libavfilter/af_dynaudnorm.c - nest-android-app/ffmpeg - Git at Google

 /*
  * Dynamic Audio Normalizer
  * Copyright (c) 2015 LoRd_MuldeR <mulder2@gmx.de>. Some rights reserved.
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 /**
  * @file
  * Dynamic Audio Normalizer
  */

 #include <float.h>

 #include "libavutil/avassert.h"
 #include "libavutil/opt.h"

 #define FF_BUFQUEUE_SIZE 302
 #include "libavfilter/bufferqueue.h"

 #include "audio.h"
 #include "avfilter.h"
 #include "internal.h"

 typedef struct cqueue {
     double *elements;
     int size;
     int nb_elements;
     int first;
 } cqueue;

 typedef struct DynamicAudioNormalizerContext {
     const AVClass *class;

     struct FFBufQueue queue;

     int frame_len;
     int frame_len_msec;
     int filter_size;
     int dc_correction;
     int channels_coupled;
     int alt_boundary_mode;

     double peak_value;
     double max_amplification;
     double target_rms;
     double compress_factor;
     double *prev_amplification_factor;
     double *dc_correction_value;
     double *compress_threshold;
     double *fade_factors[2];
     double *weights;

     int channels;
     int delay;

     cqueue **gain_history_original;
     cqueue **gain_history_minimum;
     cqueue **gain_history_smoothed;
 } DynamicAudioNormalizerContext;

 #define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption dynaudnorm_options[] = {
     { "f", "set the frame length in msec",     OFFSET(frame_len_msec),    AV_OPT_TYPE_INT,    {.i64 = 500},   10,  8000, FLAGS },
     { "g", "set the filter size",              OFFSET(filter_size),       AV_OPT_TYPE_INT,    {.i64 = 31},     3,   301, FLAGS },
     { "p", "set the peak value",               OFFSET(peak_value),        AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0,   1.0, FLAGS },
     { "m", "set the max amplification",        OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
     { "r", "set the target RMS",               OFFSET(target_rms),        AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,   1.0, FLAGS },
     { "n", "set channel coupling",             OFFSET(channels_coupled),  AV_OPT_TYPE_BOOL,   {.i64 = 1},      0,     1, FLAGS },
     { "c", "set DC correction",                OFFSET(dc_correction),     AV_OPT_TYPE_BOOL,   {.i64 = 0},      0,     1, FLAGS },
     { "b", "set alternative boundary mode",    OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL,   {.i64 = 0},      0,     1, FLAGS },
     { "s", "set the compress factor",          OFFSET(compress_factor),   AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,  30.0, FLAGS },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(dynaudnorm);

 static av_cold int init(AVFilterContext *ctx)
 {
     DynamicAudioNormalizerContext *s = ctx->priv;

     if (!(s->filter_size & 1)) {
         av_log(ctx, AV_LOG_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
         return AVERROR(EINVAL);
     }

     return 0;
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_DBLP,
         AV_SAMPLE_FMT_NONE
     };
     int ret;

     layouts = ff_all_channel_counts();
     if (!layouts)
         return AVERROR(ENOMEM);
     ret = ff_set_common_channel_layouts(ctx, layouts);
     if (ret < 0)
         return ret;

     formats = ff_make_format_list(sample_fmts);
     if (!formats)
         return AVERROR(ENOMEM);
     ret = ff_set_common_formats(ctx, formats);
     if (ret < 0)
         return ret;

     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
     return ff_set_common_samplerates(ctx, formats);
 }

 static inline int frame_size(int sample_rate, int frame_len_msec)
 {
     const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0));
     return frame_size + (frame_size % 2);
 }

 static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
 {
     const double step_size = 1.0 / frame_len;
     int pos;

     for (pos = 0; pos < frame_len; pos++) {
         fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
         fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
     }
 }

 static cqueue *cqueue_create(int size)
 {
     cqueue *q;

     q = av_malloc(sizeof(cqueue));
     if (!q)
         return NULL;

     q->size = size;
     q->nb_elements = 0;
     q->first = 0;

     q->elements = av_malloc_array(size, sizeof(double));
     if (!q->elements) {
         av_free(q);
         return NULL;
     }

     return q;
 }

 static void cqueue_free(cqueue *q)
 {
     if (q)
         av_free(q->elements);
     av_free(q);
 }

