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  1. /*
  2.  * Dynamic Audio Normalizer
  3.  * Copyright (c) 2015 LoRd_MuldeR <mulder2@gmx.de>. Some rights reserved.
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * Dynamic Audio Normalizer
  25.  */
  26.  
  27. #include <float.h>
  28.  
  29. #include "libavutil/avassert.h"
  30. #include "libavutil/opt.h"
  31.  
  32. #define FF_BUFQUEUE_SIZE 302
  33. #include "libavfilter/bufferqueue.h"
  34.  
  35. #include "audio.h"
  36. #include "avfilter.h"
  37. #include "internal.h"
  38.  
  39. typedef struct cqueue {
  40.     double *elements;
  41.     int size;
  42.     int nb_elements;
  43.     int first;
  44. } cqueue;
  45.  
  46. typedef struct DynamicAudioNormalizerContext {
  47.     const AVClass *class;
  48.  
  49.     struct FFBufQueue queue;
  50.  
  51.     int frame_len;
  52.     int frame_len_msec;
  53.     int filter_size;
  54.     int dc_correction;
  55.     int channels_coupled;
  56.     int alt_boundary_mode;
  57.  
  58.     double peak_value;
  59.     double max_amplification;
  60.     double target_rms;
  61.     double compress_factor;
  62.     double *prev_amplification_factor;
  63.     double *dc_correction_value;
  64.     double *compress_threshold;
  65.     double *fade_factors[2];
  66.     double *weights;
  67.  
  68.     int channels;
  69.     int delay;
  70.  
  71.     cqueue **gain_history_original;
  72.     cqueue **gain_history_minimum;
  73.     cqueue **gain_history_smoothed;
  74. } DynamicAudioNormalizerContext;
  75.  
  76. #define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x)
  77. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  78.  
  79. static const AVOption dynaudnorm_options[] = {
  80.     { "f", "set the frame length in msec",     OFFSET(frame_len_msec),    AV_OPT_TYPE_INT,    {.i64 = 500},   10,  8000, FLAGS },
  81.     { "g", "set the filter size",              OFFSET(filter_size),       AV_OPT_TYPE_INT,    {.i64 = 31},     3,   301, FLAGS },
  82.     { "p", "set the peak value",               OFFSET(peak_value),        AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0,   1.0, FLAGS },
  83.     { "m", "set the max amplification",        OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS },
  84.     { "r", "set the target RMS",               OFFSET(target_rms),        AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,   1.0, FLAGS },
  85.     { "n", "enable channel coupling",          OFFSET(channels_coupled),  AV_OPT_TYPE_INT,    {.i64 = 1},      0,     1, FLAGS },
  86.     { "c", "enable DC correction",             OFFSET(dc_correction),     AV_OPT_TYPE_INT,    {.i64 = 0},      0,     1, FLAGS },
  87.     { "b", "enable alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_INT,    {.i64 = 0},      0,     1, FLAGS },
  88.     { "s", "set the compress factor",          OFFSET(compress_factor),   AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,  30.0, FLAGS },
  89.     { NULL }
  90. };
  91.  
  92. AVFILTER_DEFINE_CLASS(dynaudnorm);
  93.  
  94. static av_cold int init(AVFilterContext *ctx)
  95. {
  96.     DynamicAudioNormalizerContext *s = ctx->priv;
  97.  
  98.     if (!(s->filter_size & 1)) {
  99.         av_log(ctx, AV_LOG_ERROR, "filter size %d is invalid. Must be an odd value.\n", s->filter_size);
  100.         return AVERROR(EINVAL);
  101.     }
  102.  
  103.     return 0;
  104. }
  105.  
  106. static int query_formats(AVFilterContext *ctx)
  107. {
  108.     AVFilterFormats *formats;
  109.     AVFilterChannelLayouts *layouts;
  110.     static const enum AVSampleFormat sample_fmts[] = {
  111.         AV_SAMPLE_FMT_DBLP,
  112.         AV_SAMPLE_FMT_NONE
  113.     };
  114.     int ret;
  115.  
  116.     layouts = ff_all_channel_layouts();
  117.     if (!layouts)
  118.         return AVERROR(ENOMEM);
  119.     ret = ff_set_common_channel_layouts(ctx, layouts);
  120.     if (ret < 0)
  121.         return ret;
  122.  
