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https://github.com/dolphin-emu/dolphin.git
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Added FreeSurround to Externals
Also cleaned up its source code to support only 5.1 and 7.1 setups.
This commit is contained in:
LAGonauta
310
Externals/FreeSurround/source/FreeSurroundDecoder.cpp
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310
Externals/FreeSurround/source/FreeSurroundDecoder.cpp
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/*
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Copyright (C) 2007-2010 Christian Kothe
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*/
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#include "FreeSurround/FreeSurroundDecoder.h"
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#include "FreeSurround/ChannelMaps.h"
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#include <cmath>
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#undef min
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#undef max
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// FreeSurround implementation
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// DPL2FSDecoder::Init() must be called before using the decoder.
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DPL2FSDecoder::DPL2FSDecoder() {
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initialized = false;
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buffer_empty = true;
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}
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DPL2FSDecoder::~DPL2FSDecoder() {
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#pragma warning(suppress : 4150)
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delete forward;
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#pragma warning(suppress : 4150)
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delete inverse;
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}
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void DPL2FSDecoder::Init(channel_setup chsetup, unsigned int blsize,
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unsigned int sample_rate) {
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if (!initialized) {
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setup = chsetup;
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N = blsize;
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samplerate = sample_rate;
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// Initialize the parameters
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wnd = std::vector<double>(N);
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inbuf = std::vector<float>(3 * N);
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lt = std::vector<double>(N);
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rt = std::vector<double>(N);
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dst = std::vector<double>(N);
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lf = std::vector<cplx>(N / 2 + 1);
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rf = std::vector<cplx>(N / 2 + 1);
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forward = kiss_fftr_alloc(N, 0, 0, 0);
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inverse = kiss_fftr_alloc(N, 1, 0, 0);
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C = static_cast<unsigned int>(chn_alloc[setup].size());
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// Allocate per-channel buffers
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outbuf.resize((N + N / 2) * C);
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signal.resize(C, std::vector<cplx>(N));
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// Init the window function
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for (unsigned int k = 0; k < N; k++)
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wnd[k] = sqrt(0.5 * (1 - cos(2 * pi * k / N)) / N);
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// set default parameters
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set_circular_wrap(90);
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set_shift(0);
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set_depth(1);
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set_focus(0);
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set_center_image(1);
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set_front_separation(1);
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set_rear_separation(1);
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set_low_cutoff(40.0f / samplerate * 2);
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set_high_cutoff(90.0f / samplerate * 2);
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set_bass_redirection(false);
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initialized = true;
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}
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}
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// decode a stereo chunk, produces a multichannel chunk of the same size
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// (lagged)
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float *DPL2FSDecoder::decode(float *input) {
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if (initialized) {
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// append incoming data to the end of the input buffer
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memcpy(&inbuf[N], &input[0], 8 * N);
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// process first and second half, overlapped
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buffered_decode(&inbuf[0]);
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buffered_decode(&inbuf[N]);
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// shift last half of the input to the beginning (for overlapping with a
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// future block)
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memcpy(&inbuf[0], &inbuf[2 * N], 4 * N);
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buffer_empty = false;
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return &outbuf[0];
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}
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return 0;
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}
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// flush the internal buffers
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void DPL2FSDecoder::flush() {
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memset(&outbuf[0], 0, outbuf.size() * 4);
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memset(&inbuf[0], 0, inbuf.size() * 4);
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buffer_empty = true;
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}
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// number of samples currently held in the buffer
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unsigned int DPL2FSDecoder::buffered() { return buffer_empty ? 0 : N / 2; }
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// set soundfield & rendering parameters
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void DPL2FSDecoder::set_circular_wrap(float v) { circular_wrap = v; }
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void DPL2FSDecoder::set_shift(float v) { shift = v; }
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void DPL2FSDecoder::set_depth(float v) { depth = v; }
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void DPL2FSDecoder::set_focus(float v) { focus = v; }
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void DPL2FSDecoder::set_center_image(float v) { center_image = v; }
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void DPL2FSDecoder::set_front_separation(float v) { front_separation = v; }
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void DPL2FSDecoder::set_rear_separation(float v) { rear_separation = v; }
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void DPL2FSDecoder::set_low_cutoff(float v) { lo_cut = v * (N / 2); }
