From 422619646ea0e938188a49a06226831cc42e2a6a Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Wed, 2 Jul 2014 09:39:07 +0000 Subject: [PATCH] add silenceremove filter Signed-off-by: Paul B Mahol --- Changelog | 1 + MAINTAINERS | 1 + RELEASE_NOTES | 1 + doc/filters.texi | 69 +++++ libavfilter/Makefile | 1 + libavfilter/af_silenceremove.c | 482 +++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + libavfilter/version.h | 4 +- 8 files changed, 558 insertions(+), 2 deletions(-) create mode 100644 libavfilter/af_silenceremove.c diff --git a/Changelog b/Changelog index 72c783aee9..c5185f7eec 100644 --- a/Changelog +++ b/Changelog @@ -13,6 +13,7 @@ version : - added codecview filter to visualize information exported by some codecs - Matroska 3D support thorugh side data - HTML generation using texi2html is deprecated in favor of makeinfo/texi2any +- silenceremove filter version 2.3: diff --git a/MAINTAINERS b/MAINTAINERS index eb1f98bdd9..8af02ae62a 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -343,6 +343,7 @@ Filters: af_compand.c Paul B Mahol af_ladspa.c Paul B Mahol af_pan.c Nicolas George + af_silenceremove.c Paul B Mahol avf_avectorscope.c Paul B Mahol avf_showcqt.c Muhammad Faiz vf_blend.c Paul B Mahol diff --git a/RELEASE_NOTES b/RELEASE_NOTES index 15eb81b476..113cc5e97d 100644 --- a/RELEASE_NOTES +++ b/RELEASE_NOTES @@ -42,6 +42,7 @@ • ported lenscorrection filter from frei0r filter • large optimizations in dctdnoiz to make it usable • added codecview filter to visualize information exported by some codecs + • added silenceremove filter ┌────────────────────────────┐ │ libavutil │ diff --git a/doc/filters.texi b/doc/filters.texi index d13278ef71..627f112724 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1875,6 +1875,75 @@ ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null - @end example @end itemize +@section silenceremove + +Remove silence from the beginning, middle or end of the audio. + +The filter accepts the following options: + +@table @option +@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}. + +@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}. + +@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}. + +@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 +@var{start_periods}, making it suitable for removing periods of silence +in the middle of the audio. +Default value is @code{0}. + +@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}. + +@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}. + +@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}. + +@end table + +@subsection Examples + +@itemize +@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: +@example +silenceremove=1:5:0.02 +@end example +@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 ce71ce1a24..3241b76f10 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 0000000000..3f6cb7a550 --- /dev/null +++ b/libavfilter/af_silenceremove.c @@ -0,0 +1,482 @@ +/* + * 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 /* 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); + + s->start_holdoff = av_malloc_array(FFMAX(s->start_duration, 1), + 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; + + s->stop_holdoff = av_malloc_array(FFMAX(s->stop_duration, 1), + 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; + if (!nbs) + break; + + 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; + if (!nbs) + break; + + 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; + if (!nbs) + break; + + 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 8fe2020901..670f2d1c77 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); diff --git a/libavfilter/version.h b/libavfilter/version.h index 7c9883d494..6ebb1ce609 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,8 +30,8 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 5 -#define LIBAVFILTER_VERSION_MINOR 0 -#define LIBAVFILTER_VERSION_MICRO 103 +#define LIBAVFILTER_VERSION_MINOR 1 +#define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ LIBAVFILTER_VERSION_MINOR, \