diff options
Diffstat (limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')
| -rw-r--r-- | Source/Core/AudioCommon/DPL2Decoder.cpp | 26 |
1 files changed, 12 insertions, 14 deletions
diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp index fa483249e6..8b2f0e53b2 100644 --- a/Source/Core/AudioCommon/DPL2Decoder.cpp +++ b/Source/Core/AudioCommon/DPL2Decoder.cpp @@ -105,9 +105,7 @@ static void Hamming(int n, float* w) *w++ = float(0.54 - 0.46 * cos(k * (float)i)); } -/****************************************************************************** -* FIR filter design -******************************************************************************/ +// FIR filter design /* Design FIR filter using the Window method @@ -122,10 +120,10 @@ opt beta constant used only when designing using kaiser windows returns 0 if OK, -1 if fail */ -static float* DesignFIR(unsigned int* n, float* fc, float opt) +static float* DesignFIR(unsigned int n, float fc, float opt) { - unsigned int o = *n & 1; // Indicator for odd filter length - unsigned int end = ((*n + 1) >> 1) - o; // Loop end + unsigned int o = n & 1; // Indicator for odd filter length + unsigned int end = ((n + 1) >> 1) - o; // Loop end float k1 = 2 * float(M_PI); // 2*pi*fc1 float k2 = 0.5f * (float)(1 - o); // Constant used if the filter has even length @@ -134,17 +132,17 @@ static float* DesignFIR(unsigned int* n, float* fc, float opt) float fc1; // Cutoff frequencies // Sanity check - if (*n == 0) + if (n == 0) return nullptr; - fc[0] = MathUtil::Clamp(fc[0], 0.001f, 1.0f); + fc = MathUtil::Clamp(fc, 0.001f, 1.0f); - float* w = (float*)calloc(sizeof(float), *n); + float* w = (float*)calloc(sizeof(float), n); // Get window coefficients - Hamming(*n, w); + Hamming(n, w); - fc1 = *fc; + fc1 = fc; // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1 / 2 : 0.25f; k1 *= fc1; @@ -165,13 +163,13 @@ static float* DesignFIR(unsigned int* n, float* fc, float opt) for (u32 i = 0; i < end; i++) { t1 = (float)(i + 1) - k2; - w[end - i - 1] = w[*n - end + i] = float(w[end - i - 1] * sin(k1 * t1) / (M_PI * t1)); // Sinc + w[end - i - 1] = w[n - end + i] = float(w[end - i - 1] * sin(k1 * t1) / (M_PI * t1)); // Sinc g += 2 * w[end - i - 1]; // Total gain in filter } // Normalize gain g = 1 / g; - for (u32 i = 0; i < *n; i++) + for (u32 i = 0; i < n; i++) w[i] *= g; return w; @@ -209,7 +207,7 @@ static float* CalculateCoefficients125HzLowpass(int rate) { len125 = 256; float f = 125.0f / (rate / 2); - float* coeffs = DesignFIR(&len125, &f, 0); + float* coeffs = DesignFIR(len125, f, 0); static const float M3_01DB = 0.7071067812f; for (unsigned int i = 0; i < len125; i++) { |