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Merge pull request #619 from lioncash/dpl2 

Correct the function casing for DPL2Decoder
This commit is contained in:
Pierre Bourdon 2014-07-14 10:21:20 +02:00
parent 72b15be3be 2c73472671
commit 1464cfee13
3 changed files with 36 additions and 29 deletions
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57
Source/Core/AudioCommon/DPL2Decoder.cpp
View File

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

4
Source/Core/AudioCommon/DPL2Decoder.h
View File

@ -4,5 +4,5 @@
#pragma once
void dpl2decode(float *samples, int numsamples, float *out);
void dpl2reset();
void DPL2Decode(float *samples, int numsamples, float *out);
void DPL2Reset();

4
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -58,7 +58,7 @@ bool OpenALStream::Start()
}
// Initialize DPL2 parameters
dpl2reset();
DPL2Reset();
soundTouch.clear();
return bReturn;
@ -246,7 +246,7 @@ void OpenALStream::SoundLoop()
if (surround_capable)
{
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
dpl2decode(sampleBuffer, nSamples, dpl2);
DPL2Decode(sampleBuffer, nSamples, dpl2);
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)