From 34692ab826abc8f8faa61bdb2280b742424528f1 Mon Sep 17 00:00:00 2001
From: "Jasper St. Pierre" <jstpierre@mecheye.net>
Date: Sat, 7 Dec 2013 15:14:29 -0500
Subject: Remove unnecessary Src/ folders

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 391 ++++++++++++++++++++++++++++++++
 1 file changed, 391 insertions(+)
 create mode 100644 Source/Core/AudioCommon/DPL2Decoder.cpp

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
new file mode 100644
index 0000000000..97bdb5413d
--- /dev/null
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -0,0 +1,391 @@
+// Copyright 2013 Dolphin Emulator Project
+// Licensed under GPLv2
+// Refer to the license.txt file included.
+
+// Dolby Pro Logic 2 decoder from ffdshow-tryout
+//  * Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
+//  * Copyright (c) 2004-2006 Milan Cutka
+//  * based on mplayer HRTF plugin by ylai
+
+#include <functional>
+#include <vector>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include "DPL2Decoder.h"
+
+#ifndef M_PI
+#define M_PI 3.14159265358979323846
+#endif
+#ifndef M_SQRT1_2
+#define M_SQRT1_2 0.70710678118654752440
+#endif
+
+int olddelay = -1;
+unsigned int oldfreq = 0;
+unsigned int dlbuflen;
+int cyc_pos;
+float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
+std::vector<float> fwrbuf_l, fwrbuf_r;
+float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
+std::vector<float> lf, rf, lr, rr, cf, cr;
+float LFE_buf[256];
+unsigned int lfe_pos;
+float *filter_coefs_lfe;
+unsigned int len125;
+
+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)
+	{
+		sum0+=buf[0]*coefficients[0];
+		sum1+=buf[1]*coefficients[1];
+		sum2+=buf[2]*coefficients[2];
+		sum3+=buf[3]*coefficients[3];
+	}
+	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)
+{
+	int count1, count2;
+
+	if (pos >= count)
+	{
+		pos -= count;
+		count1 = count; count2 = 0;
+	}
+	else
+	{
+		count2 = pos;
+		count1 = count - pos;
+		pos = len - count1;
+	}
+
+	// high part of window
+	const T *ptr = &buf[pos];
+
+	float r1=dotproduct(count1,ptr,coefficients);coefficients+=count1;
+	float r2=dotproduct(count2,buf,coefficients);
+	return T(r1+r2);
+}
+
+template<class T> inline const T& limit(const T& val, const T& min, const T& max)
+{
+	if (val < min) {
+		return min;
+	} else if (val > max) {
+		return max;
+	} else {
+		return val;
+	}
+}
+
+/*
+// Hamming
+//                        2*pi*k
+// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N
+//                         N-1
+//
+// n window length
+// w buffer for the window parameters
+*/
+void hamming(int n, float* w)
+{
+	int      i;
+	float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1)
+
+	// Calculate window coefficients
+	for (i=0; i<n; i++)
+		*w++ = float(0.54 - 0.46*cos(k*(float)i));
+}
+
+/******************************************************************************
+*  FIR filter design
+******************************************************************************/
+
+/* Design FIR filter using the Window method
+
+n     filter length must be odd for HP and BS filters
+w     buffer for the filter taps (must be n long)
+fc    cutoff frequencies (1 for LP and HP, 2 for BP and BS)
+0 < fc < 1 where 1 <=> Fs/2
+flags window and filter type as defined in filter.h
+variables are ored together: i.e. LP|HAMMING will give a
+low pass filter designed using a hamming window
+opt   beta constant used only when designing using kaiser windows
+
+returns 0 if OK, -1 if fail
+*/
+float* design_fir(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  i;                         // Loop index
+
+	float k1 = 2 * float(M_PI);              // 2*pi*fc1
+	float k2 = 0.5f * (float)(1 - o);        // Constant used if the filter has even length
+	float g  = 0.0f;                         // Gain
+	float t1;                                // Temporary variables
