131 Externals/FreeSurround/include/FreeSurround/KissFFT.h
View file
@@ -0,0 +1,131 @@
#ifndef KISS_FFT_H
#define KISS_FFT_H

#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#ifdef __cplusplus
extern "C" {
#endif

// we're using doubles here...
#define kiss_fft_scalar double

/*
ATTENTION!
If you would like a :
-- a utility that will handle the caching of fft objects
-- real-only (no imaginary time component ) FFT
-- a multi-dimensional FFT
-- a command-line utility to perform ffts
-- a command-line utility to perform fast-convolution filtering
Then see kfc.h kiss_fftr.h kiss_fftnd.h fftutil.c kiss_fastfir.c
in the tools/ directory.
*/

#ifdef USE_SIMD
#include <xmmintrin.h>
#define kiss_fft_scalar __m128
#define KISS_FFT_MALLOC(nbytes) _mm_malloc(nbytes, 16)
#define KISS_FFT_FREE _mm_free
#else
#define KISS_FFT_MALLOC malloc
#define KISS_FFT_FREE free
#endif

#ifdef FIXED_POINT
#include <sys/types.h>
#if (FIXED_POINT == 32)
#define kiss_fft_scalar int32_t
#else
#define kiss_fft_scalar int16_t
#endif
#else
#ifndef kiss_fft_scalar
/* default is float */
#define kiss_fft_scalar float
#endif
#endif

typedef struct {
kiss_fft_scalar r;
kiss_fft_scalar i;
} kiss_fft_cpx;

typedef struct kiss_fft_state *kiss_fft_cfg;

/*
* kiss_fft_alloc
*
* Initialize a FFT (or IFFT) algorithm's cfg/state buffer.
*
* typical usage: kiss_fft_cfg mycfg=kiss_fft_alloc(1024,0,NULL,NULL);
*
* The return value from fft_alloc is a cfg buffer used internally
* by the fft routine or NULL.
*
* If lenmem is NULL, then kiss_fft_alloc will allocate a cfg buffer using
* malloc.
* The returned value should be free()d when done to avoid memory leaks.
*
* The state can be placed in a user supplied buffer 'mem':
* If lenmem is not NULL and mem is not NULL and *lenmem is large enough,
* then the function places the cfg in mem and the size used in *lenmem
* and returns mem.
*
* If lenmem is not NULL and ( mem is NULL or *lenmem is not large enough),
* then the function returns NULL and places the minimum cfg
* buffer size in *lenmem.
* */

kiss_fft_cfg kiss_fft_alloc(int nfft, int inverse_fft, void *mem,
size_t *lenmem);

/*
* kiss_fft(cfg,in_out_buf)
*
* Perform an FFT on a complex input buffer.
* for a forward FFT,
* fin should be f[0] , f[1] , ... ,f[nfft-1]
* fout will be F[0] , F[1] , ... ,F[nfft-1]
* Note that each element is complex and can be accessed like
f[k].r and f[k].i
* */
void kiss_fft(kiss_fft_cfg cfg, const kiss_fft_cpx *fin, kiss_fft_cpx *fout);

/*
A more generic version of the above function. It reads its input from every Nth
sample.
* */
void kiss_fft_stride(kiss_fft_cfg cfg, const kiss_fft_cpx *fin,
kiss_fft_cpx *fout, int fin_stride);

/* If kiss_fft_alloc allocated a buffer, it is one contiguous
buffer and can be simply free()d when no longer needed*/
#define kiss_fft_free free

/*
Cleans up some memory that gets managed internally. Not necessary to call, but
it might clean up
your compiler output to call this before you exit.
*/
void kiss_fft_cleanup(void);

/*
* Returns the smallest integer k, such that k>=n and k has only "fast" factors
* (2,3,5)
*/
int kiss_fft_next_fast_size(int n);

/* for real ffts, we need an even size */
#define kiss_fftr_next_fast_size_real(n) \
(kiss_fft_next_fast_size(((n) + 1) >> 1) << 1)

#ifdef __cplusplus
}
#endif

#endif
47 Externals/FreeSurround/include/FreeSurround/KissFFTR.h
View file
@@ -0,0 +1,47 @@
#ifndef KISS_FTR_H
#define KISS_FTR_H