 static int cqueue_size(cqueue *q)
 {
     return q->nb_elements;
 }

 static int cqueue_empty(cqueue *q)
 {
     return !q->nb_elements;
 }

 static int cqueue_enqueue(cqueue *q, double element)
 {
     int i;

     av_assert2(q->nb_elements != q->size);

     i = (q->first + q->nb_elements) % q->size;
     q->elements[i] = element;
     q->nb_elements++;

     return 0;
 }

 static double cqueue_peek(cqueue *q, int index)
 {
     av_assert2(index < q->nb_elements);
     return q->elements[(q->first + index) % q->size];
 }

 static int cqueue_dequeue(cqueue *q, double *element)
 {
     av_assert2(!cqueue_empty(q));

     *element = q->elements[q->first];
     q->first = (q->first + 1) % q->size;
     q->nb_elements--;

     return 0;
 }

 static int cqueue_pop(cqueue *q)
 {
     av_assert2(!cqueue_empty(q));

     q->first = (q->first + 1) % q->size;
     q->nb_elements--;

     return 0;
 }

 static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
 {
     double total_weight = 0.0;
     const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
     double adjust;
     int i;

     // Pre-compute constants
     const int offset = s->filter_size / 2;
     const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
     const double c2 = 2.0 * sigma * sigma;

     // Compute weights
     for (i = 0; i < s->filter_size; i++) {
         const int x = i - offset;

         s->weights[i] = c1 * exp(-x * x / c2);
         total_weight += s->weights[i];
     }

     // Adjust weights
     adjust = 1.0 / total_weight;
     for (i = 0; i < s->filter_size; i++) {
         s->weights[i] *= adjust;
     }
 }

 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++) {
         if (s->gain_history_original)
             cqueue_free(s->gain_history_original[c]);
         if (s->gain_history_minimum)
             cqueue_free(s->gain_history_minimum[c]);
         if (s->gain_history_smoothed)
             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 int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     DynamicAudioNormalizerContext *s = ctx->priv;
     int c;

     uninit(ctx);

     s->frame_len =
     inlink->min_samples =
     inlink->max_samples =
     inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec);
     av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);

     s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
     s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));

     s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor));
     s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
     s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
     s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
     s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
     s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
     s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
     if (!s->prev_amplification_factor || !s->dc_correction_value ||
         !s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
         !s->gain_history_original || !s->gain_history_minimum ||
         !s->gain_history_smoothed || !s->weights)
         return AVERROR(ENOMEM);

     for (c = 0; c < inlink->channels; c++) {
         s->prev_amplification_factor[c] = 1.0;

         s->gain_history_original[c] = cqueue_create(s->filter_size);
         s->gain_history_minimum[c]  = cqueue_create(s->filter_size);
         s->gain_history_smoothed[c] = cqueue_create(s->filter_size);

         if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
             !s->gain_history_smoothed[c])
             return AVERROR(ENOMEM);
     }

     precalculate_fade_factors(s->fade_factors, s->frame_len);
     init_gaussian_filter(s);

     s->channels = inlink->channels;
     s->delay = s->filter_size;

     return 0;
 }

 static inline double fade(double prev, double next, int pos,
                           double *fade_factors[2])
 {
     return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
 }

 static inline double pow2(const double value)
 {
     return value * value;
 }

 static inline double bound(const double threshold, const double val)
 {
     const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
     return erf(CONST * (val / threshold)) * threshold;
 }

 static double find_peak_magnitude(AVFrame *frame, int channel)
 {
     double max = DBL_EPSILON;
     int c, i;

     if (channel == -1) {
         for (c = 0; c < av_frame_get_channels(frame); c++) {
             double *data_ptr = (double *)frame->extended_data[c];

             for (i = 0; i < frame->nb_samples; i++)
                 max = FFMAX(max, fabs(data_ptr[i]));
         }
     } else {
         double *data_ptr = (double *)frame->extended_data[channel];

         for (i = 0; i < frame->nb_samples; i++)
             max = FFMAX(max, fabs(data_ptr[i]));
     }

     return max;
 }

 static double compute_frame_rms(AVFrame *frame, int channel)
 {
     double rms_value = 0.0;
     int c, i;

     if (channel == -1) {
         for (c = 0; c < av_frame_get_channels(frame); c++) {
             const double *data_ptr = (double *)frame->extended_data[c];

