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If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
293 lines
12 KiB
C
293 lines
12 KiB
C
/*
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* Copyright (c) 2012 Clément Bœsch <u pkh me>
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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 <float.h> /* DBL_MAX */
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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 SilenceDetectContext {
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const AVClass *class;
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double noise; ///< noise amplitude ratio
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int64_t duration; ///< minimum duration of silence until notification
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int mono; ///< mono mode : check each channel separately (default = check when ALL channels are silent)
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int channels; ///< number of channels
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int independent_channels; ///< number of entries in following arrays (always 1 in mono mode)
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int64_t *nb_null_samples; ///< (array) current number of continuous zero samples
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int64_t *start; ///< (array) if silence is detected, this value contains the time of the first zero sample (default/unset = INT64_MIN)
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int64_t frame_end; ///< pts of the end of the current frame (used to compute duration of silence at EOS)
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int last_sample_rate; ///< last sample rate to check for sample rate changes
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AVRational time_base; ///< time_base
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void (*silencedetect)(struct SilenceDetectContext *s, AVFrame *insamples,
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int nb_samples, int64_t nb_samples_notify,
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AVRational time_base);
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} SilenceDetectContext;
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#define MAX_DURATION (24*3600*1000000LL)
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#define OFFSET(x) offsetof(SilenceDetectContext, x)
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#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption silencedetect_options[] = {
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{ "n", "set noise tolerance", OFFSET(noise), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0, DBL_MAX, FLAGS },
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{ "noise", "set noise tolerance", OFFSET(noise), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0, DBL_MAX, FLAGS },
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{ "d", "set minimum duration in seconds", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=2000000}, 0, MAX_DURATION,FLAGS },
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{ "duration", "set minimum duration in seconds", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64=2000000}, 0, MAX_DURATION,FLAGS },
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{ "mono", "check each channel separately", OFFSET(mono), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
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{ "m", "check each channel separately", OFFSET(mono), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(silencedetect);
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static void set_meta(AVFrame *insamples, int channel, const char *key, char *value)
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{
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char key2[128];
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if (channel)
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snprintf(key2, sizeof(key2), "lavfi.%s.%d", key, channel);
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else
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snprintf(key2, sizeof(key2), "lavfi.%s", key);
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av_dict_set(&insamples->metadata, key2, value, 0);
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}
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static av_always_inline void update(SilenceDetectContext *s, AVFrame *insamples,
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int is_silence, int current_sample, int64_t nb_samples_notify,
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AVRational time_base)
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{
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int channel = current_sample % s->independent_channels;
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if (is_silence) {
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if (s->start[channel] == INT64_MIN) {
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s->nb_null_samples[channel]++;
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if (s->nb_null_samples[channel] >= nb_samples_notify) {
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s->start[channel] = insamples->pts + av_rescale_q(current_sample / s->channels + 1 - nb_samples_notify * s->independent_channels / s->channels,
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(AVRational){ 1, s->last_sample_rate }, time_base);
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set_meta(insamples, s->mono ? channel + 1 : 0, "silence_start",
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av_ts2timestr(s->start[channel], &time_base));
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if (s->mono)
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av_log(s, AV_LOG_INFO, "channel: %d | ", channel);
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av_log(s, AV_LOG_INFO, "silence_start: %s\n",
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av_ts2timestr(s->start[channel], &time_base));
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}
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}
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} else {
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if (s->start[channel] > INT64_MIN) {
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int64_t end_pts = insamples ? insamples->pts + av_rescale_q(current_sample / s->channels,
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(AVRational){ 1, s->last_sample_rate }, time_base)
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: s->frame_end;
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int64_t duration_ts = end_pts - s->start[channel];
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if (insamples) {
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set_meta(insamples, s->mono ? channel + 1 : 0, "silence_end",
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av_ts2timestr(end_pts, &time_base));
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set_meta(insamples, s->mono ? channel + 1 : 0, "silence_duration",
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av_ts2timestr(duration_ts, &time_base));
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}
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if (s->mono)
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av_log(s, AV_LOG_INFO, "channel: %d | ", channel);
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av_log(s, AV_LOG_INFO, "silence_end: %s | silence_duration: %s\n",
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av_ts2timestr(end_pts, &time_base),
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av_ts2timestr(duration_ts, &time_base));
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}
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s->nb_null_samples[channel] = 0;
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s->start[channel] = INT64_MIN;
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}
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}
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#define SILENCE_DETECT(name, type) \
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static void silencedetect_##name(SilenceDetectContext *s, AVFrame *insamples, \
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int nb_samples, int64_t nb_samples_notify, \
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AVRational time_base) \
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{ \
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const type *p = (const type *)insamples->data[0]; \
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const type noise = s->noise; \
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int i; \
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\
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for (i = 0; i < nb_samples; i++, p++) \
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update(s, insamples, *p < noise && *p > -noise, i, \
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nb_samples_notify, time_base); \
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}
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#define SILENCE_DETECT_PLANAR(name, type) \
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static void silencedetect_##name(SilenceDetectContext *s, AVFrame *insamples, \
