1038 lines
35 KiB
C
1038 lines
35 KiB
C
/*
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* Dynamic Audio Normalizer
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* Copyright (c) 2015 LoRd_MuldeR <mulder2@gmx.de>. Some rights reserved.
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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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* Dynamic Audio Normalizer
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*/
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#include <float.h>
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#define MIN_FILTER_SIZE 3
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#define MAX_FILTER_SIZE 301
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#define FF_BUFQUEUE_SIZE (MAX_FILTER_SIZE + 1)
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#include "libavfilter/bufferqueue.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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#include "internal.h"
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static const char * const var_names[] = {
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"ch", ///< the value of the current channel
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"sn", ///< number of samples
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"nb_channels",
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"t", ///< timestamp expressed in seconds
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"sr", ///< sample rate
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"p", ///< peak value
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NULL
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};
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enum var_name {
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VAR_CH,
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VAR_SN,
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VAR_NB_CHANNELS,
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VAR_T,
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VAR_SR,
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VAR_P,
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VAR_VARS_NB
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};
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typedef struct local_gain {
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double max_gain;
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double threshold;
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} local_gain;
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typedef struct cqueue {
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double *elements;
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int size;
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int max_size;
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int nb_elements;
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} cqueue;
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typedef struct DynamicAudioNormalizerContext {
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const AVClass *class;
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struct FFBufQueue queue;
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int frame_len;
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int frame_len_msec;
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int filter_size;
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int dc_correction;
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int channels_coupled;
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int alt_boundary_mode;
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double overlap;
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char *expr_str;
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double peak_value;
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double max_amplification;
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double target_rms;
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double compress_factor;
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double threshold;
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double *prev_amplification_factor;
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double *dc_correction_value;
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double *compress_threshold;
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double *weights;
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int channels;
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int sample_advance;
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int eof;
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char *channels_to_filter;
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AVChannelLayout ch_layout;
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int64_t pts;
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cqueue **gain_history_original;
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cqueue **gain_history_minimum;
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cqueue **gain_history_smoothed;
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cqueue **threshold_history;
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cqueue *is_enabled;
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AVFrame *window;
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AVExpr *expr;
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double var_values[VAR_VARS_NB];
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} DynamicAudioNormalizerContext;
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typedef struct ThreadData {
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AVFrame *in, *out;
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int enabled;
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} ThreadData;
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#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption dynaudnorm_options[] = {
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{ "framelen", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
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{ "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
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{ "gausssize", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
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{ "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
