mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-25 19:01:51 +02:00
755 lines
23 KiB
C
755 lines
23 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/opt.h"
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#define FF_BUFQUEUE_SIZE 302
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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 "internal.h"
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typedef struct cqueue {
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double *elements;
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int size;
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int nb_elements;
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int first;
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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 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 *prev_amplification_factor;
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double *dc_correction_value;
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double *compress_threshold;
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double *fade_factors[2];
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double *weights;
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int channels;
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int delay;
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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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} DynamicAudioNormalizerContext;
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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
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static const AVOption dynaudnorm_options[] = {
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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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{ "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, 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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{ "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, 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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{ "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 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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{ "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, 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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{ 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_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
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return AVERROR(EINVAL);
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}
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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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 void precalculate_fade_factors(double *fade_factors[2], int frame_len)
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{
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const double step_size = 1.0 / frame_len;
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int pos;
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for (pos = 0; pos < frame_len; pos++) {
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fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
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fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
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}
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}
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static cqueue *cqueue_create(int size)
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{
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cqueue *q;
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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->size = size;
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q->nb_elements = 0;
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q->first = 0;
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q->elements = av_malloc_array(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;
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}
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static int cqueue_enqueue(cqueue *q, double element)
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{
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int i;
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av_assert2(q->nb_elements != q->size);
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i = (q->first + q->nb_elements) % q->size;
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q->elements[i] = 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[(q->first + index) % q->size];
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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[q->first];
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q->first = (q->first + 1) % q->size;
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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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q->first = (q->first + 1) % q->size;
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q->nb_elements--;
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return 0;
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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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int i;
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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 (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 (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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int c;
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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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av_freep(&s->fade_factors[0]);
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av_freep(&s->fade_factors[1]);
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for (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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}
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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->weights);
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ff_bufqueue_discard_all(&s->queue);
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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 c;
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uninit(ctx);
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s->frame_len =
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inlink->min_samples =
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inlink->max_samples =
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inlink->partial_buf_size = 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->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
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s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));
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s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor));
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s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
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s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
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s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
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s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
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s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
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s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
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if (!s->prev_amplification_factor || !s->dc_correction_value ||
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!s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
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!s->gain_history_original || !s->gain_history_minimum ||
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!s->gain_history_smoothed || !s->weights)
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return AVERROR(ENOMEM);
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for (c = 0; c < inlink->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);
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s->gain_history_minimum[c] = cqueue_create(s->filter_size);
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s->gain_history_smoothed[c] = cqueue_create(s->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])
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return AVERROR(ENOMEM);
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}
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precalculate_fade_factors(s->fade_factors, s->frame_len);
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init_gaussian_filter(s);
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s->channels = inlink->channels;
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s->delay = s->filter_size;
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return 0;
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}
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static inline double fade(double prev, double next, int pos,
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double *fade_factors[2])
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{
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return fade_factors[0][pos] * prev + fade_factors[1][pos] * 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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int c, i;
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if (channel == -1) {
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for (c = 0; c < frame->channels; c++) {
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double *data_ptr = (double *)frame->extended_data[c];
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for (i = 0; i < frame->nb_samples; i++)
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max = FFMAX(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 (i = 0; i < frame->nb_samples; i++)
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max = FFMAX(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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int c, i;
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if (channel == -1) {
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for (c = 0; c < frame->channels; c++) {
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const double *data_ptr = (double *)frame->extended_data[c];
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for (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->channels;
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} else {
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const double *data_ptr = (double *)frame->extended_data[channel];
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for (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 FFMAX(sqrt(rms_value), DBL_EPSILON);
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}
