[FFmpeg-cvslog] avfilter/af_dynaudnorm: reduce number of lines by using for (int ...
Paul B Mahol
git at videolan.org
Mon Feb 28 22:59:24 EET 2022
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Feb 28 21:54:28 2022 +0100| [178d8036dc4bb5f20b58e6a02da3d596b2f6f615] | committer: Paul B Mahol
avfilter/af_dynaudnorm: reduce number of lines by using for (int ...
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=178d8036dc4bb5f20b58e6a02da3d596b2f6f615
---
libavfilter/af_dynaudnorm.c | 73 +++++++++++++++++----------------------------
1 file changed, 28 insertions(+), 45 deletions(-)
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
index 19f3b528d9..837126ac0d 100644
--- a/libavfilter/af_dynaudnorm.c
+++ b/libavfilter/af_dynaudnorm.c
@@ -248,7 +248,6 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
double total_weight = 0.0;
const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0);
double adjust;
- int i;
// Pre-compute constants
const int offset = s->filter_size / 2;
@@ -256,7 +255,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
const double c2 = 2.0 * sigma * sigma;
// Compute weights
- for (i = 0; i < s->filter_size; i++) {
+ for (int i = 0; i < s->filter_size; i++) {
const int x = i - offset;
s->weights[i] = c1 * exp(-x * x / c2);
@@ -265,7 +264,7 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
// Adjust weights
adjust = 1.0 / total_weight;
- for (i = 0; i < s->filter_size; i++) {
+ for (int i = 0; i < s->filter_size; i++) {
s->weights[i] *= adjust;
}
}
@@ -273,13 +272,12 @@ static void init_gaussian_filter(DynamicAudioNormalizerContext *s)
static av_cold void uninit(AVFilterContext *ctx)
{
DynamicAudioNormalizerContext *s = ctx->priv;
- int c;
av_freep(&s->prev_amplification_factor);
av_freep(&s->dc_correction_value);
av_freep(&s->compress_threshold);
- for (c = 0; c < s->channels; c++) {
+ for (int c = 0; c < s->channels; c++) {
if (s->gain_history_original)
cqueue_free(s->gain_history_original[c]);
if (s->gain_history_minimum)
@@ -309,7 +307,6 @@ static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
DynamicAudioNormalizerContext *s = ctx->priv;
- int c;
uninit(ctx);
@@ -333,7 +330,7 @@ static int config_input(AVFilterLink *inlink)
!s->is_enabled || !s->weights)
return AVERROR(ENOMEM);
- for (c = 0; c < inlink->channels; c++) {
+ for (int c = 0; c < inlink->channels; c++) {
s->prev_amplification_factor[c] = 1.0;
s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE);
@@ -378,19 +375,18 @@ static inline double bound(const double threshold, const double val)
static double find_peak_magnitude(AVFrame *frame, int channel)
{
double max = DBL_EPSILON;
- int c, i;
if (channel == -1) {
- for (c = 0; c < frame->channels; c++) {
+ for (int c = 0; c < frame->channels; c++) {
double *data_ptr = (double *)frame->extended_data[c];
- for (i = 0; i < frame->nb_samples; i++)
+ for (int i = 0; i < frame->nb_samples; 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++)
+ for (int i = 0; i < frame->nb_samples; i++)
max = fmax(max, fabs(data_ptr[i]));
}
@@ -400,13 +396,12 @@ static double find_peak_magnitude(AVFrame *frame, int channel)
static double compute_frame_rms(AVFrame *frame, int channel)
{
double rms_value = 0.0;
- int c, i;
if (channel == -1) {
- for (c = 0; c < frame->channels; c++) {
+ for (int c = 0; c < frame->channels; c++) {
const double *data_ptr = (double *)frame->extended_data[c];
- for (i = 0; i < frame->nb_samples; i++) {
+ for (int i = 0; i < frame->nb_samples; i++) {
rms_value += pow_2(data_ptr[i]);
}
}
@@ -414,7 +409,7 @@ static double compute_frame_rms(AVFrame *frame, int channel)
rms_value /= frame->nb_samples * frame->channels;
} else {
const double *data_ptr = (double *)frame->extended_data[channel];
- for (i = 0; i < frame->nb_samples; i++) {
+ for (int i = 0; i < frame->nb_samples; i++) {
rms_value += pow_2(data_ptr[i]);
}
@@ -441,9 +436,8 @@ static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *
static double minimum_filter(cqueue *q)
{
double min = DBL_MAX;
- int i;
- for (i = 0; i < cqueue_size(q); i++) {
+ for (int i = 0; i < cqueue_size(q); i++) {
min = fmin(min, cqueue_peek(q, i));
}
@@ -454,9 +448,8 @@ static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueu
{
const double *weights = s->weights;
double result = 0.0, tsum = 0.0;
- int i;
- for (i = 0; i < cqueue_size(q); i++) {
+ for (int i = 0; i < cqueue_size(q); i++) {
double tq_item = cqueue_peek(tq, i);
double q_item = cqueue_peek(q, i);
@@ -540,21 +533,20 @@ static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *fra
{
const double diff = 1.0 / frame->nb_samples;
int is_first_frame = cqueue_empty(s->gain_history_original[0]);
