mirror of https://git.ffmpeg.org/ffmpeg.git
avfilter/af_dynaudnorm: implement threshold option
This commit is contained in:
parent
6a1305e8b7
commit
c8253cb332
|
@ -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
|
||||
|
|
|
@ -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;
|
||||
|
||||
|
|
Loading…
Reference in New Issue