             for (i = 0; i < frame->nb_samples; i++) {
                 rms_value += pow2(data_ptr[i]);
             }
         }

         rms_value /= frame->nb_samples * av_frame_get_channels(frame);
     } else {
         const double *data_ptr = (double *)frame->extended_data[channel];
         for (i = 0; i < frame->nb_samples; i++) {
             rms_value += pow2(data_ptr[i]);
         }

         rms_value /= frame->nb_samples;
     }

     return FFMAX(sqrt(rms_value), DBL_EPSILON);
 }

 static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
                                  int channel)
 {
     const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
     const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
     return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
 }

 static double minimum_filter(cqueue *q)
 {
     double min = DBL_MAX;
     int i;

     for (i = 0; i < cqueue_size(q); i++) {
         min = FFMIN(min, cqueue_peek(q, i));
     }

     return min;
 }

 static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
 {
     double result = 0.0;
     int i;

     for (i = 0; i < cqueue_size(q); i++) {
         result += cqueue_peek(q, i) * s->weights[i];
     }

     return result;
 }

 static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
                                 double current_gain_factor)
 {
     if (cqueue_empty(s->gain_history_original[channel]) ||
         cqueue_empty(s->gain_history_minimum[channel])) {
         const int pre_fill_size = s->filter_size / 2;

         s->prev_amplification_factor[channel] = s->alt_boundary_mode ? current_gain_factor : 1.0;

         while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
             cqueue_enqueue(s->gain_history_original[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
         }

         while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
             cqueue_enqueue(s->gain_history_minimum[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
         }
     }

     cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);

     while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
         double minimum;
         av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
         minimum = minimum_filter(s->gain_history_original[channel]);

         cqueue_enqueue(s->gain_history_minimum[channel], minimum);

         cqueue_pop(s->gain_history_original[channel]);
     }

     while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
         double smoothed;
         av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
         smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);

         cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);

         cqueue_pop(s->gain_history_minimum[channel]);
     }
 }

 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 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]);
     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 ((current_threshold + step_size > current_threshold) &&
                    (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 += 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];
         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], &current_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 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,
 };
	/*
	* Dynamic Audio Normalizer
	* Copyright (c) 2015 LoRd_MuldeR <mulder2@gmx.de>. Some rights reserved.
	*
	* This file is part of FFmpeg.
	*
	* FFmpeg is free software; you can redistribute it and/or
	* modify it under the terms of the GNU Lesser General Public
	* License as published by the Free Software Foundation; either
	* version 2.1 of the License, or (at your option) any later version.
	*
	* FFmpeg is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	* Lesser General Public License for more details.
	*
	* You should have received a copy of the GNU Lesser General Public
	* License along with FFmpeg; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	/**
	* @file
	* Dynamic Audio Normalizer
	*/

	#include <float.h>

	#include "libavutil/avassert.h"
	#include "libavutil/opt.h"

	#define FF_BUFQUEUE_SIZE 302
	#include "libavfilter/bufferqueue.h"

	#include "audio.h"
	#include "avfilter.h"
	#include "internal.h"

	typedef struct cqueue {
	double *elements;
	int size;
	int nb_elements;
	int first;
	} cqueue;

	typedef struct DynamicAudioNormalizerContext {
	const AVClass *class;

	struct FFBufQueue queue;

	int frame_len;
	int frame_len_msec;
	int filter_size;
	int dc_correction;
	int channels_coupled;
	int alt_boundary_mode;

	double peak_value;
	double max_amplification;
	double target_rms;
	double compress_factor;
	double *prev_amplification_factor;
	double *dc_correction_value;
	double *compress_threshold;
	double *fade_factors[2];
	double *weights;

	int channels;
	int delay;

	cqueue **gain_history_original;
	cqueue **gain_history_minimum;
	cqueue **gain_history_smoothed;
	} DynamicAudioNormalizerContext;