+	float fc1;                               // Cutoff frequencies
+
+	// Sanity check
+	if(*n==0) return NULL;
+	fc[0]=limit(fc[0],float(0.001),float(1));
+
+	float *w=(float*)calloc(sizeof(float),*n);
+
+	// Get window coefficients
+	hamming(*n,w);
+
+	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;
+
+	// Low pass filter
+
+	// If the filter length is odd, there is one point which is exactly
+	// in the middle. The value at this point is 2*fCutoff*sin(x)/x,
+	// where x is zero. To make sure nothing strange happens, we set this
+	// value separately.
+	if (o)
+	{
+		w[end] = fc1 * w[end] * 2.0f;
+		g=w[end];
+	}
+
+	// Create filter
+	for (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
+		g += 2*w[end-i-1]; // Total gain in filter
+	}
+
+
+	// Normalize gain
+	g=1/g;
+	for (i=0; i<*n; i++)
+		w[i] *= g;
+
+	return w;
+}
+
+void onSeek(void)
+{
+	l_fwr = r_fwr = lpr_fwr = lmr_fwr = 0;
+	std::fill(fwrbuf_l.begin(), fwrbuf_l.end(), 0.0f);
+	std::fill(fwrbuf_r.begin(), fwrbuf_r.end(), 0.0f);
+	adapt_l_gain = adapt_r_gain = adapt_lpr_gain = adapt_lmr_gain = 0;
+	std::fill(lf.begin(), lf.end(), 0.0f);
+	std::fill(rf.begin(), rf.end(), 0.0f);
+	std::fill(lr.begin(), lr.end(), 0.0f);
+	std::fill(rr.begin(), rr.end(), 0.0f);
+	std::fill(cf.begin(), cf.end(), 0.0f);
+	std::fill(cr.begin(), cr.end(), 0.0f);
+	lfe_pos = 0;
+	memset(LFE_buf, 0, sizeof(LFE_buf));
+}
+
+void done(void)
+{
+	onSeek();
+	if (filter_coefs_lfe)
+	{
+		free(filter_coefs_lfe);
+	}
+	filter_coefs_lfe = NULL;
+}
+
+float* calc_coefficients_125Hz_lowpass(int rate)
+{
+	len125 = 256;
+	float f = 125.0f / (rate / 2);
+	float *coeffs = design_fir(&len125, &f, 0);
+	static const float M3_01DB = 0.7071067812f;
+	for (unsigned int i = 0; i < len125; i++)
+	{
+		coeffs[i] *= M3_01DB;
+	}
+	return coeffs;
+}
+
+float passive_lock(float x)
+{
+	static const float MATAGCLOCK = 0.2f;  /* AGC range (around 1) where the matrix behaves passively */
+	const float x1 = x - 1;
+	const float ax1s = fabs(x - 1) * (1.0f / MATAGCLOCK);
+	return x1 - x1 / (1 + ax1s * ax1s) + 1;
+}
+
+void matrix_decode(const float *in, const int k, const int il,
+	const int ir, bool decode_rear,
+	const int _dlbuflen,
+	float _l_fwr, float _r_fwr,
+	float _lpr_fwr, float _lmr_fwr,
+	float *_adapt_l_gain, float *_adapt_r_gain,
+	float *_adapt_lpr_gain, float *_adapt_lmr_gain,
+	float *_lf, float *_rf, float *_lr,
+	float *_rr, float *_cf)
+{
+	static const float M9_03DB = 0.3535533906f;
+	static const float MATAGCTRIG = 8.0f;   /* (Fuzzy) AGC trigger */
+	static const float MATAGCDECAY = 1.0f;  /* AGC baseline decay rate (1/samp.) */
+	static const float MATCOMPGAIN = 0.37f; /* Cross talk compensation gain,  0.50 - 0.55 is full cancellation. */
+
+	const int kr = (k + olddelay) % _dlbuflen;
+	float l_gain = (_l_fwr + _r_fwr) / (1 + _l_fwr + _l_fwr);
+	float r_gain = (_l_fwr + _r_fwr) / (1 + _r_fwr + _r_fwr);
+	// The 2nd axis has strong gain fluctuations, and therefore require
+	// limits.  The factor corresponds to the 1 / amplification of (Lt
+	// - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
+	// dialogues).  It should be bigger than -12 dB to prevent
+	// distortion.
+	float lmr_lim_fwr = _lmr_fwr > M9_03DB * _lpr_fwr ? _lmr_fwr : M9_03DB * _lpr_fwr;
+	float lpr_gain = (_lpr_fwr + lmr_lim_fwr) / (1 + _lpr_fwr + _lpr_fwr);
+	float lmr_gain = (_lpr_fwr + lmr_lim_fwr) / (1 + lmr_lim_fwr + lmr_lim_fwr);
+	float lmr_unlim_gain = (_lpr_fwr + _lmr_fwr) / (1 + _lmr_fwr + _lmr_fwr);
+	float lpr, lmr;
+	float l_agc, r_agc, lpr_agc, lmr_agc;
+	float f, d_gain, c_gain, c_agc_cfk;
+
+	/*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
+	/* AGC adaption */
+	d_gain = (fabs(l_gain - *_adapt_l_gain) + fabs(r_gain - *_adapt_r_gain)) * 0.5f;
+	f = d_gain * (1.0f / MATAGCTRIG);
+	f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