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int nb_samples, int64_t nb_samples_notify, \
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AVRational time_base) \
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{ \
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const int channels = insamples->channels; \
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const type noise = s->noise; \
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\
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nb_samples /= channels; \
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for (int i = 0; i < nb_samples; i++) { \
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for (int ch = 0; ch < insamples->channels; ch++) { \
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const type *p = (const type *)insamples->extended_data[ch]; \
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update(s, insamples, p[i] < noise && p[i] > -noise, \
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channels * i + ch, \
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nb_samples_notify, time_base); \
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} \
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} \
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}
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SILENCE_DETECT(dbl, double)
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SILENCE_DETECT(flt, float)
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SILENCE_DETECT(s32, int32_t)
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SILENCE_DETECT(s16, int16_t)
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SILENCE_DETECT_PLANAR(dblp, double)
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SILENCE_DETECT_PLANAR(fltp, float)
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SILENCE_DETECT_PLANAR(s32p, int32_t)
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SILENCE_DETECT_PLANAR(s16p, int16_t)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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SilenceDetectContext *s = ctx->priv;
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int c;
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s->channels = inlink->channels;
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s->duration = av_rescale(s->duration, inlink->sample_rate, AV_TIME_BASE);
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s->independent_channels = s->mono ? s->channels : 1;
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s->nb_null_samples = av_calloc(s->independent_channels,
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sizeof(*s->nb_null_samples));
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if (!s->nb_null_samples)
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return AVERROR(ENOMEM);
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s->start = av_malloc_array(sizeof(*s->start), s->independent_channels);
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if (!s->start)
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return AVERROR(ENOMEM);
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for (c = 0; c < s->independent_channels; c++)
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s->start[c] = INT64_MIN;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBL: s->silencedetect = silencedetect_dbl; break;
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case AV_SAMPLE_FMT_FLT: s->silencedetect = silencedetect_flt; break;
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case AV_SAMPLE_FMT_S32:
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s->noise *= INT32_MAX;
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s->silencedetect = silencedetect_s32;
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break;
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case AV_SAMPLE_FMT_S16:
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s->noise *= INT16_MAX;
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s->silencedetect = silencedetect_s16;
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break;
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case AV_SAMPLE_FMT_DBLP: s->silencedetect = silencedetect_dblp; break;
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case AV_SAMPLE_FMT_FLTP: s->silencedetect = silencedetect_fltp; break;
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case AV_SAMPLE_FMT_S32P:
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s->noise *= INT32_MAX;
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s->silencedetect = silencedetect_s32p;
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break;
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case AV_SAMPLE_FMT_S16P:
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s->noise *= INT16_MAX;
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s->silencedetect = silencedetect_s16p;
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break;
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default:
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return AVERROR_BUG;
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
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{
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SilenceDetectContext *s = inlink->dst->priv;
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const int nb_channels = inlink->channels;
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const int srate = inlink->sample_rate;
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const int nb_samples = insamples->nb_samples * nb_channels;
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const int64_t nb_samples_notify = s->duration * (s->mono ? 1 : nb_channels);
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int c;
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// scale number of null samples to the new sample rate
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if (s->last_sample_rate && s->last_sample_rate != srate)
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for (c = 0; c < s->independent_channels; c++) {
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s->nb_null_samples[c] = srate * s->nb_null_samples[c] / s->last_sample_rate;
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}
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s->last_sample_rate = srate;
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s->time_base = inlink->time_base;
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s->frame_end = insamples->pts + av_rescale_q(insamples->nb_samples,
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(AVRational){ 1, s->last_sample_rate }, inlink->time_base);
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s->silencedetect(s, insamples, nb_samples, nb_samples_notify,
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inlink->time_base);
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return ff_filter_frame(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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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_NONE
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};
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int ret = ff_set_common_all_channel_counts(ctx);
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if (ret < 0)
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return ret;
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ret = ff_set_common_formats_from_list(ctx, sample_fmts);
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if (ret < 0)
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return ret;
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return ff_set_common_all_samplerates(ctx);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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SilenceDetectContext *s = ctx->priv;
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int c;
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for (c = 0; c < s->independent_channels; c++)
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if (s->start[c] > INT64_MIN)
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update(s, NULL, 0, c, 0, s->time_base);
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av_freep(&s->nb_null_samples);
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av_freep(&s->start);
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}
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static const AVFilterPad silencedetect_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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.filter_frame = filter_frame,
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},
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};
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static const AVFilterPad silencedetect_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_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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.uninit = uninit,
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FILTER_INPUTS(silencedetect_inputs),
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FILTER_OUTPUTS(silencedetect_outputs),
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FILTER_QUERY_FUNC(query_formats),
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.priv_class = &silencedetect_class,
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};
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