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Code Issues Releases Activity

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]; double *data_ptr = (double *)frame->extended_data[c];
for (i = 0; i < frame->nb_samples; i++) for (i = 0; i < frame->nb_samples; i++)
max = FFMAX(max, fabs(data_ptr[i])); max = fmax(max, fabs(data_ptr[i]));
} }
} else { } else {
double *data_ptr = (double *)frame->extended_data[channel]; double *data_ptr = (double *)frame->extended_data[channel];
for (i = 0; i < frame->nb_samples; i++) for (i = 0; i < frame->nb_samples; i++)
max = FFMAX(max, fabs(data_ptr[i])); max = fmax(max, fabs(data_ptr[i]));
} }
return max; return max;
@@ -421,7 +421,7 @@ static double compute_frame_rms(AVFrame *frame, int channel)
rms_value /= frame->nb_samples; 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, 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; local_gain gain;
gain.threshold = peak_magnitude > s->threshold; 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; return gain;
} }
@@ -444,7 +444,7 @@ static double minimum_filter(cqueue *q)
int i; int i;
for (i = 0; i < cqueue_size(q); 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; return min;
@@ -475,7 +475,7 @@ static void update_gain_history(DynamicAudioNormalizerContext *s, int channel,
{ {
if (cqueue_empty(s->gain_history_original[channel])) { if (cqueue_empty(s->gain_history_original[channel])) {
const int pre_fill_size = s->filter_size / 2; 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; 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) { while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) {
input++; 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); 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]); smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]);
limit = cqueue_peek(s->gain_history_original[channel], 0); 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); 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; 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) static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
@@ -616,7 +616,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
if (s->channels_coupled) { if (s->channels_coupled) {
const double standard_deviation = compute_frame_std_dev(s, frame, -1); 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]; const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0];
double prev_actual_thresh, curr_actual_thresh; 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++) { for (c = 0; c < s->channels; c++) {
const int bypass = bypass_channel(s, frame, c); const int bypass = bypass_channel(s, frame, c);
const double standard_deviation = compute_frame_std_dev(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]; const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c];
double prev_actual_thresh, curr_actual_thresh; double prev_actual_thresh, curr_actual_thresh;
double *dst_ptr; double *dst_ptr;
@@ -820,7 +820,7 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
double *dst_ptr = (double *)out->extended_data[c]; double *dst_ptr = (double *)out->extended_data[c];
for (i = 0; i < out->nb_samples; i++) { 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) { if (s->dc_correction) {
dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1; dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1;
dst_ptr[i] += s->dc_correction_value[c]; dst_ptr[i] += s->dc_correction_value[c];