+	*_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);
+	_cf[k] = (l_agc + r_agc) * (float)M_SQRT1_2;
+	if (decode_rear)
+	{
+		_lr[kr] = _rr[kr] = (l_agc - r_agc) * (float)M_SQRT1_2;
+		// Stereo rear channel is steered with the same AGC steering as
+		// the decoding matrix. Note this requires a fast updating AGC
+		// at the order of 20 ms (which is the case here).
+		_lr[kr] *= (_l_fwr + _l_fwr) / (1 + _l_fwr + _r_fwr);
+		_rr[kr] *= (_r_fwr + _r_fwr) / (1 + _l_fwr + _r_fwr);
+	}
+
+	/*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
+	lpr = (in[il] + in[ir]) * (float)M_SQRT1_2;
+	lmr = (in[il] - in[ir]) * (float)M_SQRT1_2;
+	/* AGC adaption */
+	d_gain = fabs(lmr_unlim_gain - *_adapt_lmr_gain);
+	f = d_gain * (1.0f / MATAGCTRIG);
+	f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
+	*_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);
+	_lf[k] = (lpr_agc + lmr_agc) * (float)M_SQRT1_2;
+	_rf[k] = (lpr_agc - lmr_agc) * (float)M_SQRT1_2;
+
+	/*** CENTER FRONT CANCELLATION ***/
+	// A heuristic approach exploits that Lt + Rt gain contains the
+	// information about Lt, Rt correlation.  This effectively reshapes
+	// the front and rear "cones" to concentrate Lt + Rt to C and
+	// introduce Lt - Rt in L, R.
+	/* 0.67677 is the empirical lower bound for lpr_gain. */
+	c_gain = 8 * (*_adapt_lpr_gain - 0.67677f);
+	c_gain = c_gain > 0 ? c_gain : 0;
+	// c_gain should not be too high, not even reaching full
+	// cancellation (~ 0.50 - 0.55 at current AGC implementation), or
+	// the center will sound too narrow. */
+	c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
+	c_agc_cfk = c_gain * _cf[k];
+	_lf[k] -= c_agc_cfk;
+	_rf[k] -= c_agc_cfk;
+	_cf[k] += c_agc_cfk + c_agc_cfk;
+}
+
+void dpl2decode(float *samples, int numsamples, float *out)
+{
+	static const unsigned int FWRDURATION = 240; // FWR average duration (samples)
+	static const int cfg_delay = 0;
+	static const unsigned int fmt_freq = 48000;
+	static const unsigned int fmt_nchannels = 2; // input channels
+
+	int cur = 0;
+
+	if (olddelay != cfg_delay || oldfreq != fmt_freq)
+	{
+		done();
+		olddelay = cfg_delay;
+		oldfreq = fmt_freq;
+		dlbuflen = std::max(FWRDURATION, (fmt_freq * cfg_delay / 1000)); //+(len7000-1);
+		cyc_pos = dlbuflen - 1;
+		fwrbuf_l.resize(dlbuflen);
+		fwrbuf_r.resize(dlbuflen);
+		lf.resize(dlbuflen);
+		rf.resize(dlbuflen);
+		lr.resize(dlbuflen);
+		rr.resize(dlbuflen);
+		cf.resize(dlbuflen);
+		cr.resize(dlbuflen);
+		filter_coefs_lfe = calc_coefficients_125Hz_lowpass(fmt_freq);
+		lfe_pos = 0;
+		memset(LFE_buf, 0, sizeof(LFE_buf));
+	}
+
+	float *in = samples; // Input audio data
+	float *end = in + numsamples * fmt_nchannels; // Loop end
+
+	while (in < end)
+	{
+		const int k = cyc_pos;
+
+		const int fwr_pos = (k + FWRDURATION) % dlbuflen;
+		/* Update the full wave rectified total amplitude */
+		/* Input matrix decoder */
+		l_fwr += fabs(in[0]) - fabs(fwrbuf_l[fwr_pos]);
+		r_fwr += fabs(in[1]) - fabs(fwrbuf_r[fwr_pos]);
+		lpr_fwr += fabs(in[0] + in[1]) - fabs(fwrbuf_l[fwr_pos] + fwrbuf_r[fwr_pos]);
+		lmr_fwr += fabs(in[0] - in[1]) - fabs(fwrbuf_l[fwr_pos] - fwrbuf_r[fwr_pos]);
+
+		/* Matrix encoded 2 channel sources */
+		fwrbuf_l[k] = in[0];
+		fwrbuf_r[k] = in[1];
+		matrix_decode(in, k, 0, 1, true, dlbuflen,
+			l_fwr, r_fwr,
+			lpr_fwr, lmr_fwr,
+			&adapt_l_gain, &adapt_r_gain,
+			&adapt_lpr_gain, &adapt_lmr_gain,
+			&lf[0], &rf[0], &lr[0], &rr[0], &cf[0]);
+
+		out[cur + 0] = lf[k];
+		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);
+		lfe_pos++;
+		if (lfe_pos == len125)
+		{
+			lfe_pos = 0;
+		}
+		out[cur + 4] = lr[k];
+		out[cur + 5] = rr[k];
+		// Next sample...
+		in += 2;
+		cur += 6;
+		cyc_pos--;
+		if (cyc_pos < 0)
+		{
+			cyc_pos += dlbuflen;
+		}
+	}
+}
+
+void dpl2reset()
+{
+	olddelay = -1;
+	oldfreq = 0;
+	filter_coefs_lfe = NULL;
+}
-- 
cgit v1.2.3


