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; }