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{ "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
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{ "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
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{ "maxgain", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
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{ "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
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{ "targetrms", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
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{ "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
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{ "coupling", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
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{ "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
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{ "correctdc", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
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{ "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
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{ "altboundary", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
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{ "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
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{ "compress", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
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{ "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
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{ "threshold", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
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{ "t", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
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{ "channels", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
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{ "h", "set channels to filter", OFFSET(channels_to_filter),AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
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{ "overlap", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS },
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{ "o", "set the frame overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=.0}, 0.0, 1.0, FLAGS },
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{ "curve", "set the custom peak mapping curve",OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
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{ "v", "set the custom peak mapping curve",OFFSET(expr_str), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(dynaudnorm);
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static av_cold int init(AVFilterContext *ctx)
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{
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DynamicAudioNormalizerContext *s = ctx->priv;
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if (!(s->filter_size & 1)) {
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av_log(ctx, AV_LOG_WARNING, "filter size %d is invalid. Changing to an odd value.\n", s->filter_size);
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s->filter_size |= 1;
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}
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return 0;
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}
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static inline int frame_size(int sample_rate, int frame_len_msec)
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{
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const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0));
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return frame_size + (frame_size % 2);
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}
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static cqueue *cqueue_create(int size, int max_size)
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{
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cqueue *q;
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if (max_size < size)
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return NULL;
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q = av_malloc(sizeof(cqueue));
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if (!q)
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return NULL;
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q->max_size = max_size;
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q->size = size;
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q->nb_elements = 0;
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q->elements = av_malloc_array(max_size, sizeof(double));
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if (!q->elements) {
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av_free(q);
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return NULL;
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}
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return q;
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}
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static void cqueue_free(cqueue *q)
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{
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if (q)
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av_free(q->elements);
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av_free(q);
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}
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static int cqueue_size(cqueue *q)
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{
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return q->nb_elements;
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}
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static int cqueue_empty(cqueue *q)
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{
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return q->nb_elements <= 0;
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}
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static int cqueue_enqueue(cqueue *q, double element)
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{
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av_assert2(q->nb_elements < q->max_size);
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q->elements[q->nb_elements] = element;
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q->nb_elements++;