From 6b87a0ef202c614e370d74291c1f754492524aaf Mon Sep 17 00:00:00 2001
From: Lioncash <mathew1800@gmail.com>
Date: Tue, 4 Feb 2014 20:43:07 -0500
Subject: Introduce a generic clamp function to clean up some similarly
 duplicated code.

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 14 ++------------
 1 file changed, 2 insertions(+), 12 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 97bdb5413d..0c9fd65400 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -13,6 +13,7 @@
 #include <stdlib.h>
 #include <string.h>
 #include "DPL2Decoder.h"
+#include "MathUtil.h"
 
 #ifndef M_PI
 #define M_PI 3.14159265358979323846
@@ -72,17 +73,6 @@ template<class T> static T firfilter(const T *buf, int pos, int len, int count,
 	return T(r1+r2);
 }
 
-template<class T> inline const T& limit(const T& val, const T& min, const T& max)
-{
-	if (val < min) {
-		return min;
-	} else if (val > max) {
-		return max;
-	} else {
-		return val;
-	}
-}
-
 /*
 // Hamming
 //                        2*pi*k
@@ -133,7 +123,7 @@ float* design_fir(unsigned int *n, float* fc, float opt)
 
 	// Sanity check
 	if(*n==0) return NULL;
-	fc[0]=limit(fc[0],float(0.001),float(1));
+	MathUtil::Clamp(fc[0],float(0.001),float(1));
 
 	float *w=(float*)calloc(sizeof(float),*n);
 
-- 
cgit v1.2.3


From 249b00c4691fb4dbd9e1c833ceccc53526babd21 Mon Sep 17 00:00:00 2001
From: Lioncash <mathew1800@gmail.com>
Date: Wed, 5 Feb 2014 04:04:35 -0500
Subject: Change the modified parameter in the Clamp function to be a pointer.

Makes it easier to identify the one being modified.
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 0c9fd65400..399541c1c9 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -123,7 +123,7 @@ float* design_fir(unsigned int *n, float* fc, float opt)
 
 	// Sanity check
 	if(*n==0) return NULL;
-	MathUtil::Clamp(fc[0],float(0.001),float(1));
+	MathUtil::Clamp(&fc[0],float(0.001),float(1));
 
 	float *w=(float*)calloc(sizeof(float),*n);
 
-- 
cgit v1.2.3


From 2afe2152712981e21d6bda6f029292ed2b1cf91e Mon Sep 17 00:00:00 2001
From: Lioncash <mathew1800@gmail.com>
Date: Mon, 17 Feb 2014 05:18:15 -0500
Subject: Convert all includes to relative paths.