	#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
	#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM

	static const AVOption dynaudnorm_options[] = {
	{ "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS },
	{ "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS },
	{ "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS },
	{ "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
	{ "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS },
	{ "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS },
	{ "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
	{ "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS },
	{ "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(dynaudnorm);

	static av_cold int init(AVFilterContext *ctx)
	{
	DynamicAudioNormalizerContext *s = ctx->priv;

	if (!(s->filter_size & 1)) {
	av_log(ctx, AV_LOG_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
	return AVERROR(EINVAL);
	}

	return 0;
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterFormats *formats;
	AVFilterChannelLayouts *layouts;
	static const enum AVSampleFormat sample_fmts[] = {
	AV_SAMPLE_FMT_DBLP,
	AV_SAMPLE_FMT_NONE
	};
	int ret;

	layouts = ff_all_channel_counts();
	if (!layouts)
	return AVERROR(ENOMEM);
	ret = ff_set_common_channel_layouts(ctx, layouts);
	if (ret < 0)
	return ret;

	formats = ff_make_format_list(sample_fmts);
	if (!formats)
	return AVERROR(ENOMEM);
	ret = ff_set_common_formats(ctx, formats);
	if (ret < 0)
	return ret;

	formats = ff_all_samplerates();
	if (!formats)
	return AVERROR(ENOMEM);
	return ff_set_common_samplerates(ctx, formats);
	}

	static inline int frame_size(int sample_rate, int frame_len_msec)
	{
	const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0));
	return frame_size + (frame_size % 2);
	}

	static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
	{
	const double step_size = 1.0 / frame_len;
	int pos;

	for (pos = 0; pos < frame_len; pos++) {
	fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
	fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
	}
	}

	static cqueue *cqueue_create(int size)
	{
	cqueue *q;

	q = av_malloc(sizeof(cqueue));
	if (!q)
	return NULL;

	q->size = size;
	q->nb_elements = 0;
	q->first = 0;

	q->elements = av_malloc_array(size, sizeof(double));
	if (!q->elements) {
	av_free(q);
	return NULL;
	}

	return q;
	}

	static void cqueue_free(cqueue *q)
	{
	if (q)
	av_free(q->elements);
	av_free(q);
	}

	static int cqueue_size(cqueue *q)
	{
	return q->nb_elements;
	}

	static int cqueue_empty(cqueue *q)
	{
	return !q->nb_elements;
	}

	static int cqueue_enqueue(cqueue *q, double element)
	{
	int i;

	av_assert2(q->nb_elements != q->size);

	i = (q->first + q->nb_elements) % q->size;
	q->elements[i] = element;
	q->nb_elements++;

	return 0;
	}

	static double cqueue_peek(cqueue *q, int index)
	{
	av_assert2(index < q->nb_elements);
	return q->elements[(q->first + index) % q->size];
	}

	static int cqueue_dequeue(cqueue q, double element)
	{
	av_assert2(!cqueue_empty(q));

	*element = q->elements[q->first];
	q->first = (q->first + 1) % q->size;
	q->nb_elements--;

	return 0;
	}

	static int cqueue_pop(cqueue *q)
	{
	av_assert2(!cqueue_empty(q));

	q->first = (q->first + 1) % q->size;
	q->nb_elements--;

	return 0;
	}

	static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
	{
	double total_weight = 0.0;
	const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
	double adjust;
	int i;

	// Pre-compute constants
	const int offset = s->filter_size / 2;
	const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
	const double c2 = 2.0 * sigma * sigma;

	// Compute weights
	for (i = 0; i < s->filter_size; i++) {
	const int x = i - offset;

	s->weights[i] = c1 * exp(-x * x / c2);
	total_weight += s->weights[i];
	}

	// Adjust weights
	adjust = 1.0 / total_weight;
	for (i = 0; i < s->filter_size; i++) {
	s->weights[i] *= adjust;
	}
	}

	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++) {
	if (s->gain_history_original)
	cqueue_free(s->gain_history_original[c]);
	if (s->gain_history_minimum)
	cqueue_free(s->gain_history_minimum[c]);
	if (s->gain_history_smoothed)
	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 int config_input(AVFilterLink *inlink)
	{
	AVFilterContext *ctx = inlink->dst;
	DynamicAudioNormalizerContext *s = ctx->priv;
	int c;

	uninit(ctx);

	s->frame_len =
	inlink->min_samples =
	inlink->max_samples =
	inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec);
	av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);

	s->fade_factors[0] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[0]));
	s->fade_factors[1] = av_malloc_array(s->frame_len, sizeof(*s->fade_factors[1]));

	s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor));
	s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
	s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
	s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
	s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
	s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
	s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
	if (!s->prev_amplification_factor \|\| !s->dc_correction_value \|\|
	!s->compress_threshold \|\| !s->fade_factors[0] \|\| !s->fade_factors[1] \|\|
	!s->gain_history_original \|\| !s->gain_history_minimum \|\|
	!s->gain_history_smoothed \|\| !s->weights)
	return AVERROR(ENOMEM);

