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/*
* Copyright (c) 2012 Nicolas George
*
* 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 "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct {
/**
* Number of samples at each PCM value.
* histogram[0x8000 + i] is the number of samples at value i.
* The extra element is there for symmetry.
*/
uint64_t histogram[0x10001];
} VolDetectContext;
static int query_formats(AVFilterContext *ctx)
{
enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE
};
AVFilterFormats *formats;
if (!(formats = ff_make_format_list(sample_fmts)))
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samples)
{
AVFilterContext *ctx = inlink->dst;
VolDetectContext *vd = ctx->priv;
int64_t layout = samples->audio->channel_layout;
int nb_samples = samples->audio->nb_samples;
int nb_channels = av_get_channel_layout_nb_channels(layout);
int nb_planes = nb_channels;
int plane, i;
int16_t *pcm;
if (!av_sample_fmt_is_planar(samples->format)) {
nb_samples *= nb_channels;
nb_planes = 1;
}
for (plane = 0; plane < nb_planes; plane++) {
pcm = (int16_t *)samples->extended_data[plane];
for (i = 0; i < nb_samples; i++)
vd->histogram[pcm[i] + 0x8000]++;
}
return ff_filter_samples(inlink->dst->outputs[0], samples);
}
#define MAX_DB 91
static inline double logdb(uint64_t v)
{
double d = v / (double)(0x8000 * 0x8000);
if (!v)
return MAX_DB;
return log(d) * -4.3429448190325182765112891891660508229; /* -10/log(10) */
}
static void print_stats(AVFilterContext *ctx)
{
VolDetectContext *vd = ctx->priv;
int i, max_volume, shift;
uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
uint64_t histdb[MAX_DB + 1] = { 0 };
for (i = 0; i < 0x10000; i++)
nb_samples += vd->histogram[i];
av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
if (!nb_samples)
return;
/* If nb_samples > 1<<34, there is a risk of overflow in the
multiplication or the sum: shift all histogram values to avoid that.
The total number of samples must be recomputed to avoid rounding
errors. */
shift = av_log2(nb_samples >> 33);
for (i = 0; i < 0x10000; i++) {
nb_samples_shift += vd->histogram[i] >> shift;
power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
}
if (!nb_samples_shift)
return;
power = (power + nb_samples_shift / 2) / nb_samples_shift;
av_assert0(power <= 0x8000 * 0x8000);
av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
max_volume = 0x8000;
while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
!vd->histogram[0x8000 - max_volume])
max_volume--;
av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));
for (i = 0; i < 0x10000; i++)
histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
for (i = 0; i <= MAX_DB && !histdb[i]; i++);
for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);
sum += histdb[i];
}
}
static void uninit(AVFilterContext *ctx)
{
print_stats(ctx);
}
AVFilter avfilter_af_volumedetect = {
.name = "volumedetect",
.description = NULL_IF_CONFIG_SMALL("Detect audio volume."),
.priv_size = sizeof(VolDetectContext),
.query_formats = query_formats,
.uninit = uninit,
.inputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL }
},
};
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