#include "KissFFT.h"
#ifdef __cplusplus
extern "C" {
#endif

/*
Real optimized version can save about 45% cpu time vs. complex fft of a real
seq.
*/

typedef struct kiss_fftr_state *kiss_fftr_cfg;

kiss_fftr_cfg kiss_fftr_alloc(int nfft, int inverse_fft, void *mem,
size_t *lenmem);
/*
nfft must be even
If you don't care to allocate space, use mem = lenmem = NULL
*/

void kiss_fftr(kiss_fftr_cfg cfg, const kiss_fft_scalar *timedata,
kiss_fft_cpx *freqdata);
/*
input timedata has nfft scalar points
output freqdata has nfft/2+1 complex points
*/

void kiss_fftri(kiss_fftr_cfg cfg, const kiss_fft_cpx *freqdata,
kiss_fft_scalar *timedata);
/*
input freqdata has nfft/2+1 complex points
output timedata has nfft scalar points
*/

#define kiss_fftr_free free

#ifdef __cplusplus
}
#endif
#endif
202 Externals/FreeSurround/include/FreeSurround/_KissFFTGuts.h
View file
@@ -0,0 +1,202 @@
/*
Copyright (c) 2003-2010, Mark Borgerding
All rights reserved.
Redistribution and use in source and binary forms, with or without modification,
are permitted
provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice,
this list of conditions
and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of
conditions and the following disclaimer in the documentation and/or other
materials provided with
the distribution.
* Neither the author nor the names of any contributors may be used to
endorse or promote
products derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND
FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
OWNER OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY,
OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER
IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF
THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/

/* kiss_fft.h
defines kiss_fft_scalar as either short or a float type
and defines
typedef struct { kiss_fft_scalar r; kiss_fft_scalar i; }kiss_fft_cpx; */
#include "KissFFT.h"
#include <limits.h>

#define MAXFACTORS 32
/* e.g. an fft of length 128 has 4 factors
as far as kissfft is concerned
4*4*4*2
*/

struct kiss_fft_state {
int nfft;
int inverse;
int factors[2 * MAXFACTORS];
kiss_fft_cpx twiddles[1];
};

/*
Explanation of macros dealing with complex math:
C_MUL(m,a,b) : m = a*b
C_FIXDIV( c , div ) : if a fixed point impl., c /= div. noop otherwise
C_SUB( res, a,b) : res = a - b
C_SUBFROM( res , a) : res -= a
C_ADDTO( res , a) : res += a
* */
#ifdef FIXED_POINT
#if (FIXED_POINT == 32)
#define FRACBITS 31
#define SAMPPROD int64_t
#define SAMP_MAX 2147483647
#else
#define FRACBITS 15
#define SAMPPROD int32_t
#define SAMP_MAX 32767
#endif

#define SAMP_MIN -SAMP_MAX

#if defined(CHECK_OVERFLOW)
#define CHECK_OVERFLOW_OP(a, op, b) \
if ((SAMPPROD)(a)op(SAMPPROD)(b) > SAMP_MAX || \
(SAMPPROD)(a)op(SAMPPROD)(b) < SAMP_MIN) { \
fprintf(stderr, \
"WARNING:overflow @ " __FILE__ "(%d): (%d " #op " %d) = %ld\n", \
__LINE__, (a), (b), (SAMPPROD)(a)op(SAMPPROD)(b)); \
}
#endif

#define smul(a, b) ((SAMPPROD)(a) * (b))
#define sround(x) (kiss_fft_scalar)(((x) + (1 << (FRACBITS - 1))) >> FRACBITS)

#define S_MUL(a, b) sround(smul(a, b))

#define C_MUL(m, a, b) \
do { \
(m).r = sround(smul((a).r, (b).r) - smul((a).i, (b).i)); \
(m).i = sround(smul((a).r, (b).i) + smul((a).i, (b).r)); \
} while (0)

#define DIVSCALAR(x, k) (x) = sround(smul(x, SAMP_MAX / k))

#define C_FIXDIV(c, div) \
do { \
DIVSCALAR((c).r, div); \
DIVSCALAR((c).i, div); \
} while (0)