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void DPL2FSDecoder::set_high_cutoff(float v) { hi_cut = v * (N / 2); }
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void DPL2FSDecoder::set_bass_redirection(bool v) { use_lfe = v; }
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// helper functions
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inline float DPL2FSDecoder::sqr(double x) { return static_cast<float>(x * x); }
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inline double DPL2FSDecoder::amplitude(const cplx &x) {
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return sqrt(sqr(x.real()) + sqr(x.imag()));
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}
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inline double DPL2FSDecoder::phase(const cplx &x) {
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return atan2(x.imag(), x.real());
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}
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inline cplx DPL2FSDecoder::polar(double a, double p) {
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return cplx(a * cos(p), a * sin(p));
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}
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inline float DPL2FSDecoder::min(double a, double b) {
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return static_cast<float>(a < b ? a : b);
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}
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inline float DPL2FSDecoder::max(double a, double b) {
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return static_cast<float>(a > b ? a : b);
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}
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inline float DPL2FSDecoder::clamp(double x) { return max(-1, min(1, x)); }
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inline float DPL2FSDecoder::sign(double x) {
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return static_cast<float>(x < 0 ? -1 : (x > 0 ? 1 : 0));
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}
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// get the distance of the soundfield edge, along a given angle
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inline double DPL2FSDecoder::edgedistance(double a) {
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return min(sqrt(1 + sqr(tan(a))), sqrt(1 + sqr(1 / tan(a))));
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}
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// get the index (and fractional offset!) in a piecewise-linear channel
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// allocation grid
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int DPL2FSDecoder::map_to_grid(double &x) {
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double gp = ((x + 1) * 0.5) * (grid_res - 1),
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i = min(grid_res - 2, floor(gp));
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x = gp - i;
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return static_cast<int>(i);
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}
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// decode a block of data and overlap-add it into outbuf
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void DPL2FSDecoder::buffered_decode(float *input) {
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// demultiplex and apply window function
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for (unsigned int k = 0; k < N; k++) {
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lt[k] = wnd[k] * input[k * 2 + 0];
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rt[k] = wnd[k] * input[k * 2 + 1];
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}
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// map into spectral domain
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kiss_fftr(forward, <[0], (kiss_fft_cpx *)&lf[0]);
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kiss_fftr(forward, &rt[0], (kiss_fft_cpx *)&rf[0]);
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// compute multichannel output signal in the spectral domain
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for (unsigned int f = 1; f < N / 2; f++) {
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// get Lt/Rt amplitudes & phases
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double ampL = amplitude(lf[f]), ampR = amplitude(rf[f]);
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double phaseL = phase(lf[f]), phaseR = phase(rf[f]);
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// calculate the amplitude & phase differences
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double ampDiff =
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clamp((ampL + ampR < epsilon) ? 0 : (ampR - ampL) / (ampR + ampL));
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double phaseDiff = abs(phaseL - phaseR);
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if (phaseDiff > pi)
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phaseDiff = 2 * pi - phaseDiff;
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// decode into x/y soundfield position
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double x, y;
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transform_decode(ampDiff, phaseDiff, x, y);
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// add wrap control
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transform_circular_wrap(x, y, circular_wrap);
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// add shift control
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y = clamp(y - shift);
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// add depth control
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y = clamp(1 - (1 - y) * depth);
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// add focus control
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transform_focus(x, y, focus);
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// add crossfeed control
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x = clamp(x *
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(front_separation * (1 + y) / 2 + rear_separation * (1 - y) / 2));
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// get total signal amplitude
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double amp_total = sqrt(ampL * ampL + ampR * ampR);
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// and total L/C/R signal phases
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double phase_of[] = {
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phaseL, atan2(lf[f].imag() + rf[f].imag(), lf[f].real() + rf[f].real()),
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phaseR};
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// compute 2d channel map indexes p/q and update x/y to fractional offsets
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// in the map grid
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int p = map_to_grid(x), q = map_to_grid(y);
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// map position to channel volumes
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for (unsigned int c = 0; c < C - 1; c++) {
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// look up channel map at respective position (with bilinear
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// interpolation) and build the
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// signal
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std::vector<float *> &a = chn_alloc[setup][c];
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signal[c][f] = polar(
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amp_total * ((1 - x) * (1 - y) * a[q][p] + x * (1 - y) * a[q][p + 1] +
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(1 - x) * y * a[q + 1][p] + x * y * a[q + 1][p + 1]),
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phase_of[1 + static_cast<int>(sign(chn_xsf[setup][c]))]);
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}
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// optionally redirect bass
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if (use_lfe && f < hi_cut) {
