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-rw-r--r--libavfilter/af_dynaudnorm.c10
1 files changed, 5 insertions, 5 deletions
diff --git a/libavfilter/af_dynaudnorm.c b/libavfilter/af_dynaudnorm.c
index ddbef26ab5..aa5b28e647 100644
--- a/libavfilter/af_dynaudnorm.c
+++ b/libavfilter/af_dynaudnorm.c
@@ -341,7 +341,7 @@ static inline double fade(double prev, double next, int pos,
return fade_factors[0][pos] * prev + fade_factors[1][pos] * next;
}
-static inline double pow2(const double value)
+static inline double pow_2(const double value)
{
return value * value;
}
@@ -384,7 +384,7 @@ static double compute_frame_rms(AVFrame *frame, int channel)
const double *data_ptr = (double *)frame->extended_data[c];
for (i = 0; i < frame->nb_samples; i++) {
- rms_value += pow2(data_ptr[i]);
+ rms_value += pow_2(data_ptr[i]);
}
}
@@ -392,7 +392,7 @@ static double compute_frame_rms(AVFrame *frame, int channel)
} else {
const double *data_ptr = (double *)frame->extended_data[channel];
for (i = 0; i < frame->nb_samples; i++) {
- rms_value += pow2(data_ptr[i]);
+ rms_value += pow_2(data_ptr[i]);
}
rms_value /= frame->nb_samples;
@@ -545,7 +545,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
const double *data_ptr = (double *)frame->extended_data[c];
for (i = 0; i < frame->nb_samples; i++) {
- variance += pow2(data_ptr[i]); // Assume that MEAN is *zero*
+ variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
}
}
variance /= (s->channels * frame->nb_samples) - 1;
@@ -553,7 +553,7 @@ static double compute_frame_std_dev(DynamicAudioNormalizerContext *s,
const double *data_ptr = (double *)frame->extended_data[channel];
for (i = 0; i < frame->nb_samples; i++) {
- variance += pow2(data_ptr[i]); // Assume that MEAN is *zero*
+ variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero*
}
variance /= frame->nb_samples - 1;
}