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static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
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int channel)
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{
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const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
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const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
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return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
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}
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static double minimum_filter(cqueue *q)
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{
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double min = DBL_MAX;
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int i;
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for (i = 0; i < cqueue_size(q); i++) {
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min = FFMIN(min, cqueue_peek(q, i));
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}
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return min;
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}
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static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
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{
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double result = 0.0;
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int i;
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for (i = 0; i < cqueue_size(q); i++) {
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result += cqueue_peek(q, i) * s->weights[i];
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}
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return result;
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}
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static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
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double current_gain_factor)
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{
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if (cqueue_empty(s->gain_history_original[channel]) ||
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cqueue_empty(s->gain_history_minimum[channel])) {
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const int pre_fill_size = s->filter_size / 2;
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const double initial_value = s->alt_boundary_mode ? current_gain_factor : 1.0;
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s->prev_amplification_factor[channel] = initial_value;
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while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
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cqueue_enqueue(s->gain_history_original[channel], initial_value);
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}
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}
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cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
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while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
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double minimum;
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av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
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if (cqueue_empty(s->gain_history_minimum[channel])) {
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const int pre_fill_size = s->filter_size / 2;
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double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0;
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int input = pre_fill_size;
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while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
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initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], ++input));
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cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
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}
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}
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minimum = minimum_filter(s->gain_history_original[channel]);
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cqueue_enqueue(s->gain_history_minimum[channel], minimum);
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cqueue_pop(s->gain_history_original[channel]);
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}
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while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
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double smoothed;
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av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
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smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);
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cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
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cqueue_pop(s->gain_history_minimum[channel]);
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}
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}
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static inline double update_value(double new, double old, double aggressiveness)
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{
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av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
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return aggressiveness * new + (1.0 - aggressiveness) * old;
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}
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static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
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{
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const double diff = 1.0 / frame->nb_samples;
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int is_first_frame = cqueue_empty(s->gain_history_original[0]);
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|
int c, i;
|
|
|
|
for (c = 0; c < s->channels; c++) {
|
|
double *dst_ptr = (double *)frame->extended_data[c];
|
|
double current_average_value = 0.0;
|
|
double prev_value;
|
|
|
|
for (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 (i = 0; i < frame->nb_samples; i++) {
|
|
dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors);
|
|
}
|
|
}
|
|
}
|
|
|
|
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;
|
|
int i, c;
|
|
|
|
if (channel == -1) {
|
|
for (c = 0; c < s->channels; c++) {
|
|
const double *data_ptr = (double *)frame->extended_data[c];
|
|
|
|
for (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 (i = 0; i < frame->nb_samples; i++) {
|
|
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
|
|
}
|
|
variance /= frame->nb_samples - 1;
|
|
}
|
|
|
|
return FFMAX(sqrt(variance), DBL_EPSILON);
|
|
}
|
|
|
|
static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
|
|
{
|
|
int is_first_frame = cqueue_empty(s->gain_history_original[0]);
|
|
int c, i;
|
|
|
|
if (s->channels_coupled) {
|
|
const double standard_deviation = compute_frame_std_dev(s, frame, -1);
|
|
const double current_threshold = FFMIN(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 (c = 0; c < s->channels; c++) {
|
|
double *const dst_ptr = (double *)frame->extended_data[c];
|
|
for (i = 0; i < frame->nb_samples; i++) {
|
|
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
|
|
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
|
|
}
|
|
}
|
|
} else {
|
|
for (c = 0; c < s->channels; c++) {
|
|
const double standard_deviation = compute_frame_std_dev(s, frame, c);
|
|
const double current_threshold = setup_compress_thresh(FFMIN(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 (i = 0; i < frame->nb_samples; i++) {
|
|
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
|
|
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
|
|
{
|
|
if (s->dc_correction) {
|
|
perform_dc_correction(s, frame);
|
|
}
|
|
|
|
if (s->compress_factor > DBL_EPSILON) {
|
|
perform_compression(s, frame);
|
|
}
|
|
|
|
if (s->channels_coupled) {
|
|
const double current_gain_factor = get_max_local_gain(s, frame, -1);
|
|
int c;
|
|
|
|
for (c = 0; c < s->channels; c++)
|
|
update_gain_history(s, c, current_gain_factor);
|
|
} else {
|
|
int c;
|
|
|
|
for (c = 0; c < s->channels; c++)
|
|
update_gain_history(s, c, get_max_local_gain(s, frame, c));
|
|
}
|
|
}
|
|
|
|
static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
|
|
{
|
|
int c, i;
|
|
|
|
for (c = 0; c < s->channels; c++) {
|
|
double *dst_ptr = (double *)frame->extended_data[c];
|
|
double current_amplification_factor;
|
|
|
|
cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor);
|
|
|
|
for (i = 0; i < frame->nb_samples; i++) {
|
|
const double amplification_factor = fade(s->prev_amplification_factor[c],
|
|
current_amplification_factor, i,
|
|
s->fade_factors);
|
|
|
|
dst_ptr[i] *= amplification_factor;
|
|
|
|
if (fabs(dst_ptr[i]) > s->peak_value)
|
|
dst_ptr[i] = copysign(s->peak_value, dst_ptr[i]);
|
|
}
|
|
|
|
s->prev_amplification_factor[c] = current_amplification_factor;
|
|
}
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
AVFilterLink *outlink = inlink->dst->outputs[0];
|
|
int ret = 0;
|
|
|
|
if (!cqueue_empty(s->gain_history_smoothed[0])) {
|
|
AVFrame *out = ff_bufqueue_get(&s->queue);
|
|
|
|
amplify_frame(s, out);
|
|
ret = ff_filter_frame(outlink, out);
|
|
}
|
|
|
|
analyze_frame(s, in);
|
|
ff_bufqueue_add(ctx, &s->queue, in);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
|
|
AVFilterLink *outlink)
|
|
{
|
|
AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len);
|
|
int c, i;
|
|
|
|
if (!out)
|
|
return AVERROR(ENOMEM);
|
|
|
|
for (c = 0; c < s->channels; c++) {
|
|
double *dst_ptr = (double *)out->extended_data[c];
|
|
|
|
for (i = 0; i < out->nb_samples; i++) {
|
|
dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(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];
|
|
}
|
|
}
|
|
}
|
|
|
|
s->delay--;
|
|
return filter_frame(inlink, out);
|
|
}
|
|
|
|
static int request_frame(AVFilterLink *outlink)
|
|
{
|
|
AVFilterContext *ctx = outlink->src;
|
|
DynamicAudioNormalizerContext *s = ctx->priv;
|
|
int ret = 0;
|
|
|
|
ret = ff_request_frame(ctx->inputs[0]);
|
|
|
|
if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay) {
|
|
if (!cqueue_empty(s->gain_history_smoothed[0])) {
|
|
ret = flush_buffer(s, ctx->inputs[0], outlink);
|
|
} else if (s->queue.available) {
|
|
AVFrame *out = ff_bufqueue_get(&s->queue);
|
|
|
|
ret = ff_filter_frame(outlink, out);
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
.config_props = config_input,
|
|
.needs_writable = 1,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.request_frame = request_frame,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
AVFilter ff_af_dynaudnorm = {
|
|
.name = "dynaudnorm",
|
|
.description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(DynamicAudioNormalizerContext),
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.inputs = avfilter_af_dynaudnorm_inputs,
|
|
.outputs = avfilter_af_dynaudnorm_outputs,
|
|
.priv_class = &dynaudnorm_class,
|
|
};
|