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 399541c1c9..418239fae4 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -7,13 +7,14 @@
 //  * Copyright (c) 2004-2006 Milan Cutka
 //  * based on mplayer HRTF plugin by ylai
 
+#include <cmath>
+#include <cstdlib>
 #include <functional>
-#include <vector>
-#include <math.h>
-#include <stdlib.h>
 #include <string.h>
-#include "DPL2Decoder.h"
-#include "MathUtil.h"
+#include <vector>
+
+#include "AudioCommon/DPL2Decoder.h"
+#include "Common/MathUtil.h"
 
 #ifndef M_PI
 #define M_PI 3.14159265358979323846
-- 
cgit v1.2.3


From d802d392811be44d34ae9cd23f616db93e54c50f Mon Sep 17 00:00:00 2001
From: Tillmann Karras <tilkax@gmail.com>
Date: Sun, 9 Mar 2014 21:14:26 +0100
Subject: clang-modernize -use-nullptr

and s/\bNULL\b/nullptr/g for *.cpp/h/mm files not compiled on my machine
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 418239fae4..29a5f73d14 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -123,7 +123,7 @@ float* design_fir(unsigned int *n, float* fc, float opt)
 	float fc1;                               // Cutoff frequencies
 
 	// Sanity check
-	if(*n==0) return NULL;
+	if(*n==0) return nullptr;
 	MathUtil::Clamp(&fc[0],float(0.001),float(1));
 
 	float *w=(float*)calloc(sizeof(float),*n);
@@ -188,7 +188,7 @@ void done(void)
 	{
 		free(filter_coefs_lfe);
 	}
-	filter_coefs_lfe = NULL;
+	filter_coefs_lfe = nullptr;
 }
 
 float* calc_coefficients_125Hz_lowpass(int rate)
@@ -378,5 +378,5 @@ void dpl2reset()
 {
 	olddelay = -1;
 	oldfreq = 0;
-	filter_coefs_lfe = NULL;
+	filter_coefs_lfe = nullptr;
 }
-- 
cgit v1.2.3


From 31cfc73a09a8685cbab20502b4bc132e98e2feb5 Mon Sep 17 00:00:00 2001
From: Matthew Parlane <parlane@gmail.com>
Date: Tue, 11 Mar 2014 00:30:55 +1300
Subject: Fixes spacing for "for", "while", "switch" and "if" Also moved && and
 || to ends of lines instead of start. Fixed misc vertical alignments and some
 { needed newlining.

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 29a5f73d14..1cddf54600 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -123,7 +123,7 @@ float* design_fir(unsigned int *n, float* fc, float opt)
 	float fc1;                               // Cutoff frequencies
 
 	// Sanity check
-	if(*n==0) return nullptr;
+	if (*n==0) return nullptr;
 	MathUtil::Clamp(&fc[0],float(0.001),float(1));
 
 	float *w=(float*)calloc(sizeof(float),*n);
-- 
cgit v1.2.3


From 22e1aa5bb4a159d6d66a321f978917614aa36331 Mon Sep 17 00:00:00 2001
From: degasus <wickmarkus@web.de>
Date: Tue, 8 Jul 2014 14:29:26 +0200
Subject: mark all local functions as static

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 1cddf54600..883584e650 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -83,7 +83,7 @@ template<class T> static T firfilter(const T *buf, int pos, int len, int count,
 // n window length
 // w buffer for the window parameters
 */
-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 +110,7 @@ opt   beta constant used only when designing using kaiser windows
 
 returns 0 if OK, -1 if fail
 */
-float* design_fir(unsigned int *n, float* fc, float opt)
+static float* design_fir(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
@@ -165,7 +165,7 @@ float* design_fir(unsigned int *n, float* fc, float opt)
 	return w;
 }
 
-void onSeek(void)
+static void onSeek(void)
 {
 	l_fwr = r_fwr = lpr_fwr = lmr_fwr = 0;
 	std::fill(fwrbuf_l.begin(), fwrbuf_l.end(), 0.0f);
@@ -181,7 +181,7 @@ void onSeek(void)
 	memset(LFE_buf, 0, sizeof(LFE_buf));
 }
 
-void done(void)
+static void done(void)
 {
 	onSeek();
 	if (filter_coefs_lfe)
@@ -191,7 +191,7 @@ void done(void)
 	filter_coefs_lfe = nullptr;
 }
 
-float* calc_coefficients_125Hz_lowpass(int rate)
+static float* calc_coefficients_125Hz_lowpass(int rate)
 {
 	len125 = 256;
 	float f = 125.0f / (rate / 2);
@@ -204,7 +204,7 @@ float* calc_coefficients_125Hz_lowpass(int rate)
 	return coeffs;
 }
 
-float passive_lock(float x)
+static float passive_lock(float x)
 {
 	static const float MATAGCLOCK = 0.2f;  /* AGC range (around 1) where the matrix behaves passively */
 	const float x1 = x - 1;
@@ -212,7 +212,7 @@ float passive_lock(float x)
 	return x1 - x1 / (1 + ax1s * ax1s) + 1;
 }
 
