) {
+ out->nb_samples = *nb_samples_written / outlink->channels;
+ *ret = ff_filter_frame(outlink, out);
+ *nb_samples_written = 0;
+ } else {
+ av_frame_free(&out);
+ }
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ SilenceRemoveContext *s = ctx->priv;
+ int i, j, threshold, ret = 0;
+ int nbs, nb_samples_read, nb_samples_written;
+ double *obuf, *ibuf = (double *)in->data[0];
+ AVFrame *out;
+
+ nb_samples_read = nb_samples_written = 0;
+
+ switch (s->mode) {
+ case SILENCE_TRIM:
+silence_trim:
+ nbs = in->nb_samples - nb_samples_read / inlink->channels;
+
+ for (i = 0; i < nbs; i++) {
+ threshold = 0;
+ for (j = 0; j < inlink->channels; j++) {
+ threshold |= compute_rms(s, ibuf[j]) > s->start_threshold;
+ }
+
+ if (threshold) {
+ for (j = 0; j < inlink->channels; j++) {
+ update_rms(s, *ibuf);
+ s->start_holdoff[s->start_holdoff_end++] = *ibuf++;
+ nb_samples_read++;
+ }
+
+ if (s->start_holdoff_end >= s->start_duration * inlink->channels) {
+ if (++s->start_found_periods >= s->start_periods) {
+ s->mode = SILENCE_TRIM_FLUSH;
+ goto silence_trim_flush;
+ }
+
+ s->start_holdoff_offset = 0;
+ s->start_holdoff_end = 0;
+ }
+ } else {
+ s->start_holdoff_end = 0;
+
+ for (j = 0; j < inlink->channels; j++)
+ update_rms(s, ibuf[j]);
+
+ ibuf += inlink->channels;
+ nb_samples_read += inlink->channels;
+ }
+ }
+ break;
+
+ case SILENCE_TRIM_FLUSH:
+silence_trim_flush:
+ nbs = s->start_holdoff_end - s->start_holdoff_offset;
+ nbs -= nbs % inlink->channels;
+
+ out = ff_get_audio_buffer(inlink, nbs / inlink->channels);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+
+ memcpy(out->data[0], &s->start_holdoff[s->start_holdoff_offset],
+ nbs * sizeof(double));
+ s->start_holdoff_offset += nbs;
+
+ ret = ff_filter_frame(outlink, out);
+
+ if (s->start_holdoff_offset == s->start_holdoff_end) {
+ s->start_holdoff_offset = 0;
+ s->start_holdoff_end = 0;
+ s->mode = SILENCE_COPY;
+ goto silence_copy;
+ }
+ break;
+
+ case SILENCE_COPY:
+silence_copy:
+ nbs = in->nb_samples - nb_samples_read / inlink->channels;
+ if (!nbs)
+ break;
+
+ out = ff_get_audio_buffer(inlink, nbs);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ obuf = (double *)out->data[0];
+
+ if (s->stop_periods) {
+ for (i = 0; i < nbs; i++) {
+ threshold = 1;
+ for (j = 0; j < inlink->channels; j++)
+ threshold &= compute_rms(s, ibuf[j]) > s->stop_threshold;
+
+ if (threshold && s->stop_holdoff_end && !s->leave_silence) {
+ s->mode = SILENCE_COPY_FLUSH;
+ flush(out, outlink, &nb_samples_written, &ret);
+ goto silence_copy_flush;
+ } else if (threshold) {
+ for (j = 0; j < inlink->channels; j++) {
+ update_rms(s, *ibuf);
+ *obuf++ = *ibuf++;
+ nb_samples_read++;
+ nb_samples_written++;
+ }
+ } else if (!threshold) {
+ for (j = 0; j < inlink->channels; j++) {
+ update_rms(s, *ibuf);
+ if (s->leave_silence) {
+ *obuf++ = *ibuf;
+ nb_samples_written++;
+ }
+
+ s->stop_holdoff[s->stop_holdoff_end++] = *ibuf++;
+ nb_samples_read++;
+ }
+
+ if (s->stop_holdoff_end >= s->stop_duration * inlink->channels) {
+ if (++s->stop_found_periods >= s->stop_periods) {
+ s->stop_holdoff_offset = 0;
+ s->stop_holdoff_end = 0;
+
+ if (!s->restart) {
+ s->mode = SILENCE_STOP;
+ flush(out, outlink, &nb_samples_written, &ret);
+ goto silence_stop;
+ } else {
+ s->stop_found_periods = 0;
+ s->start_found_periods = 0;
+ s->start_holdoff_offset = 0;
+ s->start_holdoff_end = 0;
+ clear_rms(s);