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return 0;
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}
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static double cqueue_peek(cqueue *q, int index)
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{
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av_assert2(index < q->nb_elements);
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return q->elements[index];
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}
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static int cqueue_dequeue(cqueue *q, double *element)
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{
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av_assert2(!cqueue_empty(q));
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*element = q->elements[0];
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memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
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q->nb_elements--;
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return 0;
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}
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static int cqueue_pop(cqueue *q)
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{
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av_assert2(!cqueue_empty(q));
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memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double));
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q->nb_elements--;
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return 0;
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}
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static void cqueue_resize(cqueue *q, int new_size)
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{
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av_assert2(q->max_size >= new_size);
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av_assert2(MIN_FILTER_SIZE <= new_size);
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if (new_size > q->nb_elements) {
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const int side = (new_size - q->nb_elements) / 2;
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memmove(q->elements + side, q->elements, sizeof(double) * q->nb_elements);
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for (int i = 0; i < side; i++)
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q->elements[i] = q->elements[side];
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q->nb_elements = new_size - 1 - side;
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} else {
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int count = (q->size - new_size + 1) / 2;
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while (count-- > 0)
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cqueue_pop(q);
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}
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q->size = new_size;
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}
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static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
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{
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double total_weight = 0.0;
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const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
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double adjust;
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// Pre-compute constants
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const int offset = s->filter_size / 2;
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const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
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const double c2 = 2.0 * sigma * sigma;
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// Compute weights
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for (int i = 0; i < s->filter_size; i++) {
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const int x = i - offset;
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s->weights[i] = c1 * exp(-x * x / c2);
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total_weight += s->weights[i];
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}
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// Adjust weights
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adjust = 1.0 / total_weight;
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for (int i = 0; i < s->filter_size; i++) {
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s->weights[i] *= adjust;
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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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DynamicAudioNormalizerContext *s = ctx->priv;
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av_freep(&s->prev_amplification_factor);
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av_freep(&s->dc_correction_value);
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av_freep(&s->compress_threshold);
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for (int c = 0; c < s->channels; c++) {
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if (s->gain_history_original)
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cqueue_free(s->gain_history_original[c]);
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if (s->gain_history_minimum)
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cqueue_free(s->gain_history_minimum[c]);
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if (s->gain_history_smoothed)
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cqueue_free(s->gain_history_smoothed[c]);
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if (s->threshold_history)
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cqueue_free(s->threshold_history[c]);
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}
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av_freep(&s->gain_history_original);
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av_freep(&s->gain_history_minimum);
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av_freep(&s->gain_history_smoothed);
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av_freep(&s->threshold_history);
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cqueue_free(s->is_enabled);
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s->is_enabled = NULL;
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av_freep(&s->weights);