  123.     formats = ff_make_format_list(sample_fmts);
  124.     if (!formats)
  125.         return AVERROR(ENOMEM);
  126.     ret = ff_set_common_formats(ctx, formats);
  127.     if (ret < 0)
  128.         return ret;
  129.  
  130.     formats = ff_all_samplerates();
  131.     if (!formats)
  132.         return AVERROR(ENOMEM);
  133.     return ff_set_common_samplerates(ctx, formats);
  134. }
  135.  
  136. static inline int frame_size(int sample_rate, int frame_len_msec)
  137. {
  138.     const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0));
  139.     return frame_size + (frame_size % 2);
  140. }
  141.  
  142. static void precalculate_fade_factors(double *fade_factors[2], int frame_len)
  143. {
  144.     const double step_size = 1.0 / frame_len;
  145.     int pos;
  146.  
  147.     for (pos = 0; pos < frame_len; pos++) {
  148.         fade_factors[0][pos] = 1.0 - (step_size * (pos + 1.0));
  149.         fade_factors[1][pos] = 1.0 - fade_factors[0][pos];
  150.     }
  151. }
  152.  
  153. static cqueue *cqueue_create(int size)
  154. {
  155.     cqueue *q;
  156.  
  157.     q = av_malloc(sizeof(cqueue));
  158.     if (!q)
  159.         return NULL;
  160.  
  161.     q->size = size;
  162.     q->nb_elements = 0;
  163.     q->first = 0;
  164.  
  165.     q->elements = av_malloc(sizeof(double) * size);
  166.     if (!q->elements) {
  167.         av_free(q);
  168.         return NULL;
  169.     }
  170.  
  171.     return q;
  172. }
  173.  
  174. static void cqueue_free(cqueue *q)
  175. {
  176.     av_free(q->elements);
  177.     av_free(q);
  178. }
  179.  
  180. static int cqueue_size(cqueue *q)
  181. {
  182.     return q->nb_elements;
  183. }
  184.  
  185. static int cqueue_empty(cqueue *q)
  186. {
  187.     return !q->nb_elements;
  188. }
  189.  
  190. static int cqueue_enqueue(cqueue *q, double element)
  191. {
  192.     int i;
  193.  
  194.     av_assert2(q->nb_elements != q->size);
  195.  
  196.     i = (q->first + q->nb_elements) % q->size;
  197.     q->elements[i] = element;
  198.     q->nb_elements++;
  199.  
  200.     return 0;
  201. }
  202.  
  203. static double cqueue_peek(cqueue *q, int index)
  204. {
  205.     av_assert2(index < q->nb_elements);
  206.     return q->elements[(q->first + index) % q->size];
  207. }
  208.  
  209. static int cqueue_dequeue(cqueue *q, double *element)
  210. {
  211.     av_assert2(!cqueue_empty(q));
  212.  
  213.     *element = q->elements[q->first];
  214.     q->first = (q->first + 1) % q->size;
  215.     q->nb_elements--;
  216.  
  217.     return 0;
  218. }
  219.  
  220. static int cqueue_pop(cqueue *q)
  221. {
  222.     av_assert2(!cqueue_empty(q));
  223.  
  224.     q->first = (q->first + 1) % q->size;
  225.     q->nb_elements--;
  226.  
  227.     return 0;
  228. }
  229.  
  230. static const double s_pi = 3.1415926535897932384626433832795028841971693993751058209749445923078164062862089986280348253421170679;
  231.  
  232. static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
  233. {
  234.     double total_weight = 0.0;
  235.     const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
  236.     double adjust;
  237.     int i;
  238.  
  239.     // Pre-compute constants
  240.     const int offset = s->filter_size / 2;
  241.     const double c1 = 1.0 / (sigma * sqrt(2.0 * s_pi));
  242.     const double c2 = 2.0 * pow(sigma, 2.0);
  243.  
  244.     // Compute weights
  245.     for (i = 0; i < s->filter_size; i++) {
  246.         const int x = i - offset;
  247.  
  248.         s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2));
  249.         total_weight += s->weights[i];
  250.     }
  251.  
  252.     // Adjust weights
  253.     adjust = 1.0 / total_weight;
  254.     for (i = 0; i < s->filter_size; i++) {
  255.         s->weights[i] *= adjust;
  256.     }
  257. }
  258.  
