avfilter/af_dynaudnorm: implement threshold option

doc/filters.texi

@@ -3438,6 +3438,14 @@ value. Instead, the threshold value will be adjusted for each individual
 frame.
 In general, smaller parameters result in stronger compression, and vice versa.
 Values below 3.0 are not recommended, because audible distortion may appear.
+
+@item threshold, t
+Set the target threshold value. This specifies the loweset permissible
+magnitude level for the audio input which will be normalized.
+If input frame volume is above this value frame will be normalized.
+Otherwise frame may not be normalized at all. The default value is set
+to 0, which means all input frames will be normalized.
+This option is mostly useful if digital noise is not wanted to be amplified.
 @end table
 
 @section earwax

libavfilter/af_dynaudnorm.c

@@ -37,6 +37,11 @@
 #include "filters.h"
 #include "internal.h"
 
+typedef struct local_gain {
+    double max_gain;
+    double threshold;
+} local_gain;
+
 typedef struct cqueue {
     double *elements;
     int size;
@@ -60,6 +65,7 @@ typedef struct DynamicAudioNormalizerContext {
     double max_amplification;
     double target_rms;
     double compress_factor;
+    double threshold;
     double *prev_amplification_factor;
     double *dc_correction_value;
     double *compress_threshold;
@@ -73,6 +79,7 @@ typedef struct DynamicAudioNormalizerContext {
     cqueue **gain_history_original;
     cqueue **gain_history_minimum;
     cqueue **gain_history_smoothed;
+    cqueue **threshold_history;
 
     cqueue *is_enabled;
 } DynamicAudioNormalizerContext;
@@ -99,6 +106,8 @@ static const AVOption dynaudnorm_options[] = {
     { "b",           "set alternative boundary mode",    OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL,   {.i64 = 0},      0,     1, FLAGS },
     { "compress",    "set the compress factor",          OFFSET(compress_factor),   AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,  30.0, FLAGS },
     { "s",           "set the compress factor",          OFFSET(compress_factor),   AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,  30.0, FLAGS },
+    { "threshold",   "set the threshold value",          OFFSET(threshold),         AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,   1.0, FLAGS },
+    { "t",           "set the threshold value",          OFFSET(threshold),         AV_OPT_TYPE_DOUBLE, {.dbl = 0.0},  0.0,   1.0, FLAGS },
     { NULL }
 };
 
@@ -286,11 +295,14 @@ static av_cold void uninit(AVFilterContext *ctx)
             cqueue_free(s->gain_history_minimum[c]);
         if (s->gain_history_smoothed)
             cqueue_free(s->gain_history_smoothed[c]);
+        if (s->threshold_history)
+            cqueue_free(s->threshold_history[c]);
     }
 
     av_freep(&s->gain_history_original);
     av_freep(&s->gain_history_minimum);
     av_freep(&s->gain_history_smoothed);
+    av_freep(&s->threshold_history);
 
     cqueue_free(s->is_enabled);
     s->is_enabled = NULL;
@@ -320,12 +332,14 @@ static int config_input(AVFilterLink *inlink)
     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->threshold_history = av_calloc(inlink->channels, sizeof(*s->threshold_history));
     s->weights = av_malloc_array(s->filter_size, sizeof(*s->weights));
     s->is_enabled = cqueue_create(s->filter_size);
     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->is_enabled || !s->weights)
+        !s->gain_history_smoothed || !s->threshold_history ||
+        !s->is_enabled || !s->weights)
         return AVERROR(ENOMEM);
 
     for (c = 0; c < inlink->channels; c++) {
@@ -334,9 +348,10 @@ static int config_input(AVFilterLink *inlink)
         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);
+        s->threshold_history[c]     = cqueue_create(s->filter_size);
 
         if (!s->gain_history_original[c] || !s->gain_history_minimum[c] ||
-            !s->gain_history_smoothed[c])
+            !s->gain_history_smoothed[c] || !s->threshold_history[c])
             return AVERROR(ENOMEM);
     }
 
@@ -414,12 +429,18 @@ static double compute_frame_rms(AVFrame *frame, int channel)
     return FFMAX(sqrt(rms_value), DBL_EPSILON);
 }
 
-static double get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
+static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
-                                 int channel)
+                                     int channel)
 {
-    const double maximum_gain = s->peak_value / find_peak_magnitude(frame, channel);
+    const double peak_magnitude = find_peak_magnitude(frame, channel);
+    const double maximum_gain = s->peak_value / peak_magnitude;
     const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX;
-    return bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
+    local_gain gain;
+
+    gain.threshold = peak_magnitude > s->threshold;
+    gain.max_gain  = bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
+
+    return gain;
 }
 
 static double minimum_filter(cqueue *q)
@@ -434,34 +455,40 @@ static double minimum_filter(cqueue *q)
     return min;
 }
 
-static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q)
+static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueue *tq)
 {
-    double result = 0.0;
+    double result = 0.0, tsum = 0.0;
     int i;
 
     for (i = 0; i < cqueue_size(q); i++) {
-        result += cqueue_peek(q, i) * s->weights[i];
+        tsum += cqueue_peek(tq, i) * s->weights[i];
+        result += cqueue_peek(q, i) * s->weights[i] * cqueue_peek(tq, i);
     }
 
+    if (tsum == 0.0)
+        result = 1.0;
+
     return result;
 }
 
 static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
-                                double current_gain_factor)
+                                local_gain gain)
 {
     if (cqueue_empty(s->gain_history_original[channel]) ||
         cqueue_empty(s->gain_history_minimum[channel])) {
         const int pre_fill_size = s->filter_size / 2;
-        const double initial_value = s->alt_boundary_mode ? current_gain_factor : 1.0;
+        const double initial_value = s->alt_boundary_mode ? gain.max_gain : 1.0;
 
         s->prev_amplification_factor[channel] = initial_value;
 
         while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) {
             cqueue_enqueue(s->gain_history_original[channel], initial_value);
+            cqueue_enqueue(s->threshold_history[channel], gain.threshold);
         }
     }
 
-    cqueue_enqueue(s->gain_history_original[channel], current_gain_factor);
+    cqueue_enqueue(s->gain_history_original[channel], gain.max_gain);
+    cqueue_enqueue(s->threshold_history[channel], gain.threshold);
 
     while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) {
         double minimum;
@@ -489,12 +516,13 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int 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]);
+        smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
         smoothed = FFMIN(smoothed, cqueue_peek(s->gain_history_minimum[channel], s->filter_size / 2));
 
         cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
 
         cqueue_pop(s->gain_history_minimum[channel]);
+        cqueue_pop(s->threshold_history[channel]);
     }
 }
 
@@ -632,11 +660,11 @@ static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
     }
 
     if (s->channels_coupled) {
-        const double current_gain_factor = get_max_local_gain(s, frame, -1);
+        const local_gain gain = get_max_local_gain(s, frame, -1);
         int c;
 
         for (c = 0; c < s->channels; c++)
-            update_gain_history(s, c, current_gain_factor);
+            update_gain_history(s, c, gain);
     } else {
         int c;