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av_channel_layout_uninit(&s->ch_layout);
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ff_bufqueue_discard_all(&s->queue);
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av_frame_free(&s->window);
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av_expr_free(s->expr);
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s->expr = NULL;
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}
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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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DynamicAudioNormalizerContext *s = ctx->priv;
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int ret = 0;
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uninit(ctx);
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s->channels = inlink->ch_layout.nb_channels;
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s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
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av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
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s->prev_amplification_factor = av_malloc_array(inlink->ch_layout.nb_channels, sizeof(*s->prev_amplification_factor));
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s->dc_correction_value = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->dc_correction_value));
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s->compress_threshold = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->compress_threshold));
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s->gain_history_original = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_original));
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s->gain_history_minimum = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_minimum));
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s->gain_history_smoothed = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->gain_history_smoothed));
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s->threshold_history = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->threshold_history));
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s->weights = av_malloc_array(MAX_FILTER_SIZE, sizeof(*s->weights));
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s->is_enabled = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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if (!s->prev_amplification_factor || !s->dc_correction_value ||
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!s->compress_threshold ||
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!s->gain_history_original || !s->gain_history_minimum ||
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!s->gain_history_smoothed || !s->threshold_history ||
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!s->is_enabled || !s->weights)
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return AVERROR(ENOMEM);
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for (int c = 0; c < inlink->ch_layout.nb_channels; c++) {
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s->prev_amplification_factor[c] = 1.0;
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s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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s->gain_history_minimum[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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s->gain_history_smoothed[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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s->threshold_history[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
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if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
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!s->gain_history_smoothed[c] || !s->threshold_history[c])
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return AVERROR(ENOMEM);
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}
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init_gaussian_filter(s);
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s->window = ff_get_audio_buffer(ctx->outputs[0], s->frame_len * 2);
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if (!s->window)
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return AVERROR(ENOMEM);
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s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap)));
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s->var_values[VAR_SR] = inlink->sample_rate;
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s->var_values[VAR_NB_CHANNELS] = s->channels;
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if (s->expr_str)
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ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
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NULL, NULL, 0, ctx);
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return ret;
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}
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static inline double fade(double prev, double next, int pos, int length)
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{
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const double step_size = 1.0 / length;
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const double f0 = 1.0 - (step_size * (pos + 1.0));
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const double f1 = 1.0 - f0;
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return f0 * prev + f1 * next;
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}
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static inline double pow_2(const double value)
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{
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return value * value;
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}
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static inline double bound(const double threshold, const double val)
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{
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const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
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return erf(CONST * (val / threshold)) * threshold;
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}
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static double find_peak_magnitude(AVFrame *frame, int channel)