  259. static int config_input(AVFilterLink *inlink)
  260. {
  261.     AVFilterContext *ctx = inlink->dst;
  262.     DynamicAudioNormalizerContext *s = ctx->priv;
  263.     int c;
  264.  
  265.     s->frame_len =
  266.     inlink->min_samples =
  267.     inlink->max_samples =
  268.     inlink->partial_buf_size = frame_size(inlink->sample_rate, s->frame_len_msec);
  269.     av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len);
  270.  
  271.     s->fade_factors[0] = av_malloc(s->frame_len * sizeof(*s->fade_factors[0]));
  272.     s->fade_factors[1] = av_malloc(s->frame_len * sizeof(*s->fade_factors[1]));
  273.  
  274.     s->prev_amplification_factor = av_malloc(inlink->channels * sizeof(*s->prev_amplification_factor));
  275.     s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value));
  276.     s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold));
  277.     s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original));
  278.     s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum));
  279.     s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed));
  280.     s->weights = av_malloc(s->filter_size * sizeof(*s->weights));
  281.     if (!s->prev_amplification_factor || !s->dc_correction_value ||
  282.         !s->compress_threshold || !s->fade_factors[0] || !s->fade_factors[1] ||
  283.         !s->gain_history_original || !s->gain_history_minimum ||
  284.         !s->gain_history_smoothed || !s->weights)
  285.         return AVERROR(ENOMEM);
  286.  
  287.     for (c = 0; c < inlink->channels; c++) {
  288.         s->prev_amplification_factor[c] = 1.0;
  289.  
  290.         s->gain_history_original[c] = cqueue_create(s->filter_size);
  291.         s->gain_history_minimum[c]  = cqueue_create(s->filter_size);
  292.         s->gain_history_smoothed[c] = cqueue_create(s->filter_size);
  293.  
  294.         if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
  295.             !s->gain_history_smoothed[c])
  296.             return AVERROR(ENOMEM);
  297.     }
  298.  
  299.     precalculate_fade_factors(s->fade_factors, s->frame_len);
  300.     init_gaussian_filter(s);
  301.  
  302.     s->channels = inlink->channels;
  303.     s->delay = s->filter_size;
  304.  
  305.     return 0;
  306. }
  307.  
  308. static int config_output(AVFilterLink *outlink)
  309. {
  310.     outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
  311.     return 0;
  312. }
  313.  
  314. static inline double fade(double prev, double next, int pos,
  315.                           double *fade_factors[2])
  316. {
  317.     return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
  318. }
  319.  
  320. static inline double pow2(const double value)
  321. {
  322.     return value * value;
  323. }
  324.  
  325. static inline double bound(const double threshold, const double val)
  326. {
  327.     const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0
  328.     return erf(CONST * (val / threshold)) * threshold;
  329. }
  330.  
  331. static double find_peak_magnitude(AVFrame *frame, int channel)
  332. {
  333.     double max = DBL_EPSILON;
  334.     int c, i;
  335.  
  336.     if (channel == -1) {
  337.         for (c = 0; c < av_frame_get_channels(frame); c++) {
  338.             double *data_ptr = (double *)frame->extended_data[c];
  339.  
  340.             for (i = 0; i < frame->nb_samples; i++)
  341.                 max = FFMAX(max, fabs(data_ptr[i]));
  342.         }
  343.     } else {
  344.         double *data_ptr = (double *)frame->extended_data[channel];
  345.  
  346.         for (i = 0; i < frame->nb_samples; i++)
  347.             max = FFMAX(max, fabs(data_ptr[i]));
  348.     }
  349.  
  350.     return max;
  351. }
  352.  
  353. static double compute_frame_rms(AVFrame *frame, int channel)
  354. {
  355.     double rms_value = 0.0;
  356.     int c, i;
  357.  
  358.     if (channel == -1) {
  359.         for (c = 0; c < av_frame_get_channels(frame); c++) {
  360.             const double *data_ptr = (double *)frame->extended_data[c];
  361.  
  362.             for (i = 0; i < frame->nb_samples; i++) {
  363.                 rms_value += pow2(data_ptr[i]);
  364.             }
  365.         }
  366.  
  367.         rms_value /= frame->nb_samples * av_frame_get_channels(frame);
  368.     } else {
  369.         const double *data_ptr = (double *)frame->extended_data[channel];
  370.         for (i = 0; i < frame->nb_samples; i++) {
  371.             rms_value += pow2(data_ptr[i]);
  372.         }
  373.  