-void matrix_decode(const float *in, const int k, const int il,
+static void matrix_decode(const float *in, const int k, const int il,
 	const int ir, bool decode_rear,
 	const int _dlbuflen,
 	float _l_fwr, float _r_fwr,
-- 
cgit v1.2.3


From 6d3f249dcc746cc7845ef88ddb8ce3bcc9221aca Mon Sep 17 00:00:00 2001
From: degasus <wickmarkus@web.de>
Date: Tue, 8 Jul 2014 15:58:25 +0200
Subject: mark all local variables as static

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 24 ++++++++++++------------
 1 file changed, 12 insertions(+), 12 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 883584e650..c82a160b29 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -23,18 +23,18 @@
 #define M_SQRT1_2 0.70710678118654752440
 #endif
 
-int olddelay = -1;
-unsigned int oldfreq = 0;
-unsigned int dlbuflen;
-int cyc_pos;
-float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
-std::vector<float> fwrbuf_l, fwrbuf_r;
-float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
-std::vector<float> lf, rf, lr, rr, cf, cr;
-float LFE_buf[256];
-unsigned int lfe_pos;
-float *filter_coefs_lfe;
-unsigned int len125;
+static int olddelay = -1;
+static unsigned int oldfreq = 0;
+static unsigned int dlbuflen;
+static int cyc_pos;
+static float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
+static std::vector<float> fwrbuf_l, fwrbuf_r;
+static float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
+static std::vector<float> lf, rf, lr, rr, cf, cr;
+static float LFE_buf[256];
+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)
 {
-- 
cgit v1.2.3


From 2c734726711c7dc4ae1e49c57e1a17b4c5bf6376 Mon Sep 17 00:00:00 2001
From: Lioncash <mathew1800@gmail.com>
Date: Mon, 14 Jul 2014 03:03:05 -0400
Subject: Correct the function casing for DPL2Decoder

Brings it more in-line with the rest of the codebase.
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 57 ++++++++++++++++++---------------
 1 file changed, 32 insertions(+), 25 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index c82a160b29..17b74ea85b 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -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;
-- 
cgit v1.2.3


From 2f7df4a803ebb451d5f7a1ef765220a189cb7d6b Mon Sep 17 00:00:00 2001
From: Lioncash <mathew1800@gmail.com>
Date: Sat, 30 Aug 2014 21:04:09 -0400
Subject: AudioCommon: Declare iterator variable in loop body in DPL2Decoder

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 17b74ea85b..680d193c45 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -7,6 +7,7 @@
 //  * Copyright (c) 2004-2006 Milan Cutka
 //  * based on mplayer HRTF plugin by ylai
 
+#include <algorithm>
 #include <cmath>
 #include <cstdlib>
 #include <functional>
@@ -14,6 +15,7 @@
 #include <vector>
 
 #include "AudioCommon/DPL2Decoder.h"
+#include "Common/CommonTypes.h"
 #include "Common/MathUtil.h"
 
 #ifndef M_PI
@@ -90,11 +92,10 @@ static T FIRFilter(const T *buf, int pos, int len, int count, const float *coeff
 */
 static void Hamming(int n, float* w)
 {
-	int      i;
 	float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1)
 
 	// Calculate window coefficients
-	for (i=0; i<n; i++)
+	for (int i = 0; i < n; i++)
 		*w++ = float(0.54 - 0.46*cos(k*(float)i));
 }
 
@@ -119,7 +120,6 @@ 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  i;                         // Loop index
 
 	float k1 = 2 * float(M_PI);              // 2*pi*fc1
 	float k2 = 0.5f * (float)(1 - o);        // Constant used if the filter has even length
@@ -154,7 +154,7 @@ static float* DesignFIR(unsigned int *n, float* fc, float opt)
 	}
 
 	// Create filter
-	for (i=0 ; i<end ; i++)
+	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
@@ -164,7 +164,7 @@ static float* DesignFIR(unsigned int *n, float* fc, float opt)
 
 	// Normalize gain
 	g=1/g;
-	for (i=0; i<*n; i++)
+	for (u32 i = 0; i < *n; i++)
 		w[i] *= g;
 
 	return w;
-- 
cgit v1.2.3


From f324f3f68f30f1bc578d148c63f9756c890b1052 Mon Sep 17 00:00:00 2001
From: Adam Moss <c@yotes.com>
Date: Wed, 7 Jan 2015 10:24:25 +0000
Subject: Audio: Fix subwoofer output in software 5.1 decoding