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{
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double max = DBL_EPSILON;
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if (channel == -1) {
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for (int c = 0; c < frame->ch_layout.nb_channels; c++) {
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double *data_ptr = (double *)frame->extended_data[c];
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for (int i = 0; i < frame->nb_samples; i++)
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max = fmax(max, fabs(data_ptr[i]));
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}
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} else {
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double *data_ptr = (double *)frame->extended_data[channel];
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for (int i = 0; i < frame->nb_samples; i++)
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max = fmax(max, fabs(data_ptr[i]));
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}
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return max;
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}
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static double compute_frame_rms(AVFrame *frame, int channel)
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{
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double rms_value = 0.0;
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if (channel == -1) {
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for (int c = 0; c < frame->ch_layout.nb_channels; c++) {
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const double *data_ptr = (double *)frame->extended_data[c];
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for (int i = 0; i < frame->nb_samples; i++) {
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rms_value += pow_2(data_ptr[i]);
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}
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}
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rms_value /= frame->nb_samples * frame->ch_layout.nb_channels;
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} else {
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const double *data_ptr = (double *)frame->extended_data[channel];
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for (int i = 0; i < frame->nb_samples; i++) {
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rms_value += pow_2(data_ptr[i]);
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}
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rms_value /= frame->nb_samples;
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}
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return fmax(sqrt(rms_value), DBL_EPSILON);
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}
|
|
|
|
static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
|
|
int channel)
|
|
{
|
|
const double peak_magnitude = find_peak_magnitude(frame, channel);
|
|
const double maximum_gain = s->peak_value / peak_magnitude;
|
|
const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
|
|
double target_gain = DBL_MAX;
|
|
local_gain gain;
|
|
|
|
if (s->expr_str) {
|
|
double var_values[VAR_VARS_NB];
|
|
|
|
memcpy(var_values, s->var_values, sizeof(var_values));
|
|
|
|
var_values[VAR_CH] = channel;
|
|
var_values[VAR_P] = peak_magnitude;
|
|
|
|
target_gain = av_expr_eval(s->expr, var_values, s) / peak_magnitude;
|
|
}
|
|
|
|
gain.threshold = peak_magnitude > s->threshold;
|
|
gain.max_gain = bound(s->max_amplification, fmin(target_gain, fmin(maximum_gain, rms_gain)));
|
|
|
|
return gain;
|
|
}
|
|
|
|
static double minimum_filter(cqueue *q)
|
|
{
|
|
double min = DBL_MAX;
|
|
|
|
for (int i = 0; i < cqueue_size(q); i++) {
|
|
min = fmin(min, cqueue_peek(q, i));
|
|
}
|
|
|
|
return min;
|
|
}
|
|
|
|
static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueue *tq)
|
|
{
|
|
const double *weights = s->weights;
|
|
double result = 0.0, tsum = 0.0;
|
|
|
|
for (int i = 0; i < cqueue_size(q); i++) {
|
|
double tq_item = cqueue_peek(tq, i);
|
|
double q_item = cqueue_peek(q, i);
|
|
|
|
tsum += tq_item * weights[i];
|
|
result += tq_item * weights[i] * q_item;
|
|
}
|
|
|
|
if (tsum == 0.0)
|
|
result = 1.0;
|
|
|
|
return result;
|
|
}
|
|
|
|
static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
|
|
local_gain gain)
|
|
{
|
|
if (cqueue_empty(s->gain_history_original[channel])) {
|
|
const int pre_fill_size = s->filter_size / 2;
|
|
const double initial_value = s->alt_boundary_mode ? gain.max_gain : fmin(1.0, gain.max_gain);
|
|
|
|
s->prev_amplification_factor[channel] = initial_value;
|
|
|
|
while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
|
|
cqueue_enqueue(s->gain_history_original[channel], initial_value);
|
|
cqueue_enqueue(s->threshold_history[channel], gain.threshold);
|
|
}
|
|
}
|
|
|
|
cqueue_enqueue(s->gain_history_original[channel], gain.max_gain);
|
|
|
|
while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
|
|
double minimum;
|
|
|
|
if (cqueue_empty(s->gain_history_minimum[channel])) {
|
|
const int pre_fill_size = s->filter_size / 2;
|
|
double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0;
|
|
int input = pre_fill_size;
|
|
|
|
while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
|
|
input++;
|
|
initial_value = fmin(initial_value, cqueue_peek(s->gain_history_original[channel], input));
|
|
cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
|
|
}
|
|
}
|
|
|
|
minimum = minimum_filter(s->gain_history_original[channel]);
|
|
|
|
cqueue_enqueue(s->gain_history_minimum[channel], minimum);
|
|
|
|
cqueue_enqueue(s->threshold_history[channel], gain.threshold);
|
|
|
|
cqueue_pop(s->gain_history_original[channel]);
|
|
}
|
|
|
|
while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
|
|
double smoothed, limit;
|
|
|
|
smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
|
|
limit = cqueue_peek(s->gain_history_original[channel], 0);
|
|
smoothed = fmin(smoothed, limit);
|
|
|
|
cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
|
|
|
|
cqueue_pop(s->gain_history_minimum[channel]);
|
|
cqueue_pop(s->threshold_history[channel]);
|
|
}
|
|
}
|
|
|
|
static int update_gain_histories(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
|
|
{
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
AVFrame *analyze_frame = arg;
|
|
const int channels = s->channels;
|
|
const int start = (channels * jobnr) / nb_jobs;
|
|
const int end = (channels * (jobnr+1)) / nb_jobs;
|
