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https://github.com/FFmpeg/FFmpeg.git
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f8911b987d
* qatar/master: mss3: use standard zigzag table mss3: split DSP functions that are used in MTS2(MSS4) into separate file motion-test: do not use getopt() tcp: add initial timeout limit for incoming connections configure: Change the rdtsc check to a linker check avconv: propagate fatal errors from lavfi. lavfi: add error handling to filter_samples(). fate-run: make avconv() properly deal with multiple inputs. asplit: don't leak the input buffer. af_resample: fix request_frame() behavior. af_asyncts: fix request_frame() behavior. libx264: support aspect ratio switching matroskadec: honor error_recognition when encountering unknown elements. lavr: resampling: add support for s32p, fltp, and dblp internal sample formats lavr: resampling: add filter type and Kaiser window beta to AVOptions lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format lavr: mix: validate internal sample format in ff_audio_mix_init() Conflicts: ffmpeg.c ffplay.c libavcodec/libx264.c libavfilter/audio.c libavfilter/split.c libavformat/tcp.c tests/fate-run.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>
171 lines
6.2 KiB
C
171 lines
6.2 KiB
C
/*
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* Copyright (c) 2012 Clément Bœsch <ubitux@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Audio silence detector
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/opt.h"
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#include "libavutil/timestamp.h"
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#include "audio.h"
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#include "formats.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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const AVClass *class;
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char *noise_str; ///< noise option string
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double noise; ///< noise amplitude ratio
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int duration; ///< minimum duration of silence until notification
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int64_t nb_null_samples; ///< current number of continuous zero samples
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int64_t start; ///< if silence is detected, this value contains the time of the first zero sample
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int last_sample_rate; ///< last sample rate to check for sample rate changes
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} SilenceDetectContext;
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#define OFFSET(x) offsetof(SilenceDetectContext, x)
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static const AVOption silencedetect_options[] = {
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{ "n", "set noise tolerance", OFFSET(noise_str), AV_OPT_TYPE_STRING, {.str="-60dB"}, CHAR_MIN, CHAR_MAX },
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{ "noise", "set noise tolerance", OFFSET(noise_str), AV_OPT_TYPE_STRING, {.str="-60dB"}, CHAR_MIN, CHAR_MAX },
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{ "d", "set minimum duration in seconds", OFFSET(duration), AV_OPT_TYPE_INT, {.dbl=2}, 0, INT_MAX},
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{ "duration", "set minimum duration in seconds", OFFSET(duration), AV_OPT_TYPE_INT, {.dbl=2}, 0, INT_MAX},
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{ NULL },
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};
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AVFILTER_DEFINE_CLASS(silencedetect);
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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int ret;
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char *tail;
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SilenceDetectContext *silence = ctx->priv;
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silence->class = &silencedetect_class;
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av_opt_set_defaults(silence);
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if ((ret = av_set_options_string(silence, args, "=", ":")) < 0) {
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av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
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return ret;
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}
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silence->noise = strtod(silence->noise_str, &tail);
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if (!strcmp(tail, "dB")) {
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silence->noise = pow(10, silence->noise/20);
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} else if (*tail) {
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av_log(ctx, AV_LOG_ERROR, "Invalid value '%s' for noise parameter.\n",
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silence->noise_str);
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return AVERROR(EINVAL);
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}
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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int i;
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SilenceDetectContext *silence = inlink->dst->priv;
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const int nb_channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
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const int srate = inlink->sample_rate;
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const int nb_samples = insamples->audio->nb_samples * nb_channels;
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const int64_t nb_samples_notify = srate * silence->duration * nb_channels;
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// scale number of null samples to the new sample rate
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if (silence->last_sample_rate && silence->last_sample_rate != srate)
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silence->nb_null_samples =
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srate * silence->nb_null_samples / silence->last_sample_rate;
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silence->last_sample_rate = srate;
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// TODO: support more sample formats
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if (insamples->format == AV_SAMPLE_FMT_DBL) {
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double *p = (double *)insamples->data[0];
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for (i = 0; i < nb_samples; i++, p++) {
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if (*p < silence->noise && *p > -silence->noise) {
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if (!silence->start) {
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silence->nb_null_samples++;
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if (silence->nb_null_samples >= nb_samples_notify) {
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silence->start = insamples->pts - silence->duration / av_q2d(inlink->time_base);
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av_log(silence, AV_LOG_INFO,
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"silence_start: %s\n", av_ts2timestr(silence->start, &inlink->time_base));
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}
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}
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} else {
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if (silence->start)
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av_log(silence, AV_LOG_INFO,
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"silence_end: %s | silence_duration: %s\n",
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av_ts2timestr(insamples->pts, &inlink->time_base),
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av_ts2timestr(insamples->pts - silence->start, &inlink->time_base));
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silence->nb_null_samples = silence->start = 0;
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}
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}
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}
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return ff_filter_samples(inlink->dst->outputs[0], insamples);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts = NULL;
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enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBL,
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AV_SAMPLE_FMT_NONE
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};
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_set_common_channel_layouts(ctx, layouts);
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_formats(ctx, formats);
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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AVFilter avfilter_af_silencedetect = {
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.name = "silencedetect",
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.description = NULL_IF_CONFIG_SMALL("Detect silence."),
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.priv_size = sizeof(SilenceDetectContext),
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.init = init,
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.query_formats = query_formats,
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.inputs = (const AVFilterPad[]) {
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{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.get_audio_buffer = ff_null_get_audio_buffer,
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.filter_samples = filter_samples, },
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{ .name = NULL }
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},
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.outputs = (const AVFilterPad[]) {
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{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO, },
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{ .name = NULL }
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},
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};
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