Correct the function casing for DPL2Decoder
Brings it more in-line with the rest of the codebase.
This commit is contained in:
Lioncash
parent
dee6f226a3
commit
2c73472671
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@ -36,7 +36,8 @@ static unsigned int lfe_pos;
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static float *filter_coefs_lfe;
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static unsigned int len125;
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template<class T,class _ftype_t> static _ftype_t dotproduct(int count,const T *buf,const _ftype_t *coefficients)
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template<class T, class _ftype_t>
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static _ftype_t DotProduct(int count,const T *buf,const _ftype_t *coefficients)
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{
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float sum0=0,sum1=0,sum2=0,sum3=0;
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for (;count>=4;buf+=4,coefficients+=4,count-=4)
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@ -46,11 +47,15 @@ template<class T,class _ftype_t> static _ftype_t dotproduct(int count,const T *b
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sum2+=buf[2]*coefficients[2];
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sum3+=buf[3]*coefficients[3];
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}
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while (count--) sum0+= *buf++ * *coefficients++;
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while (count--)
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sum0+= *buf++ * *coefficients++;
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return sum0+sum1+sum2+sum3;
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}
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template<class T> static T firfilter(const T *buf, int pos, int len, int count, const float *coefficients)
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template<class T>
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static T FIRFilter(const T *buf, int pos, int len, int count, const float *coefficients)
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{
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int count1, count2;
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@ -69,8 +74,8 @@ template<class T> static T firfilter(const T *buf, int pos, int len, int count,
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// high part of window
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const T *ptr = &buf[pos];
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float r1=dotproduct(count1,ptr,coefficients);coefficients+=count1;
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float r2=dotproduct(count2,buf,coefficients);
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float r1=DotProduct(count1,ptr,coefficients);coefficients+=count1;
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float r2=DotProduct(count2,buf,coefficients);
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return T(r1+r2);
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}
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@ -83,7 +88,7 @@ template<class T> static T firfilter(const T *buf, int pos, int len, int count,
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// n window length
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// w buffer for the window parameters
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*/
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static void hamming(int n, float* w)
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static void Hamming(int n, float* w)
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{
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int i;
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float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1)
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@ -110,7 +115,7 @@ opt beta constant used only when designing using kaiser windows
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returns 0 if OK, -1 if fail
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*/
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static float* design_fir(unsigned int *n, float* fc, float opt)
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static float* DesignFIR(unsigned int *n, float* fc, float opt)
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{
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unsigned int o = *n & 1; // Indicator for odd filter length
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unsigned int end = ((*n + 1) >> 1) - o; // Loop end
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@ -129,7 +134,7 @@ static float* design_fir(unsigned int *n, float* fc, float opt)
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float *w=(float*)calloc(sizeof(float),*n);
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// Get window coefficients
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hamming(*n,w);
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Hamming(*n,w);
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fc1=*fc;
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// Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
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@ -165,7 +170,7 @@ static float* design_fir(unsigned int *n, float* fc, float opt)
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return w;
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}
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static void onSeek(void)
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static void OnSeek()
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{
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l_fwr = r_fwr = lpr_fwr = lmr_fwr = 0;
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std::fill(fwrbuf_l.begin(), fwrbuf_l.end(), 0.0f);
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@ -181,21 +186,23 @@ static void onSeek(void)
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memset(LFE_buf, 0, sizeof(LFE_buf));
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}
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static void done(void)
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static void Done()
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{
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onSeek();
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OnSeek();
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if (filter_coefs_lfe)
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{
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free(filter_coefs_lfe);
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}
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filter_coefs_lfe = nullptr;
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}
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static float* calc_coefficients_125Hz_lowpass(int rate)
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static float* CalculateCoefficients125HzLowpass(int rate)
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{
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len125 = 256;
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float f = 125.0f / (rate / 2);
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float *coeffs = design_fir(&len125, &f, 0);
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float *coeffs = DesignFIR(&len125, &f, 0);
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static const float M3_01DB = 0.7071067812f;
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for (unsigned int i = 0; i < len125; i++)
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{
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@ -204,7 +211,7 @@ static float* calc_coefficients_125Hz_lowpass(int rate)
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return coeffs;
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}
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static float passive_lock(float x)
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static float PassiveLock(float x)
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{
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static const float MATAGCLOCK = 0.2f; /* AGC range (around 1) where the matrix behaves passively */
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const float x1 = x - 1;
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@ -212,7 +219,7 @@ static float passive_lock(float x)
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return x1 - x1 / (1 + ax1s * ax1s) + 1;
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}