  374.         rms_value /= frame->nb_samples;
  375.     }
  376.  
  377.     return FFMAX(sqrt(rms_value), DBL_EPSILON);
  378. }
  379.  
  380. static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
  381.                                  int channel)
  382. {
  383.     const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
  384.     const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
  385.     return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
  386. }
  387.  
  388. static double minimum_filter(cqueue *q)
  389. {
  390.     double min = DBL_MAX;
  391.     int i;
  392.  
  393.     for (i = 0; i < cqueue_size(q); i++) {
  394.         min = FFMIN(min, cqueue_peek(q, i));
  395.     }
  396.  
  397.     return min;
  398. }
  399.  
  400. static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
  401. {
  402.     double result = 0.0;
  403.     int i;
  404.  
  405.     for (i = 0; i < cqueue_size(q); i++) {
  406.         result += cqueue_peek(q, i) * s->weights[i];
  407.     }
  408.  
  409.     return result;
  410. }
  411.  
  412. static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
  413.                                 double current_gain_factor)
  414. {
  415.     if (cqueue_empty(s->gain_history_original[channel]) ||
  416.         cqueue_empty(s->gain_history_minimum[channel])) {
  417.         const int pre_fill_size = s->filter_size / 2;
  418.  
  419.         s->prev_amplification_factor[channel] = s->alt_boundary_mode ? current_gain_factor : 1.0;
  420.  
  421.         while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
  422.             cqueue_enqueue(s->gain_history_original[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
  423.         }
  424.  
  425.         while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
  426.             cqueue_enqueue(s->gain_history_minimum[channel], s->alt_boundary_mode ? current_gain_factor : 1.0);
  427.         }
  428.     }
  429.  
  430.     cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
  431.  
  432.     while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
  433.         double minimum;
  434.         av_assert0(cqueue_size(s->gain_history_original[channel]) == s->filter_size);
  435.         minimum = minimum_filter(s->gain_history_original[channel]);
  436.  
  437.         cqueue_enqueue(s->gain_history_minimum[channel], minimum);
  438.  
  439.         cqueue_pop(s->gain_history_original[channel]);
  440.     }
  441.  
  442.     while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) {
  443.         double smoothed;
  444.         av_assert0(cqueue_size(s->gain_history_minimum[channel]) == s->filter_size);
  445.         smoothed = gaussian_filter(s, s->gain_history_minimum[channel]);
  446.  
  447.         cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
  448.  
  449.         cqueue_pop(s->gain_history_minimum[channel]);
  450.     }
  451. }
  452.  
  453. static inline double update_value(double new, double old, double aggressiveness)
  454. {
  455.     av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0));
  456.     return aggressiveness * new + (1.0 - aggressiveness) * old;
  457. }
  458.  
  459. static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame)
  460. {
  461.     const double diff = 1.0 / frame->nb_samples;
  462.     int is_first_frame = cqueue_empty(s->gain_history_original[0]);
  463.     int c, i;
  464.  
  465.     for (c = 0; c < s->channels; c++) {
  466.         double *dst_ptr = (double *)frame->extended_data[c];
  467.         double current_average_value = 0.0;
  468.         double prev_value;
  469.  
  470.         for (i = 0; i < frame->nb_samples; i++)
  471.             current_average_value += dst_ptr[i] * diff;
  472.  
  473.         prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
  474.         s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
  475.  
  476.         for (i = 0; i < frame->nb_samples; i++) {
  477.             dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, s->fade_factors);
  478.         }
  479.     }
  480. }
  481.  
  482. static double setup_compress_thresh(double threshold)
  483. {
  484.     if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) {
  485.         double current_threshold = threshold;
  486.         double step_size = 1.0;
  487.  
  488.         while (step_size > DBL_EPSILON) {
  489.             while ((current_threshold + step_size > current_threshold) &&
  490.                    (bound(current_threshold + step_size, 1.0) <= threshold)) {
  491.                 current_threshold += step_size;
  492.             }
  493.  
  494.             step_size /= 2.0;
  495.         }
  496.  
  497.         return current_threshold;
  498.     } else {
  499.         return threshold;
  500.     }
  501. }
  502.  
