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avfilter/af_dynaudnorm: use fmin/fmax for doubles

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This commit is contained in:
Paul B Mahol 2022-02-28 10:15:25 +01:00
parent 456d48c752
commit aa6b9066b9
1 changed files with 12 additions and 12 deletions
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24
libavfilter/af_dynaudnorm.c
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@ -385,13 +385,13 @@ static double find_peak_magnitude(AVFrame *frame, int channel)
double *data_ptr = (double *)frame->extended_data[c];
for (i = 0; i < frame->nb_samples; i++)
max = FFMAX(max, fabs(data_ptr[i]));
max = fmax(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]));
max = fmax(max, fabs(data_ptr[i]));
}
return max;
@ -421,7 +421,7 @@ static double compute_frame_rms(AVFrame *frame, int channel)
rms_value /= frame->nb_samples;
}
return FFMAX(sqrt(rms_value), DBL_EPSILON);
return fmax(sqrt(rms_value), DBL_EPSILON);
}
static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame,
@ -433,7 +433,7 @@ static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *
local_gain gain;
gain.threshold = peak_magnitude > s->threshold;
gain.max_gain = bound(s->max_amplification, FFMIN(maximum_gain, rms_gain));
gain.max_gain = bound(s->max_amplification, fmin(maximum_gain, rms_gain));
return gain;
}
@ -444,7 +444,7 @@ static double minimum_filter(cqueue *q)
int i;
for (i = 0; i < cqueue_size(q); i++) {
min = FFMIN(min, cqueue_peek(q, i));
min = fmin(min, cqueue_peek(q, i));
}
return min;
@ -475,7 +475,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
{
if (cqueue_empty(s->gain_history_original[channel])) {
const int pre_fill_size = s->filter_size / 2;
const double initial_value = s->alt_boundary_mode ? gain.max_gain : FFMIN(1.0, gain.max_gain);
const double initial_value = s->alt_boundary_mode ? gain.max_gain : fmin(1.0, gain.max_gain);
s->prev_amplification_factor[channel] = initial_value;
@ -497,7 +497,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
input++;
initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], input));
initial_value = fmin(initial_value, cqueue_peek(s->gain_history_original[channel], input));
cqueue_enqueue(s->gain_history_minimum[channel], initial_value);
}
}
@ -516,7 +516,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
limit = cqueue_peek(s->gain_history_original[channel], 0);
smoothed = FFMIN(smoothed, limit);
smoothed = fmin(smoothed, limit);
cqueue_enqueue(s->gain_history_smoothed[channel], smoothed);
@ -606,7 +606,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
variance /= frame->nb_samples - 1;
}
return FFMAX(sqrt(variance), DBL_EPSILON);
return fmax(sqrt(variance), DBL_EPSILON);
}
static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
@ -616,7 +616,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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 current_threshold = fmin(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;
@ -641,7 +641,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
for (c = 0; c < s->channels; c++) {
const int bypass = bypass_channel(s, frame, 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 current_threshold = setup_compress_thresh(fmin(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;
@ -820,7 +820,7 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
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);
dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? fmin(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];