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static void matrix_decode(const float *in, const int k, const int il,
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static void MatrixDecode(const float *in, const int k, const int il,
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const int ir, bool decode_rear,
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const int _dlbuflen,
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float _l_fwr, float _r_fwr,
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@ -251,8 +258,8 @@ static void matrix_decode(const float *in, const int k, const int il,
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*_adapt_l_gain = (1 - f) * *_adapt_l_gain + f * l_gain;
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*_adapt_r_gain = (1 - f) * *_adapt_r_gain + f * r_gain;
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/* Matrix */
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l_agc = in[il] * passive_lock(*_adapt_l_gain);
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r_agc = in[ir] * passive_lock(*_adapt_r_gain);
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l_agc = in[il] * PassiveLock(*_adapt_l_gain);
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r_agc = in[ir] * PassiveLock(*_adapt_r_gain);
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_cf[k] = (l_agc + r_agc) * (float)M_SQRT1_2;
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if (decode_rear)
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{
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@ -274,8 +281,8 @@ static void matrix_decode(const float *in, const int k, const int il,
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*_adapt_lpr_gain = (1 - f) * *_adapt_lpr_gain + f * lpr_gain;
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*_adapt_lmr_gain = (1 - f) * *_adapt_lmr_gain + f * lmr_gain;
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/* Matrix */
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lpr_agc = lpr * passive_lock(*_adapt_lpr_gain);
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lmr_agc = lmr * passive_lock(*_adapt_lmr_gain);
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lpr_agc = lpr * PassiveLock(*_adapt_lpr_gain);
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lmr_agc = lmr * PassiveLock(*_adapt_lmr_gain);
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_lf[k] = (lpr_agc + lmr_agc) * (float)M_SQRT1_2;
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_rf[k] = (lpr_agc - lmr_agc) * (float)M_SQRT1_2;
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@ -297,7 +304,7 @@ static void matrix_decode(const float *in, const int k, const int il,
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_cf[k] += c_agc_cfk + c_agc_cfk;
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}
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void dpl2decode(float *samples, int numsamples, float *out)
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void DPL2Decode(float *samples, int numsamples, float *out)
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{
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static const unsigned int FWRDURATION = 240; // FWR average duration (samples)
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static const int cfg_delay = 0;
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@ -308,7 +315,7 @@ void dpl2decode(float *samples, int numsamples, float *out)
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if (olddelay != cfg_delay || oldfreq != fmt_freq)
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{
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done();
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Done();
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olddelay = cfg_delay;
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oldfreq = fmt_freq;
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dlbuflen = std::max(FWRDURATION, (fmt_freq * cfg_delay / 1000)); //+(len7000-1);
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@ -321,7 +328,7 @@ void dpl2decode(float *samples, int numsamples, float *out)
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rr.resize(dlbuflen);
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cf.resize(dlbuflen);
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cr.resize(dlbuflen);
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filter_coefs_lfe = calc_coefficients_125Hz_lowpass(fmt_freq);
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filter_coefs_lfe = CalculateCoefficients125HzLowpass(fmt_freq);
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lfe_pos = 0;
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memset(LFE_buf, 0, sizeof(LFE_buf));
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}
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@ -344,7 +351,7 @@ void dpl2decode(float *samples, int numsamples, float *out)
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/* Matrix encoded 2 channel sources */
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fwrbuf_l[k] = in[0];
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fwrbuf_r[k] = in[1];
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matrix_decode(in, k, 0, 1, true, dlbuflen,
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MatrixDecode(in, k, 0, 1, true, dlbuflen,
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l_fwr, r_fwr,
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lpr_fwr, lmr_fwr,
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&adapt_l_gain, &adapt_r_gain,
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@ -355,7 +362,7 @@ void dpl2decode(float *samples, int numsamples, float *out)
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out[cur + 1] = rf[k];
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out[cur + 2] = cf[k];
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LFE_buf[lfe_pos] = (out[0] + out[1]) / 2;
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out[cur + 3] = firfilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
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out[cur + 3] = FIRFilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
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lfe_pos++;
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if (lfe_pos == len125)
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{
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@ -374,7 +381,7 @@ void dpl2decode(float *samples, int numsamples, float *out)
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}
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}
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void dpl2reset()
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void DPL2Reset()
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{
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olddelay = -1;
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oldfreq = 0;
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@ -4,5 +4,5 @@
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#pragma once
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void dpl2decode(float *samples, int numsamples, float *out);
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void dpl2reset();
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void DPL2Decode(float *samples, int numsamples, float *out);
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void DPL2Reset();
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@ -58,7 +58,7 @@ bool OpenALStream::Start()
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}
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// Initialize DPL2 parameters
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dpl2reset();
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DPL2Reset();
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soundTouch.clear();
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return bReturn;
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@ -246,7 +246,7 @@ void OpenALStream::SoundLoop()
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if (surround_capable)
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{
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float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
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dpl2decode(sampleBuffer, nSamples, dpl2);
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DPL2Decode(sampleBuffer, nSamples, dpl2);
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
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ALenum err = alGetError();
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if (err == AL_INVALID_ENUM)
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