|
|
|
for (int c = start; c < end; c++)
|
|
update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static inline double update_value(double new, double old, double aggressiveness)
|
|
{
|
|
av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
|
|
return aggressiveness * new + (1.0 - aggressiveness) * old;
|
|
}
|
|
|
|
static inline int bypass_channel(DynamicAudioNormalizerContext *s, AVFrame *frame, int ch)
|
|
{
|
|
enum AVChannel channel = av_channel_layout_channel_from_index(&frame->ch_layout, ch);
|
|
|
|
return av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0;
|
|
}
|
|
|
|
static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
|
|
{
|
|
const double diff = 1.0 / frame->nb_samples;
|
|
int is_first_frame = cqueue_empty(s->gain_history_original[0]);
|
|
|
|
for (int c = 0; c < s->channels; c++) {
|
|
const int bypass = bypass_channel(s, frame, c);
|
|
double *dst_ptr = (double *)frame->extended_data[c];
|
|
double current_average_value = 0.0;
|
|
double prev_value;
|
|
|
|
for (int i = 0; i < frame->nb_samples; i++)
|
|
current_average_value += dst_ptr[i] * diff;
|
|
|
|
prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
|
|
s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
|
|
|
|
for (int i = 0; i < frame->nb_samples && !bypass; i++) {
|
|
dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
|
|
}
|
|
}
|
|
}
|
|
|
|
static double setup_compress_thresh(double threshold)
|
|
{
|
|
if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) {
|
|
double current_threshold = threshold;
|
|
double step_size = 1.0;
|
|
|
|
while (step_size > DBL_EPSILON) {
|
|
while ((llrint((current_threshold + step_size) * (UINT64_C(1) << 63)) >
|
|
llrint(current_threshold * (UINT64_C(1) << 63))) &&
|
|
(bound(current_threshold + step_size, 1.0) <= threshold)) {
|
|
current_threshold += step_size;
|
|
}
|
|
|
|
step_size /= 2.0;
|
|
}
|
|
|
|
return current_threshold;
|
|
} else {
|
|
return threshold;
|
|
}
|
|
}
|
|
|
|
static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
|
|
AVFrame *frame, int channel)
|
|
{
|
|
double variance = 0.0;
|
|
|
|
if (channel == -1) {
|
|
for (int c = 0; c < s->channels; c++) {
|
|
const double *data_ptr = (double *)frame->extended_data[c];
|
|
|
|
for (int i = 0; i < frame->nb_samples; i++) {
|
|
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
|
|
}
|
|
}
|
|
variance /= (s->channels * frame->nb_samples) - 1;
|
|
} else {
|
|
const double *data_ptr = (double *)frame->extended_data[channel];
|
|
|
|
for (int i = 0; i < frame->nb_samples; i++) {
|
|
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
|
|
}
|
|
variance /= frame->nb_samples - 1;
|
|
}
|
|
|
|
return fmax(sqrt(variance), DBL_EPSILON);
|
|
}
|
|
|
|
static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
|
|
{
|
|
int is_first_frame = cqueue_empty(s->gain_history_original[0]);
|
|
|
|
if (s->channels_coupled) {
|
|
const double standard_deviation = compute_frame_std_dev(s, frame, -1);
|
|
const double current_threshold = fmin(1.0, s->compress_factor * standard_deviation);
|
|
|
|
const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
|
|
double prev_actual_thresh, curr_actual_thresh;
|
|
s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
|
|
|
|
prev_actual_thresh = setup_compress_thresh(prev_value);
|
|
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
|
|
|
|
for (int c = 0; c < s->channels; c++) {
|
|
double *const dst_ptr = (double *)frame->extended_data[c];
|
|
const int bypass = bypass_channel(s, frame, c);
|
|
|
|
if (bypass)
|
|
continue;
|
|
|
|
for (int i = 0; i < frame->nb_samples; i++) {
|
|
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
|
|
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
|
|
}
|
|
}
|
|
} else {
|
|
for (int c = 0; c < s->channels; c++) {
|
|
const int bypass = bypass_channel(s, frame, c);
|
|
const double standard_deviation = compute_frame_std_dev(s, frame, c);
|
|
const double current_threshold = setup_compress_thresh(fmin(1.0, s->compress_factor * standard_deviation));
|
|
const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
|
|
double prev_actual_thresh, curr_actual_thresh;
|
|
double *dst_ptr;
|
|
|
|
s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
|
|
|
|
prev_actual_thresh = setup_compress_thresh(prev_value);
|
|
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
|
|
|
|
dst_ptr = (double *)frame->extended_data[c];
|
|
for (int i = 0; i < frame->nb_samples && !bypass; i++) {
|
|
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
|
|
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static int analyze_frame(AVFilterContext *ctx, AVFilterLink *outlink, AVFrame **frame)
|
|
{
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
AVFrame *analyze_frame;
|
|
|
|
if (s->dc_correction || s->compress_factor > DBL_EPSILON) {
|
|
int ret;
|
|
|
|
if (!av_frame_is_writable(*frame)) {
|
|
AVFrame *out = ff_get_audio_buffer(outlink, (*frame)->nb_samples);
|
|
|
|
if (!out) {
|
|
av_frame_free(frame);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
ret = av_frame_copy_props(out, *frame);
|
|
if (ret < 0) {
|
|
av_frame_free(frame);
|
|
av_frame_free(&out);
|
|
return ret;
|
|
}
|
|
ret = av_frame_copy(out, *frame);
|
|
if (ret < 0) {
|
|
av_frame_free(frame);
|
|
av_frame_free(&out);
|
|
return ret;
|
|
}
|
|
|
|
av_frame_free(frame);
|
|
*frame = out;
|
|
}
|
|
}
|
|
|
|
if (s->dc_correction)
|
|
perform_dc_correction(s, *frame);
|
|
|
|
if (s->compress_factor > DBL_EPSILON)
|
|
perform_compression(s, *frame);
|
|
|
|
if (s->frame_len != s->sample_advance) {
|
|
const int offset = s->frame_len - s->sample_advance;
|
|
|
|
for (int c = 0; c < s->channels; c++) {
|
|
double *src = (double *)s->window->extended_data[c];
|
|
|
|
memmove(src, &src[s->sample_advance], offset * sizeof(double));
|
|
memcpy(&src[offset], (*frame)->extended_data[c], (*frame)->nb_samples * sizeof(double));
|
|
memset(&src[offset + (*frame)->nb_samples], 0, (s->sample_advance - (*frame)->nb_samples) * sizeof(double));
|
|
}
|
|
|
|
analyze_frame = s->window;
|
|
} else {
|
|
av_samples_copy(s->window->extended_data, (*frame)->extended_data, 0, 0,
|
|
FFMIN(s->frame_len, (*frame)->nb_samples), (*frame)->ch_layout.nb_channels, (*frame)->format);
|
|
analyze_frame = *frame;
|
|
}
|
|
|
|
s->var_values[VAR_SN] = outlink->sample_count_in;
|
|
s->var_values[VAR_T] = s->var_values[VAR_SN] * (double)1/outlink->sample_rate;
|
|
|
|
if (s->channels_coupled) {
|
|
const local_gain gain = get_max_local_gain(s, analyze_frame, -1);
|
|
for (int c = 0; c < s->channels; c++)