The FIR filter was only using the first sample of each input packet.
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 680d193c45..df1bb76fd5 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -361,7 +361,7 @@ void DPL2Decode(float *samples, int numsamples, float *out)
 		out[cur + 0] = lf[k];
 		out[cur + 1] = rf[k];
 		out[cur + 2] = cf[k];
-		LFE_buf[lfe_pos] = (out[0] + out[1]) / 2;
+		LFE_buf[lfe_pos] = (out[cur + 0] + out[cur + 1]) / 2;
 		out[cur + 3] = FIRFilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
 		lfe_pos++;
 		if (lfe_pos == len125)
-- 
cgit v1.2.3


From 7580069deaf0455b9dc0fe4a576b5ad5cbd8120b Mon Sep 17 00:00:00 2001
From: Adam Moss <c@yotes.com>
Date: Wed, 7 Jan 2015 11:58:02 +0000
Subject: 	Audio: Fuller subwoofer processing for software 5.1 decode

Code was only using front-left and front-right to calculate bass, but
HRTF code - which this was once based on - uses all five channels and
this sounds fuller.
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index df1bb76fd5..93a041d136 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -361,7 +361,7 @@ void DPL2Decode(float *samples, int numsamples, float *out)
 		out[cur + 0] = lf[k];
 		out[cur + 1] = rf[k];
 		out[cur + 2] = cf[k];
-		LFE_buf[lfe_pos] = (out[cur + 0] + out[cur + 1]) / 2;
+		LFE_buf[lfe_pos] = (lf[k] + rf[k] + 2.0 * cf[k] + lr[k] + rr[k]) / 2.0;
 		out[cur + 3] = FIRFilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
 		lfe_pos++;
 		if (lfe_pos == len125)
-- 
cgit v1.2.3


From b1c9f56acd967d4c06bb85c4e85c1706876f6c8d Mon Sep 17 00:00:00 2001
From: Adam Moss <c@yotes.com>
Date: Thu, 8 Jan 2015 10:48:30 +0000
Subject: 	Audio: Fix warning in DPL2 subwoofer change

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 93a041d136..da16ee4af8 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -361,7 +361,7 @@ void DPL2Decode(float *samples, int numsamples, float *out)
 		out[cur + 0] = lf[k];
 		out[cur + 1] = rf[k];
 		out[cur + 2] = cf[k];
-		LFE_buf[lfe_pos] = (lf[k] + rf[k] + 2.0 * cf[k] + lr[k] + rr[k]) / 2.0;
+		LFE_buf[lfe_pos] = (lf[k] + rf[k] + 2.0f * cf[k] + lr[k] + rr[k]) / 2.0f;
 		out[cur + 3] = FIRFilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
 		lfe_pos++;
 		if (lfe_pos == len125)
-- 
cgit v1.2.3


From 7add7b685f930c2de0e8b08c8b2a30a4b6bd092d Mon Sep 17 00:00:00 2001
From: Scott Mansell <phiren@gmail.com>
Date: Mon, 16 Feb 2015 12:59:45 +1300
Subject: Remove unneeded spaces from code.

Line now matches the style of the rest of the function.
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index da16ee4af8..ca7afc9afa 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -51,7 +51,7 @@ static _ftype_t DotProduct(int count,const T *buf,const _ftype_t *coefficients)
 	}
 
 	while (count--)
-		sum0+= *buf++ * *coefficients++;
+		sum0+=*buf++**coefficients++;
 
 	return sum0+sum1+sum2+sum3;
 }
-- 
cgit v1.2.3


From 43dfec07c92149ba71743c893d231a7efd3c25d2 Mon Sep 17 00:00:00 2001
From: Scott Mansell <phiren@gmail.com>
Date: Mon, 16 Feb 2015 13:38:11 +1300
Subject: Cleanup DotProduct function to make it more readable.