  503. static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
  504.                                     AVFrame *frame, int channel)
  505. {
  506.     double variance = 0.0;
  507.     int i, c;
  508.  
  509.     if (channel == -1) {
  510.         for (c = 0; c < s->channels; c++) {
  511.             const double *data_ptr = (double *)frame->extended_data[c];
  512.  
  513.             for (i = 0; i < frame->nb_samples; i++) {
  514.                 variance += pow2(data_ptr[i]);  // Assume that MEAN is *zero*
  515.             }
  516.         }
  517.         variance /= (s->channels * frame->nb_samples) - 1;
  518.     } else {
  519.         const double *data_ptr = (double *)frame->extended_data[channel];
  520.  
  521.         for (i = 0; i < frame->nb_samples; i++) {
  522.             variance += pow2(data_ptr[i]);      // Assume that MEAN is *zero*
  523.         }
  524.         variance /= frame->nb_samples - 1;
  525.     }
  526.  
  527.     return FFMAX(sqrt(variance), DBL_EPSILON);
  528. }
  529.  
  530. static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
  531. {
  532.     int is_first_frame = cqueue_empty(s->gain_history_original[0]);
  533.     int c, i;
  534.  
  535.     if (s->channels_coupled) {
  536.         const double standard_deviation = compute_frame_std_dev(s, frame, -1);
  537.         const double current_threshold  = FFMIN(1.0, s->compress_factor * standard_deviation);
  538.  
  539.         const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
  540.         double prev_actual_thresh, curr_actual_thresh;
  541.         s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0));
  542.  
  543.         prev_actual_thresh = setup_compress_thresh(prev_value);
  544.         curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
  545.  
  546.         for (c = 0; c < s->channels; c++) {
  547.             double *const dst_ptr = (double *)frame->extended_data[c];
  548.             for (i = 0; i < frame->nb_samples; i++) {
  549.                 const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
  550.                 dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
  551.             }
  552.         }
  553.     } else {
  554.         for (c = 0; c < s->channels; c++) {
  555.             const double standard_deviation = compute_frame_std_dev(s, frame, c);
  556.             const double current_threshold  = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation));
  557.  
  558.             const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
  559.             double prev_actual_thresh, curr_actual_thresh;
  560.             double *dst_ptr;
  561.             s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0);
  562.  
  563.             prev_actual_thresh = setup_compress_thresh(prev_value);
  564.             curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
  565.  
  566.             dst_ptr = (double *)frame->extended_data[c];
  567.             for (i = 0; i < frame->nb_samples; i++) {
  568.                 const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, s->fade_factors);
  569.                 dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
  570.             }
  571.         }
  572.     }
  573. }
  574.  
  575. static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
  576. {
  577.     if (s->dc_correction) {
  578.         perform_dc_correction(s, frame);
  579.     }
  580.  
  581.     if (s->compress_factor > DBL_EPSILON) {
  582.         perform_compression(s, frame);
  583.     }
  584.  
  585.     if (s->channels_coupled) {
  586.         const double current_gain_factor = get_max_local_gain(s, frame, -1);
  587.         int c;
  588.  
  589.         for (c = 0; c < s->channels; c++)
  590.             update_gain_history(s, c, current_gain_factor);
  591.     } else {
  592.         int c;
  593.  
  594.         for (c = 0; c < s->channels; c++)
  595.             update_gain_history(s, c, get_max_local_gain(s, frame, c));
  596.     }
  597. }
  598.  
  599. static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
  600. {
  601.     int c, i;
  602.  
  603.     for (c = 0; c < s->channels; c++) {
  604.         double *dst_ptr = (double *)frame->extended_data[c];
  605.         double current_amplification_factor;
  606.  
  607.         cqueue_dequeue(s->gain_history_smoothed[c], &current_amplification_factor);
  608.  
  609.         for (i = 0; i < frame->nb_samples; i++) {
  610.             const double amplification_factor = fade(s->prev_amplification_factor[c],
  611.                                                      current_amplification_factor, i,
  612.                                                      s->fade_factors);
  613.  
  614.             dst_ptr[i] *= amplification_factor;
  615.  
  616.             if (fabs(dst_ptr[i]) > s->peak_value)
  617.                 dst_ptr[i] = copysign(s->peak_value, dst_ptr[i]);
  618.         }
  619.  
  620.         s->prev_amplification_factor[c] = current_amplification_factor;
  621.     }
  622. }
  623.  
