mirror of https://github.com/FFmpeg/FFmpeg.git
LoRd_MuldeR
10 years ago
committed by
Paul B Mahol
parent
3b365dda5c
commit
21436b95dc
4 changed files with 894 additions and 0 deletions
@ -0,0 +1,734 @@ |
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/*
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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", "enable channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, FLAGS }, |
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{ "c", "enable DC correction", OFFSET(dc_correction), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, FLAGS }, |
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{ "b", "enable alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_INT, {.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_layouts(); |
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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 = round((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(sizeof(double) * size); |
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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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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 const double s_pi = 3.1415926535897932384626433832795028841971693993751058209749445923078164062862089986280348253421170679; |
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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 * s_pi)); |
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const double c2 = 2.0 * pow(sigma, 2.0); |
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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(-(pow(x, 2.0) / 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 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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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(s->frame_len * sizeof(*s->fade_factors[0])); |
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s->fade_factors[1] = av_malloc(s->frame_len * sizeof(*s->fade_factors[1])); |
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s->prev_amplification_factor = av_malloc(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(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 int config_output(AVFilterLink *outlink) |
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{ |
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outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP; |
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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 pow2(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 += pow2(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 += pow2(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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s->prev_amplification_factor[channel] = s->alt_boundary_mode ? current_gain_factor : 1.0; |
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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], s->alt_boundary_mode ? current_gain_factor : 1.0); |
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} |
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while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) { |
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cqueue_enqueue(s->gain_history_minimum[channel], s->alt_boundary_mode ? current_gain_factor : 1.0); |
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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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av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size); |
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const double 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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av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size); |
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const double 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; |
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for (c = 0; c < s->channels; c++) { |
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double *dst_ptr = (double *)frame->extended_data[c]; |
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double current_average_value = 0.0; |
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for (i = 0; i < frame->nb_samples; i++) |
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current_average_value += dst_ptr[i] * diff; |
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const double prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c]; |
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s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1); |
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for (i = 0; i < frame->nb_samples; i++) { |
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dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors); |
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} |
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} |
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} |
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static double setup_compress_thresh(double threshold) |
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{ |
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if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) { |
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double current_threshold = threshold; |
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double step_size = 1.0; |
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while (step_size > DBL_EPSILON) { |
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while ((current_threshold + step_size > current_threshold) && |
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(bound(current_threshold + step_size, 1.0) <= threshold)) { |
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current_threshold += step_size; |
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} |
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step_size /= 2.0; |
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} |
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return current_threshold; |
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} else { |
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return threshold; |
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} |
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} |
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static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, |
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AVFrame *frame, int channel) |
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{ |
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double variance = 0.0; |
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int i, c; |
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if (channel == -1) { |
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for (c = 0; c < s->channels; c++) { |
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const double *data_ptr = (double *)frame->extended_data[c]; |
||||
|
||||
for (i = 0; i < frame->nb_samples; i++) { |
||||
variance += pow2(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 += pow2(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]; |
||||
s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0)); |
||||
|
||||
const double prev_actual_thresh = setup_compress_thresh(prev_value); |
||||
const double 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]; |
||||
s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0); |
||||
|
||||
const double prev_actual_thresh = setup_compress_thresh(prev_value); |
||||
const double curr_actual_thresh = setup_compress_thresh(s->compress_threshold[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]); |
||||
} |
||||
} |
||||
} |
||||
} |
||||
|
||||
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) |
||||
ret = flush_buffer(s, ctx->inputs[0], outlink); |
||||
|
||||
return ret; |
||||
} |
||||
|
||||
static av_cold void uninit(AVFilterContext *ctx) |
||||
{ |
||||
DynamicAudioNormalizerContext *s = ctx->priv; |
||||
int c; |
||||
|
||||
av_freep(&s->prev_amplification_factor); |
||||
av_freep(&s->dc_correction_value); |
||||
av_freep(&s->compress_threshold); |
||||
av_freep(&s->fade_factors[0]); |
||||
av_freep(&s->fade_factors[1]); |
||||
|
||||
for (c = 0; c < s->channels; c++) { |
||||
cqueue_free(s->gain_history_original[c]); |
||||
cqueue_free(s->gain_history_minimum[c]); |
||||
cqueue_free(s->gain_history_smoothed[c]); |
||||
} |
||||
|
||||
av_freep(&s->gain_history_original); |
||||
av_freep(&s->gain_history_minimum); |
||||
av_freep(&s->gain_history_smoothed); |
||||
|
||||
av_freep(&s->weights); |
||||
|
||||
ff_bufqueue_discard_all(&s->queue); |
||||
} |
||||
|
||||
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, |
||||
.config_props = config_output, |
||||
.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, |
||||
}; |
||||
Loading…
Reference in new issue