dolphin/Source/Core/AudioCommon/Src/DPL2Decoder.cpp

401 lines
12 KiB
C++

// Copyright (C) 2003 Dolphin Project.
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License 2.0 for more details.
// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/
// Official SVN repository and contact information can be found at
// http://code.google.com/p/dolphin-emu/
// 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"
#define M_PI 3.14159265358979323846
#define M_SQRT1_2 0.70710678118654752440
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;
}