  624. static int filter_frame(AVFilterLink *inlink, AVFrame *in)
  625. {
  626.     AVFilterContext *ctx = inlink->dst;
  627.     DynamicAudioNormalizerContext *s = ctx->priv;
  628.     AVFilterLink *outlink = inlink->dst->outputs[0];
  629.     int ret = 0;
  630.  
  631.     if (!cqueue_empty(s->gain_history_smoothed[0])) {
  632.         AVFrame *out = ff_bufqueue_get(&s->queue);
  633.  
  634.         amplify_frame(s, out);
  635.         ret = ff_filter_frame(outlink, out);
  636.     }
  637.  
  638.     analyze_frame(s, in);
  639.     ff_bufqueue_add(ctx, &s->queue, in);
  640.  
  641.     return ret;
  642. }
  643.  
  644. static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
  645.                         AVFilterLink *outlink)
  646. {
  647.     AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len);
  648.     int c, i;
  649.  
  650.     if (!out)
  651.         return AVERROR(ENOMEM);
  652.  
  653.     for (c = 0; c < s->channels; c++) {
  654.         double *dst_ptr = (double *)out->extended_data[c];
  655.  
  656.         for (i = 0; i < out->nb_samples; i++) {
  657.             dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(s->peak_value, s->target_rms) : s->peak_value);
  658.             if (s->dc_correction) {
  659.                 dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
  660.                 dst_ptr[i] += s->dc_correction_value[c];
  661.             }
  662.         }
  663.     }
  664.  
  665.     s->delay--;
  666.     return filter_frame(inlink, out);
  667. }
  668.  
  669. static int request_frame(AVFilterLink *outlink)
  670. {
  671.     AVFilterContext *ctx = outlink->src;
  672.     DynamicAudioNormalizerContext *s = ctx->priv;
  673.     int ret = 0;
  674.  
  675.     ret = ff_request_frame(ctx->inputs[0]);
  676.  
  677.     if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay)
  678.         ret = flush_buffer(s, ctx->inputs[0], outlink);
  679.  
  680.     return ret;
  681. }
  682.  
  683. static av_cold void uninit(AVFilterContext *ctx)
  684. {
  685.     DynamicAudioNormalizerContext *s = ctx->priv;
  686.     int c;
  687.  
  688.     av_freep(&s->prev_amplification_factor);
  689.     av_freep(&s->dc_correction_value);
  690.     av_freep(&s->compress_threshold);
  691.     av_freep(&s->fade_factors[0]);
  692.     av_freep(&s->fade_factors[1]);
  693.  
  694.     for (c = 0; c < s->channels; c++) {
  695.         cqueue_free(s->gain_history_original[c]);
  696.         cqueue_free(s->gain_history_minimum[c]);
  697.         cqueue_free(s->gain_history_smoothed[c]);
  698.     }
  699.  
  700.     av_freep(&s->gain_history_original);
  701.     av_freep(&s->gain_history_minimum);
  702.     av_freep(&s->gain_history_smoothed);
  703.  
  704.     av_freep(&s->weights);
  705.  
  706.     ff_bufqueue_discard_all(&s->queue);
  707. }
  708.  
  709. static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = {
  710.     {
  711.         .name           = "default",
  712.         .type           = AVMEDIA_TYPE_AUDIO,
  713.         .filter_frame   = filter_frame,
  714.         .config_props   = config_input,
  715.         .needs_writable = 1,
  716.     },
  717.     { NULL }
  718. };
  719.  
  720. static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = {
  721.     {
  722.         .name          = "default",
  723.         .type          = AVMEDIA_TYPE_AUDIO,
  724.         .config_props  = config_output,
  725.         .request_frame = request_frame,
  726.     },
  727.     { NULL }
  728. };
  729.  
  730. AVFilter ff_af_dynaudnorm = {
  731.     .name          = "dynaudnorm",
  732.     .description   = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."),
  733.     .query_formats = query_formats,
  734.     .priv_size     = sizeof(DynamicAudioNormalizerContext),
  735.     .init          = init,
  736.     .uninit        = uninit,
  737.     .inputs        = avfilter_af_dynaudnorm_inputs,
  738.     .outputs       = avfilter_af_dynaudnorm_outputs,
  739.     .priv_class    = &dynaudnorm_class,
  740. };
  741.