|
|
update_gain_history(s, c, gain);
|
|
} else {
|
|
ff_filter_execute(ctx, update_gain_histories, analyze_frame, NULL,
|
|
FFMIN(s->channels, ff_filter_get_nb_threads(ctx)));
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void amplify_channel(DynamicAudioNormalizerContext *s, AVFrame *in,
|
|
AVFrame *frame, int enabled, int c)
|
|
{
|
|
const int bypass = bypass_channel(s, frame, c);
|
|
const double *src_ptr = (const double *)in->extended_data[c];
|
|
double *dst_ptr = (double *)frame->extended_data[c];
|
|
double current_amplification_factor;
|
|
|
|
cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor);
|
|
|
|
for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) {
|
|
const double amplification_factor = fade(s->prev_amplification_factor[c],
|
|
current_amplification_factor, i,
|
|
frame->nb_samples);
|
|
|
|
dst_ptr[i] = src_ptr[i] * amplification_factor;
|
|
}
|
|
|
|
s->prev_amplification_factor[c] = current_amplification_factor;
|
|
}
|
|
|
|
static int amplify_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
|
|
{
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
ThreadData *td = arg;
|
|
AVFrame *out = td->out;
|
|
AVFrame *in = td->in;
|
|
const int enabled = td->enabled;
|
|
const int channels = s->channels;
|
|
const int start = (channels * jobnr) / nb_jobs;
|
|
const int end = (channels * (jobnr+1)) / nb_jobs;
|
|
|
|
for (int ch = start; ch < end; ch++)
|
|
amplify_channel(s, in, out, enabled, ch);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
ThreadData td;
|
|
int ret;
|
|
|
|
while (((s->queue.available >= s->filter_size) ||
|
|
(s->eof && s->queue.available)) &&
|
|
!cqueue_empty(s->gain_history_smoothed[0])) {
|
|
AVFrame *in = ff_bufqueue_get(&s->queue);
|
|
AVFrame *out;
|
|
double is_enabled;
|
|
|
|
cqueue_dequeue(s->is_enabled, &is_enabled);
|
|
|
|
if (av_frame_is_writable(in)) {
|
|
out = in;
|
|
} else {
|
|
out = ff_get_audio_buffer(outlink, in->nb_samples);
|
|
if (!out) {
|
|
av_frame_free(&in);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
av_frame_copy_props(out, in);
|
|
}
|
|
|
|
td.in = in;
|
|
td.out = out;
|
|
td.enabled = is_enabled > 0.;
|
|
ff_filter_execute(ctx, amplify_channels, &td, NULL,
|
|
FFMIN(s->channels, ff_filter_get_nb_threads(ctx)));
|
|
|
|
s->pts = out->pts + av_rescale_q(out->nb_samples, av_make_q(1, outlink->sample_rate),
|
|
outlink->time_base);
|
|
if (out != in)
|
|
av_frame_free(&in);
|
|
ret = ff_filter_frame(outlink, out);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
ret = analyze_frame(ctx, outlink, &in);
|
|
if (ret < 0)
|
|
return ret;
|
|
if (!s->eof) {
|
|
ff_bufqueue_add(ctx, &s->queue, in);
|
|
cqueue_enqueue(s->is_enabled, !ctx->is_disabled);
|
|
} else {
|
|
av_frame_free(&in);
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
|
|
AVFilterLink *outlink)
|
|
{
|
|
AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance);
|
|
|
|
if (!out)
|
|
return AVERROR(ENOMEM);
|
|
|
|
for (int c = 0; c < s->channels; c++) {
|
|
double *dst_ptr = (double *)out->extended_data[c];
|
|
|
|
for (int i = 0; i < out->nb_samples; i++) {
|
|
dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(s->peak_value, s->target_rms) : s->peak_value);
|
|
if (s->dc_correction) {
|
|
dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
|
|
dst_ptr[i] += s->dc_correction_value[c];
|
|
}
|
|
}
|
|
}
|
|
|
|
return filter_frame(inlink, out);
|
|
}
|
|
|
|
static int flush(AVFilterLink *outlink)
|
|
{
|
|
AVFilterContext *ctx = outlink->src;
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
|
|
while (s->eof && cqueue_empty(s->gain_history_smoothed[0])) {
|
|
for (int c = 0; c < s->channels; c++)
|
|
update_gain_history(s, c, (local_gain){ cqueue_peek(s->gain_history_original[c], 0), 1.0});
|
|
}
|
|
|
|
return flush_buffer(s, inlink, outlink);
|
|
}
|
|
|
|
static int activate(AVFilterContext *ctx)
|
|
{
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
AVFrame *in = NULL;
|
|
int ret = 0, status;
|
|
int64_t pts;
|
|
|
|
ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout);
|
|
if (ret < 0)
|
|
return ret;
|
|
if (strcmp(s->channels_to_filter, "all"))
|
|
av_channel_layout_from_string(&s->ch_layout, s->channels_to_filter);
|
|
|
|
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
|
|
|
|
if (!s->eof) {
|
|
ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
|
|
if (ret < 0)
|
|
return ret;
|
|
if (ret > 0) {
|
|
ret = filter_frame(inlink, in);
|
|
if (ret <= 0)
|
|
return ret;
|
|
}
|
|
|
|
if (ff_inlink_check_available_samples(inlink, s->sample_advance) > 0) {
|
|
ff_filter_set_ready(ctx, 10);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
|
|
if (status == AVERROR_EOF)
|
|
s->eof = 1;
|
|
}
|
|
|
|
if (s->eof && s->queue.available)
|
|
return flush(outlink);
|
|
|
|
if (s->eof && !s->queue.available) {
|
|
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
|
|
return 0;
|
|
}
|
|
|
|
if (!s->eof)
|
|
FF_FILTER_FORWARD_WANTED(outlink, inlink);
|
|
|
|
return FFERROR_NOT_READY;
|
|
}
|
|
|
|
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
|
|
char *res, int res_len, int flags)
|
|
{
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
int prev_filter_size = s->filter_size;
|
|
int ret;
|
|
|
|
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
s->filter_size |= 1;
|
|
if (prev_filter_size != s->filter_size) {
|
|
init_gaussian_filter(s);
|
|
|
|
for (int c = 0; c < s->channels; c++) {
|
|
cqueue_resize(s->gain_history_original[c], s->filter_size);
|
|
cqueue_resize(s->gain_history_minimum[c], s->filter_size);
|
|
cqueue_resize(s->threshold_history[c], s->filter_size);
|
|
}
|
|
}
|
|
|
|
s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec);
|
|
s->sample_advance = FFMAX(1, lrint(s->frame_len * (1. - s->overlap)));
|
|
if (s->expr_str) {
|
|
ret = av_expr_parse(&s->expr, s->expr_str, var_names, NULL, NULL,
|
|
NULL, NULL, 0, ctx);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_input,
|
|
},
|
|
};
|
|
|
|
const AVFilter ff_af_dynaudnorm = {
|
|
.name = "dynaudnorm",
|
|
.description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
|
|
.priv_size = sizeof(DynamicAudioNormalizerContext),
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.activate = activate,
|
|
FILTER_INPUTS(avfilter_af_dynaudnorm_inputs),
|
|
FILTER_OUTPUTS(ff_audio_default_filterpad),
|
|
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
|
|
.priv_class = &dynaudnorm_class,
|
|
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
|
|
AVFILTER_FLAG_SLICE_THREADS,
|
|
.process_command = process_command,
|
|
};
|