	for (c = 0; c < inlink->channels; c++) {
	s->prev_amplification_factor[c] = 1.0;

	s->gain_history_original[c] = cqueue_create(s->filter_size);
	s->gain_history_minimum[c] = cqueue_create(s->filter_size);
	s->gain_history_smoothed[c] = cqueue_create(s->filter_size);

	if (!s->gain_history_original[c] \|\| !s->gain_history_minimum[c] \|\|
	!s->gain_history_smoothed[c])
	return AVERROR(ENOMEM);
	}

	precalculate_fade_factors(s->fade_factors, s->frame_len);
	init_gaussian_filter(s);

	s->channels = inlink->channels;
	s->delay = s->filter_size;

	return 0;
	}

	static inline double fade(double prev, double next, int pos,
	double *fade_factors[2])
	{
	return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
	}

	static inline double pow2(const double value)
	{
	return value * value;
	}

	static inline double bound(const double threshold, const double val)
	{
	const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
	return erf(CONST * (val / threshold)) * threshold;
	}

	static double find_peak_magnitude(AVFrame *frame, int channel)
	{
	double max = DBL_EPSILON;
	int c, i;

	if (channel == -1) {
	for (c = 0; c < av_frame_get_channels(frame); c++) {
	double data_ptr = (double )frame->extended_data[c];

	for (i = 0; i < frame->nb_samples; i++)
	max = FFMAX(max, fabs(data_ptr[i]));
	}
	} else {
	double data_ptr = (double )frame->extended_data[channel];

	for (i = 0; i < frame->nb_samples; i++)
	max = FFMAX(max, fabs(data_ptr[i]));
	}

	return max;
	}

	static double compute_frame_rms(AVFrame *frame, int channel)
	{
	double rms_value = 0.0;
	int c, i;

	if (channel == -1) {
	for (c = 0; c < av_frame_get_channels(frame); c++) {
	const double data_ptr = (double )frame->extended_data[c];

	for (i = 0; i < frame->nb_samples; i++) {
	rms_value += pow2(data_ptr[i]);
	}
	}

	rms_value /= frame->nb_samples * av_frame_get_channels(frame);
	} else {
	const double data_ptr = (double )frame->extended_data[channel];
	for (i = 0; i < frame->nb_samples; i++) {
	rms_value += pow2(data_ptr[i]);
	}

	rms_value /= frame->nb_samples;
	}

	return FFMAX(sqrt(rms_value), DBL_EPSILON);
	}

	static double get_max_local_gain(DynamicAudioNormalizerContext s, AVFrame frame,
	int channel)
	{
	const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
	const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
	return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
	}

	static double minimum_filter(cqueue *q)
	{
	double min = DBL_MAX;
	int i;

	for (i = 0; i < cqueue_size(q); i++) {
	min = FFMIN(min, cqueue_peek(q, i));
	}

	return min;
	}

	static double gaussian_filter(DynamicAudioNormalizerContext s, cqueue q)
	{
	double result = 0.0;
	int i;

	for (i = 0; i < cqueue_size(q); i++) {
	result += cqueue_peek(q, i) * s->weights[i];
	}

	return result;
	}

	static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
	double current_gain_factor)
	{
	if (cqueue_empty(s->gain_history_original[channel]) \|\|
	cqueue_empty(s->gain_history_minimum[channel])) {
	const int pre_fill_size = s->filter_size / 2;

	s->prev_amplification_factor[channel] = s->alt_boundary_mode ? current_gain_factor : 1.0;

	while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
	cqueue_enqueue(s->gain_history_original[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
	}

	while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
	cqueue_enqueue(s->gain_history_minimum[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
	}
	}

	cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);

	while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
	double minimum;
	av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
	minimum = minimum_filter(s->gain_history_original[channel]);

	cqueue_enqueue(s->gain_history_minimum[channel], minimum);

	cqueue_pop(s->gain_history_original[channel]);
	}

	while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
	double smoothed;
	av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
	smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);

	cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);

	cqueue_pop(s->gain_history_minimum[channel]);
	}
	}

	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 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]);
	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 ((current_threshold + step_size > current_threshold) &&
	(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 += 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];
	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], &current_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 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,
	};