+ s->mode = SILENCE_TRIM;
+ flush(out, outlink, &nb_samples_written, &ret);
+ goto silence_trim;
+ }
+ } else {
+ s->mode = SILENCE_COPY_FLUSH;
+ flush(out, outlink, &nb_samples_written, &ret);
+ goto silence_copy_flush;
+ }
+ flush(out, outlink, &nb_samples_written, &ret);
+ break;
+ }
+ }
+ }
+ flush(out, outlink, &nb_samples_written, &ret);
+ } else {
+ memcpy(obuf, ibuf, sizeof(double) * nbs * inlink->channels);
+ ret = ff_filter_frame(outlink, out);
+ }
+ break;
+
+ case SILENCE_COPY_FLUSH:
+silence_copy_flush:
+ nbs = s->stop_holdoff_end - s->stop_holdoff_offset;
+ nbs -= nbs % inlink->channels;
+
+ out = ff_get_audio_buffer(inlink, nbs / inlink->channels);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+
+ memcpy(out->data[0], &s->stop_holdoff[s->stop_holdoff_offset],
+ nbs * sizeof(double));
+ s->stop_holdoff_offset += nbs;
+
+ ret = ff_filter_frame(outlink, out);
+
+ if (s->stop_holdoff_offset == s->stop_holdoff_end) {
+ s->stop_holdoff_offset = 0;
+ s->stop_holdoff_end = 0;
+ s->mode = SILENCE_COPY;
+ goto silence_copy;
+ }
+ break;
+ case SILENCE_STOP:
+silence_stop:
+ break;
+ }
+
+ av_frame_free(&in);
+
+ return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ SilenceRemoveContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+ if (ret == AVERROR_EOF && (s->mode == SILENCE_COPY_FLUSH ||
+ s->mode == SILENCE_COPY)) {
+ int nbs = s->stop_holdoff_end - s->stop_holdoff_offset;
+ if (nbs) {
+ AVFrame *frame;
+
+ frame = ff_get_audio_buffer(outlink, nbs / outlink->channels);
+ if (!frame)
+ return AVERROR(ENOMEM);
+
+ memcpy(frame->data[0], &s->stop_holdoff[s->stop_holdoff_offset],
+ nbs * sizeof(double));
+ ret = ff_filter_frame(ctx->inputs[0], frame);
+ }
+ s->mode = SILENCE_STOP;
+ }
+ return ret;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_NONE
+ };
+
+ layouts = ff_all_channel_layouts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ff_set_common_channel_layouts(ctx, layouts);
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_set_common_formats(ctx, formats);
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_set_common_samplerates(ctx, formats);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ SilenceRemoveContext *s = ctx->priv;
+
+ av_freep(&s->start_holdoff);
+ av_freep(&s->stop_holdoff);
+ av_freep(&s->window);
+}
+
+static const AVFilterPad silenceremove_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad silenceremove_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ .request_frame = request_frame,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_silenceremove = {
+ .name = "silenceremove",
+ .description = NULL_IF_CONFIG_SMALL("Remove silence."),
+ .priv_size = sizeof(SilenceRemoveContext),
+ .priv_class = &silenceremove_class,
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = silenceremove_inputs,
+ .outputs = silenceremove_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 8fe2020..670f2d1 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -96,6 +96,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(REPLAYGAIN, replaygain, af);
REGISTER_FILTER(RESAMPLE, resample, af);
REGISTER_FILTER(SILENCEDETECT, silencedetect, af);
+ REGISTER_FILTER(SILENCEREMOVE, silenceremove, af);
REGISTER_FILTER(TREBLE, treble, af);
REGISTER_FILTER(VOLUME, volume, af);
REGISTER_FILTER(VOLUMEDETECT, volumedetect, af);
--
1.7.11.2
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