Welcome to mirror list, hosted at ThFree Co, Russian Federation.

github.com/FFmpeg/FFmpeg.git - Unnamed repository; edit this file 'description' to name the repository.
summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorPaul B Mahol <onemda@gmail.com>2020-01-04 20:17:32 +0300
committerPaul B Mahol <onemda@gmail.com>2020-01-04 20:17:32 +0300
commitc8253cb3328421b3825787e2e2f853ffe1128aa9 (patch)
tree4776358fed17cb167faa58a2301632d950ec6eee /libavfilter/af_dynaudnorm.c
parent6a1305e8b7d191e6b8647f89a22efbad7e81cc05 (diff)
avfilter/af_dynaudnorm: implement threshold option
Diffstat (limited to 'libavfilter/af_dynaudnorm.c')
-rw-r--r--libavfilter/af_dynaudnorm.c58
1 files changed, 43 insertions, 15 deletions
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
index 09d2aeb816..1330433fbc 100644
--- a/libavfilter/af_dynaudnorm.c
+++ b/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,
- int channel)
+static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
+ 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;