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// level of LFE channel according to normalized frequency
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double lfe_level =
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f < lo_cut ? 1
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: 0.5 * (1 + cos(pi * (f - lo_cut) / (hi_cut - lo_cut)));
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// assign LFE channel
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signal[C - 1][f] = lfe_level * polar(amp_total, phase_of[1]);
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// subtract the signal from the other channels
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for (unsigned int c = 0; c < C - 1; c++)
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signal[c][f] *= (1 - lfe_level);
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}
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}
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// shift the last 2/3 to the first 2/3 of the output buffer
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memcpy(&outbuf[0], &outbuf[C * N / 2], N * C * 4);
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// and clear the rest
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memset(&outbuf[C * N], 0, C * 4 * N / 2);
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// backtransform each channel and overlap-add
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for (unsigned int c = 0; c < C; c++) {
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// back-transform into time domain
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kiss_fftri(inverse, (kiss_fft_cpx *)&signal[c][0], &dst[0]);
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// add the result to the last 2/3 of the output buffer, windowed (and
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// remultiplex)
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for (unsigned int k = 0; k < N; k++)
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outbuf[C * (k + N / 2) + c] += static_cast<float>(wnd[k] * dst[k]);
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}
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}
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// transform amp/phase difference space into x/y soundfield space
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void DPL2FSDecoder::transform_decode(double a, double p, double &x, double &y) {
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x = clamp(1.0047 * a + 0.46804 * a * p * p * p - 0.2042 * a * p * p * p * p +
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0.0080586 * a * p * p * p * p * p * p * p -
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0.0001526 * a * p * p * p * p * p * p * p * p * p * p -
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0.073512 * a * a * a * p - 0.2499 * a * a * a * p * p * p * p +
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0.016932 * a * a * a * p * p * p * p * p * p * p -
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0.00027707 * a * a * a * p * p * p * p * p * p * p * p * p * p +
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0.048105 * a * a * a * a * a * p * p * p * p * p * p * p -
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0.0065947 * a * a * a * a * a * p * p * p * p * p * p * p * p * p *
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p +
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0.0016006 * a * a * a * a * a * p * p * p * p * p * p * p * p * p *
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p * p -
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0.0071132 * a * a * a * a * a * a * a * p * p * p * p * p * p * p *
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p * p +
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0.0022336 * a * a * a * a * a * a * a * p * p * p * p * p * p * p *
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p * p * p * p -
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0.0004804 * a * a * a * a * a * a * a * p * p * p * p * p * p * p *
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p * p * p * p * p);
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y = clamp(
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0.98592 - 0.62237 * p + 0.077875 * p * p - 0.0026929 * p * p * p * p * p +
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0.4971 * a * a * p - 0.00032124 * a * a * p * p * p * p * p * p +
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9.2491e-006 * a * a * a * a * p * p * p * p * p * p * p * p * p * p +
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0.051549 * a * a * a * a * a * a * a * a +
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1.0727e-014 * a * a * a * a * a * a * a * a * a * a);
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}
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// apply a circular_wrap transformation to some position
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void DPL2FSDecoder::transform_circular_wrap(double &x, double &y,
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double refangle) {
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if (refangle == 90)
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return;
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refangle = refangle * pi / 180;
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double baseangle = 90 * pi / 180;
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// translate into edge-normalized polar coordinates
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double ang = atan2(x, y), len = sqrt(x * x + y * y);
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len = len / edgedistance(ang);
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// apply circular_wrap transform
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if (abs(ang) < baseangle / 2)
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// angle falls within the front region (to be enlarged)
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ang *= refangle / baseangle;
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else
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// angle falls within the rear region (to be shrunken)
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ang = pi - (-(((refangle - 2 * pi) * (pi - abs(ang)) * sign(ang)) /
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(2 * pi - baseangle)));
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// translate back into soundfield position
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len = len * edgedistance(ang);
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x = clamp(sin(ang) * len);
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y = clamp(cos(ang) * len);
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}
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// apply a focus transformation to some position
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void DPL2FSDecoder::transform_focus(double &x, double &y, double focus) {
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if (focus == 0)
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return;
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// translate into edge-normalized polar coordinates
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double ang = atan2(x, y),
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len = clamp(sqrt(x * x + y * y) / edgedistance(ang));
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// apply focus
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len = focus > 0 ? 1 - pow(1 - len, 1 + focus * 20) : pow(len, 1 - focus * 20);
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// back-transform into euclidian soundfield position
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len = len * edgedistance(ang);
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x = clamp(sin(ang) * len);
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y = clamp(cos(ang) * len);
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}
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