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FFmpeg/libavfilter/af_volumedetect.c
Andreas Rheinhardt 8be701d9f7 avfilter/avfilter: Add numbers of (in|out)pads directly to AVFilter
Up until now, an AVFilter's lists of input and output AVFilterPads
were terminated by a sentinel and the only way to get the length
of these lists was by using avfilter_pad_count(). This has two
drawbacks: first, sizeof(AVFilterPad) is not negligible
(i.e. 64B on 64bit systems); second, getting the size involves
a function call instead of just reading the data.

This commit therefore changes this. The sentinels are removed and new
private fields nb_inputs and nb_outputs are added to AVFilter that
contain the number of elements of the respective AVFilterPad array.

Given that AVFilter.(in|out)puts are the only arrays of zero-terminated
AVFilterPads an API user has access to (AVFilterContext.(in|out)put_pads
are not zero-terminated and they already have a size field) the argument
to avfilter_pad_count() is always one of these lists, so it just has to
find the filter the list belongs to and read said number. This is slower
than before, but a replacement function that just reads the internal numbers
that users are expected to switch to will be added soon; and furthermore,
avfilter_pad_count() is probably never called in hot loops anyway.

This saves about 49KiB from the binary; notice that these sentinels are
not in .bss despite being zeroed: they are in .data.rel.ro due to the
non-sentinels.

Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-08-20 12:53:58 +02:00

155 lines
4.7 KiB
C

/*
* 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/channel_layout.h"
#include "libavutil/avassert.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct VolDetectContext {
/**
* 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)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
return ff_set_common_formats_from_list(ctx, sample_fmts);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
{
AVFilterContext *ctx = inlink->dst;
VolDetectContext *vd = ctx->priv;
int nb_samples = samples->nb_samples;
int nb_channels = samples->channels;
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_frame(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 -log10(d) * 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 av_cold void uninit(AVFilterContext *ctx)
{
print_stats(ctx);
}
static const AVFilterPad volumedetect_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad volumedetect_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_volumedetect = {
.name = "volumedetect",
.description = NULL_IF_CONFIG_SMALL("Detect audio volume."),
.priv_size = sizeof(VolDetectContext),
.query_formats = query_formats,
.uninit = uninit,
FILTER_INPUTS(volumedetect_inputs),
FILTER_OUTPUTS(volumedetect_outputs),
};