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 24 ++++++++++++++----------
 1 file changed, 14 insertions(+), 10 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index ca7afc9afa..355d721e15 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -39,21 +39,25 @@ 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)
+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)
+	int i;
+	float sum0 = 0.0f, sum1 = 0.0f, sum2 = 0.0f, sum3 = 0.0f;
+
+	// Unrolled loop
+	for (i = 0; (i + 3) < count; i += 4)
 	{
-		sum0+=buf[0]*coefficients[0];
-		sum1+=buf[1]*coefficients[1];
-		sum2+=buf[2]*coefficients[2];
-		sum3+=buf[3]*coefficients[3];
+		sum0 += buf[i + 0] * coefficients[i + 0];
+		sum1 += buf[i + 1] * coefficients[i + 1];
+		sum2 += buf[i + 2] * coefficients[i + 2];
+		sum3 += buf[i + 3] * coefficients[i + 3];
 	}
 
-	while (count--)
-		sum0+=*buf++**coefficients++;
+	// Epilogue of unrolled loop
+	for (; i < count; i++)
+		sum0 += buf[i] * coefficients[i];
 
-	return sum0+sum1+sum2+sum3;
+	return sum0 + sum1 + sum2 + sum3;
 }
 
 template<class T>
-- 
cgit v1.2.3


From 93b16a4a2d5f3e6d467d1315c461aff12852b26c Mon Sep 17 00:00:00 2001
From: Stevoisiak <Stevoisiak@gmail.com>
Date: Sun, 15 Feb 2015 14:43:31 -0500
Subject: Formatting/Whitespace Cleanup

Various fixes to formatting and whitespace
---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 35 +++++++++++++++++----------------
 1 file changed, 18 insertions(+), 17 deletions(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 355d721e15..4025a073c5 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -80,9 +80,9 @@ static T FIRFilter(const T *buf, int pos, int len, int count, const float *coeff
 	// high part of window
 	const T *ptr = &buf[pos];
 
-	float r1=DotProduct(count1,ptr,coefficients);coefficients+=count1;
-	float r2=DotProduct(count2,buf,coefficients);
-	return T(r1+r2);
+	float r1 = DotProduct(count1, ptr, coefficients); coefficients += count1;
+	float r2 = DotProduct(count2, buf, coefficients);
+	return T(r1 + r2);
 }
 
 /*
@@ -96,7 +96,7 @@ static T FIRFilter(const T *buf, int pos, int len, int count, const float *coeff
 */
 static void Hamming(int n, float* w)
 {
-	float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1)
+	float k = float(2*M_PI/((float)(n - 1))); // 2*pi/(N-1)
 
 	// Calculate window coefficients
 	for (int i = 0; i < n; i++)
@@ -122,27 +122,28 @@ returns 0 if OK, -1 if fail
 */
 static float* DesignFIR(unsigned int *n, float* fc, float opt)
 {
-	unsigned int  o   = *n & 1;              // Indicator for odd filter length
+	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
-	float g  = 0.0f;                         // Gain
+	float g = 0.0f;                          // Gain
 	float t1;                                // Temporary variables
 	float fc1;                               // Cutoff frequencies
 
 	// Sanity check
-	if (*n==0) return nullptr;
-	MathUtil::Clamp(&fc[0],float(0.001),float(1));
+	if (*n == 0)
+		return nullptr;
+	MathUtil::Clamp(&fc[0], float(0.001), float(1));
 
-	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;
+	fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1 / 2 : 0.25f;
 	k1 *= fc1;
 
 	// Low pass filter
@@ -154,20 +155,20 @@ static float* DesignFIR(unsigned int *n, float* fc, float opt)
 	if (o)
 	{
 		w[end] = fc1 * w[end] * 2.0f;
-		g=w[end];
+		g = w[end];
 	}
 
 	// Create filter
 	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
-		g += 2*w[end-i-1]; // Total gain in filter
+		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
+		g += 2*w[end - i - 1]; // Total gain in filter
 	}
 
 
 	// Normalize gain
-	g=1/g;
+	g = 1/g;
 	for (u32 i = 0; i < *n; i++)
 		w[i] *= g;
 
-- 
cgit v1.2.3


From cefcb0ace9d363b3679b4e93bcc9ec05f1e5f4f8 Mon Sep 17 00:00:00 2001
From: Tillmann Karras <tilkax@gmail.com>
Date: Mon, 18 May 2015 01:08:10 +0200
Subject: Update license headers to GPLv2+

---
 Source/Core/AudioCommon/DPL2Decoder.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'Source/Core/AudioCommon/DPL2Decoder.cpp')

diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp
index 4025a073c5..b1b63250e4 100644
--- a/Source/Core/AudioCommon/DPL2Decoder.cpp
+++ b/Source/Core/AudioCommon/DPL2Decoder.cpp
@@ -1,5 +1,5 @@
 // Copyright 2013 Dolphin Emulator Project
-// Licensed under GPLv2
+// Licensed under GPLv2+
 // Refer to the license.txt file included.
 
 // Dolby Pro Logic 2 decoder from ffdshow-tryout
-- 
cgit v1.2.3