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134 lines
4.3 KiB
134 lines
4.3 KiB
/* |
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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 "filters.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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