- int c, i;
- for (c = 0; c < s->channels; c++) {
+ for (int c = 0; c < s->channels; c++) {
const int bypass = bypass_channel(s, frame, c);
double *dst_ptr = (double *)frame->extended_data[c];
double current_average_value = 0.0;
double prev_value;
- for (i = 0; i < frame->nb_samples; i++)
+ for (int i = 0; i < frame->nb_samples; i++)
current_average_value += dst_ptr[i] * diff;
prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c];
s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1);
- for (i = 0; i < frame->nb_samples && !bypass; i++) {
+ for (int i = 0; i < frame->nb_samples && !bypass; i++) {
dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples);
}
}
@@ -586,13 +578,12 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
AVFrame *frame, int channel)
{
double variance = 0.0;
- int i, c;
if (channel == -1) {
- for (c = 0; c < s->channels; c++) {
+ for (int c = 0; c < s->channels; c++) {
const double *data_ptr = (double *)frame->extended_data[c];
- for (i = 0; i < frame->nb_samples; i++) {
+ for (int i = 0; i < frame->nb_samples; i++) {
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
}
}
@@ -600,7 +591,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
} else {
const double *data_ptr = (double *)frame->extended_data[channel];
- for (i = 0; i < frame->nb_samples; i++) {
+ for (int i = 0; i < frame->nb_samples; i++) {
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
}
variance /= frame->nb_samples - 1;
@@ -612,7 +603,6 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame)
{
int is_first_frame = cqueue_empty(s->gain_history_original[0]);
- int c, i;
if (s->channels_coupled) {
const double standard_deviation = compute_frame_std_dev(s, frame, -1);
@@ -625,20 +615,20 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
prev_actual_thresh = setup_compress_thresh(prev_value);
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]);
- for (c = 0; c < s->channels; c++) {
+ for (int c = 0; c < s->channels; c++) {
double *const dst_ptr = (double *)frame->extended_data[c];
const int bypass = bypass_channel(s, frame, c);
if (bypass)
continue;
- for (i = 0; i < frame->nb_samples; i++) {
+ for (int i = 0; i < frame->nb_samples; i++) {
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
}
}
} else {
- for (c = 0; c < s->channels; c++) {
+ for (int 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(fmin(1.0, s->compress_factor * standard_deviation));
@@ -652,7 +642,7 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]);
dst_ptr = (double *)frame->extended_data[c];
- for (i = 0; i < frame->nb_samples && !bypass; i++) {
+ for (int i = 0; i < frame->nb_samples && !bypass; i++) {
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples);
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]);
}
@@ -718,14 +708,10 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
if (s->channels_coupled) {
const local_gain gain = get_max_local_gain(s, analyze_frame, -1);
- int c;
-
- for (c = 0; c < s->channels; c++)
+ for (int c = 0; c < s->channels; c++)
update_gain_history(s, c, gain);
} else {
- int c;
-
- for (c = 0; c < s->channels; c++)
+ for (int c = 0; c < s->channels; c++)
update_gain_history(s, c, get_max_local_gain(s, analyze_frame, c));
}
@@ -735,9 +721,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink
static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in,
AVFrame *frame, int enabled)
{
- int c, i;
-
- for (c = 0; c < s->channels; c++) {
+ for (int c = 0; c < s->channels; c++) {
const int bypass = bypass_channel(s, frame, c);
const double *src_ptr = (const double *)in->extended_data[c];
double *dst_ptr = (double *)frame->extended_data[c];
@@ -745,7 +729,7 @@ static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *in,
cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor);
- for (i = 0; i < frame->nb_samples && enabled && !bypass; i++) {
+ for (int i = 0; i < frame->nb_samples && enabled && !bypass; i++) {
const double amplification_factor = fade(s->prev_amplification_factor[c],
current_amplification_factor, i,
frame->nb_samples);
@@ -811,15 +795,14 @@ static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink,
AVFilterLink *outlink)
{
AVFrame *out = ff_get_audio_buffer(outlink, s->sample_advance);
- int c, i;
if (!out)
return AVERROR(ENOMEM);
- for (c = 0; c < s->channels; c++) {
+ for (int c = 0; c < s->channels; c++) {
double *dst_ptr = (double *)out->extended_data[c];
- for (i = 0; i < out->nb_samples; i++) {
+ for (int i = 0; i < out->nb_samples; i++) {
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;
More information about the ffmpeg-cvslog
mailing list