diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj b/Source/Core/AudioCommon/AudioCommon.vcxproj index 5413cc1eb6..d1dc1bde89 100644 --- a/Source/Core/AudioCommon/AudioCommon.vcxproj +++ b/Source/Core/AudioCommon/AudioCommon.vcxproj @@ -40,7 +40,6 @@ - @@ -58,7 +57,6 @@ - diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters index fdc0e186d2..44ba796b38 100644 --- a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters +++ b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters @@ -9,7 +9,6 @@ - @@ -36,7 +35,6 @@ - diff --git a/Source/Core/AudioCommon/CMakeLists.txt b/Source/Core/AudioCommon/CMakeLists.txt index e00de45877..577bdb41fb 100644 --- a/Source/Core/AudioCommon/CMakeLists.txt +++ b/Source/Core/AudioCommon/CMakeLists.txt @@ -3,7 +3,6 @@ add_library(audiocommon AudioStretcher.cpp CubebStream.cpp CubebUtils.cpp - DPL2Decoder.cpp Mixer.cpp SurroundDecoder.cpp NullSoundStream.cpp diff --git a/Source/Core/AudioCommon/CubebStream.cpp b/Source/Core/AudioCommon/CubebStream.cpp index 754db1d957..da515aa8f3 100644 --- a/Source/Core/AudioCommon/CubebStream.cpp +++ b/Source/Core/AudioCommon/CubebStream.cpp @@ -6,7 +6,6 @@ #include "AudioCommon/CubebStream.h" #include "AudioCommon/CubebUtils.h" -#include "AudioCommon/DPL2Decoder.h" #include "Common/CommonTypes.h" #include "Common/Logging/Log.h" #include "Common/Thread.h" diff --git a/Source/Core/AudioCommon/DPL2Decoder.cpp b/Source/Core/AudioCommon/DPL2Decoder.cpp deleted file mode 100644 index c48026003a..0000000000 --- a/Source/Core/AudioCommon/DPL2Decoder.cpp +++ /dev/null @@ -1,364 +0,0 @@ -// 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 -#include -#include -#include -#include -#include -#include - -#include "AudioCommon/DPL2Decoder.h" -#include "Common/CommonTypes.h" -#include "Common/MathUtil.h" - -#ifndef M_PI -#define M_PI 3.14159265358979323846 -#endif -#ifndef M_SQRT1_2 -#define M_SQRT1_2 0.70710678118654752440 -#endif - -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 fwrbuf_l, fwrbuf_r; -static float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain; -static std::vector lf, rf, lr, rr, cf, cr; -static float LFE_buf[256]; -static unsigned int lfe_pos; -static std::vector filter_coefs_lfe; -static unsigned int len125; - -template -static float DotProduct(int count, const T* buf, const std::vector& coeffs, int offset) -{ - return std::inner_product(buf, buf + count, coeffs.begin() + offset, T(0)); -} - -template -static T FIRFilter(const T* buf, int pos, int len, int count, const std::vector& coeffs) -{ - 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, coeffs, 0); - float r2 = DotProduct(count2, buf, coeffs, count1); - return T(r1 + r2); -} - -/* -// Hamming -// 2*pi*k -// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N -// N-1 -// -// n window length -// returns buffer with the window parameters -*/ -static std::vector Hamming(int n) -{ - std::vector w(n); - - float k = static_cast(2.0 * M_PI / (n - 1)); - - // Calculate window coefficients - for (int i = 0; i < n; i++) - w[i] = static_cast(0.54 - 0.46 * cos(k * i)); - - return w; -} - -// FIR filter design - -/* Design FIR filter using the Window method - -n filter length must be odd for HP and BS filters -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 buffer for the filter taps (will be n long) -*/ -static std::vector DesignFIR(unsigned int n, float fc, float opt) -{ - const unsigned int o = n & 1; // Indicator for odd filter length - const unsigned int end = ((n + 1) >> 1) - o; // Loop end - - // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2 - const float fc1 = MathUtil::Clamp(fc, 0.001f, 1.0f) / 2; - - const float k1 = 2 * static_cast(M_PI) * fc1; // Cutoff frequency in rad/s - const float k2 = 0.5f * static_cast(1 - o); // Time offset if filter has even length - float g = 0.0f; // Gain - - // Sanity check - if (n == 0) - return {}; - - // Get window coefficients - std::vector w = Hamming(n); - - // 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 (u32 i = 0; i < end; i++) - { - float t1 = static_cast(i + 1) - k2; - w[end - i - 1] = w[n - end + i] = - static_cast(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++) - w[i] *= g; - - return w; -} - -static void OnSeek() -{ - 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)); -} - -static void Done() -{ - OnSeek(); - - filter_coefs_lfe.clear(); -} - -static std::vector CalculateCoefficients125HzLowpass(int rate) -{ - len125 = 256; - float f = 125.0f / (rate / 2); - std::vector coeffs = DesignFIR(len125, f, 0); - static const float M3_01DB = 0.7071067812f; - for (unsigned int i = 0; i < len125; i++) - { - coeffs[i] *= M3_01DB; - } - return coeffs; -} - -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; - const float ax1s = fabs(x - 1) * (1.0f / MATAGCLOCK); - return x1 - x1 / (1 + ax1s * ax1s) + 1; -} - -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, 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] * PassiveLock(*_adapt_l_gain); - r_agc = in[ir] * PassiveLock(*_adapt_r_gain); - _cf[k] = (l_agc + r_agc) * static_cast(M_SQRT1_2); - if (decode_rear) - { - _lr[kr] = _rr[kr] = (l_agc - r_agc) * static_cast(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]) * static_cast(M_SQRT1_2); - lmr = (in[il] - in[ir]) * static_cast(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 * PassiveLock(*_adapt_lpr_gain); - lmr_agc = lmr * PassiveLock(*_adapt_lmr_gain); - _lf[k] = (lpr_agc + lmr_agc) * static_cast(M_SQRT1_2); - _rf[k] = (lpr_agc - lmr_agc) * static_cast(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 = CalculateCoefficients125HzLowpass(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]; - MatrixDecode(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] = (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) - { - 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.clear(); -} diff --git a/Source/Core/AudioCommon/DPL2Decoder.h b/Source/Core/AudioCommon/DPL2Decoder.h deleted file mode 100644 index 7e5678d8f2..0000000000 --- a/Source/Core/AudioCommon/DPL2Decoder.h +++ /dev/null @@ -1,8 +0,0 @@ -// Copyright 2008 Dolphin Emulator Project -// Licensed under GPLv2+ -// Refer to the license.txt file included. - -#pragma once - -void DPL2Decode(float* samples, int numsamples, float* out); -void DPL2Reset();