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50ea7389ec
Lots of audio filters use very simple inputs or outputs: An array with a single AVFilterPad whose name is "default" and whose type is AVMEDIA_TYPE_AUDIO; everything else is unset. Given that we never use pointer equality for inputs or outputs*, we can simply use a single AVFilterPad instead of dozens; this even saves .data.rel.ro (4784B here) as well as relocations. *: In fact, several filters (like the filters in af_biquads.c) already use the same inputs; furthermore, ff_filter_alloc() duplicates the input and output pads so that we do not even work with the pads directly. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
135 lines
4.3 KiB
C
135 lines
4.3 KiB
C
/*
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* Copyright (c) 2012 Nicolas George
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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 License
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* 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
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* GNU 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 License
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* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/avassert.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct VolDetectContext {
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/**
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* Number of samples at each PCM value.
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* histogram[0x8000 + i] is the number of samples at value i.
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* The extra element is there for symmetry.
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*/
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uint64_t histogram[0x10001];
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} VolDetectContext;
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static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
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{
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AVFilterContext *ctx = inlink->dst;
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VolDetectContext *vd = ctx->priv;
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int nb_samples = samples->nb_samples;
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int nb_channels = samples->ch_layout.nb_channels;
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int nb_planes = nb_channels;
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int plane, i;
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int16_t *pcm;
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if (!av_sample_fmt_is_planar(samples->format)) {
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nb_samples *= nb_channels;
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nb_planes = 1;
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}
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for (plane = 0; plane < nb_planes; plane++) {
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pcm = (int16_t *)samples->extended_data[plane];
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for (i = 0; i < nb_samples; i++)
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vd->histogram[pcm[i] + 0x8000]++;
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}
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return ff_filter_frame(inlink->dst->outputs[0], samples);
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}
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#define MAX_DB 91
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static inline double logdb(uint64_t v)
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{
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double d = v / (double)(0x8000 * 0x8000);
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if (!v)
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return MAX_DB;
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return -log10(d) * 10;
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}
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static void print_stats(AVFilterContext *ctx)
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{
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VolDetectContext *vd = ctx->priv;
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int i, max_volume, shift;
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uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
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uint64_t histdb[MAX_DB + 1] = { 0 };
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for (i = 0; i < 0x10000; i++)
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nb_samples += vd->histogram[i];
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av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
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if (!nb_samples)
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return;
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/* If nb_samples > 1<<34, there is a risk of overflow in the
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multiplication or the sum: shift all histogram values to avoid that.
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The total number of samples must be recomputed to avoid rounding
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errors. */
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shift = av_log2(nb_samples >> 33);
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for (i = 0; i < 0x10000; i++) {
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nb_samples_shift += vd->histogram[i] >> shift;
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power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
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}
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if (!nb_samples_shift)
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return;
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power = (power + nb_samples_shift / 2) / nb_samples_shift;
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av_assert0(power <= 0x8000 * 0x8000);
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av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
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max_volume = 0x8000;
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while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
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!vd->histogram[0x8000 - max_volume])
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max_volume--;
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av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));
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for (i = 0; i < 0x10000; i++)
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histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
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for (i = 0; i <= MAX_DB && !histdb[i]; i++);
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for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
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av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);
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sum += histdb[i];
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}
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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print_stats(ctx);
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}
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static const AVFilterPad volumedetect_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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};
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const AVFilter ff_af_volumedetect = {
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.name = "volumedetect",
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.description = NULL_IF_CONFIG_SMALL("Detect audio volume."),
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.priv_size = sizeof(VolDetectContext),
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.uninit = uninit,
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.flags = AVFILTER_FLAG_METADATA_ONLY,
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FILTER_INPUTS(volumedetect_inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P),
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};
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