#define C_MULBYSCALAR(c, s) \
do { \
(c).r = sround(smul((c).r, s)); \
(c).i = sround(smul((c).i, s)); \
} while (0)

#else /* not FIXED_POINT*/

#define S_MUL(a, b) ((a) * (b))
#define C_MUL(m, a, b) \
do { \
(m).r = (a).r * (b).r - (a).i * (b).i; \
(m).i = (a).r * (b).i + (a).i * (b).r; \
} while (0)
#define C_FIXDIV(c, div) /* NOOP */
#define C_MULBYSCALAR(c, s) \
do { \
(c).r *= (s); \
(c).i *= (s); \
} while (0)
#endif

#ifndef CHECK_OVERFLOW_OP
#define CHECK_OVERFLOW_OP(a, op, b) /* noop */
#endif

#define C_ADD(res, a, b) \
do { \
CHECK_OVERFLOW_OP((a).r, +, (b).r) \
CHECK_OVERFLOW_OP((a).i, +, (b).i) \
(res).r = (a).r + (b).r; \
(res).i = (a).i + (b).i; \
} while (0)
#define C_SUB(res, a, b) \
do { \
CHECK_OVERFLOW_OP((a).r, -, (b).r) \
CHECK_OVERFLOW_OP((a).i, -, (b).i) \
(res).r = (a).r - (b).r; \
(res).i = (a).i - (b).i; \
} while (0)
#define C_ADDTO(res, a) \
do { \
CHECK_OVERFLOW_OP((res).r, +, (a).r) \
CHECK_OVERFLOW_OP((res).i, +, (a).i) \
(res).r += (a).r; \
(res).i += (a).i; \
} while (0)

#define C_SUBFROM(res, a) \
do { \
CHECK_OVERFLOW_OP((res).r, -, (a).r) \
CHECK_OVERFLOW_OP((res).i, -, (a).i) \
(res).r -= (a).r; \
(res).i -= (a).i; \
} while (0)

#ifdef FIXED_POINT
#define KISS_FFT_COS(phase) floor(.5 + SAMP_MAX * cos(phase))
#define KISS_FFT_SIN(phase) floor(.5 + SAMP_MAX * sin(phase))
#define HALF_OF(x) ((x) >> 1)
#elif defined(USE_SIMD)
#define KISS_FFT_COS(phase) _mm_set1_ps(cos(phase))
#define KISS_FFT_SIN(phase) _mm_set1_ps(sin(phase))
#define HALF_OF(x) ((x)*_mm_set1_ps(.5))
#else
#define KISS_FFT_COS(phase) (kiss_fft_scalar) cos(phase)
#define KISS_FFT_SIN(phase) (kiss_fft_scalar) sin(phase)
#define HALF_OF(x) ((x)*.5)
#endif

#define kf_cexp(x, phase) \
do { \
(x)->r = KISS_FFT_COS(phase); \
(x)->i = KISS_FFT_SIN(phase); \
} while (0)

/* a debugging function */
#define pcpx(c) \
fprintf(stderr, "%g + %gi\n", (double)((c)->r), (double)((c)->i))

#ifdef KISS_FFT_USE_ALLOCA
// define this to allow use of alloca instead of malloc for temporary buffers
// Temporary buffers are used in two case:
// 1. FFT sizes that have "bad" factors. i.e. not 2,3 and 5
// 2. "in-place" FFTs. Notice the quotes, since kissfft does not really do an
// in-place transform.
#include <alloca.h>
#define KISS_FFT_TMP_ALLOC(nbytes) alloca(nbytes)
#define KISS_FFT_TMP_FREE(ptr)
#else
#define KISS_FFT_TMP_ALLOC(nbytes) KISS_FFT_MALLOC(nbytes)
#define KISS_FFT_TMP_FREE(ptr) KISS_FFT_FREE(ptr)
#endif
1,148 Externals/FreeSurround/source/ChannelMaps.cpp
View file

Large diffs are not rendered by default.

310 Externals/FreeSurround/source/FreeSurroundDecoder.cpp
View file
@@ -0,0 +1,310 @@
/*
Copyright (C) 2007-2010 Christian Kothe
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; either version 2
of the License, or (at your option) any later version.
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 for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/

#include "FreeSurround/FreeSurroundDecoder.h"
#include "FreeSurround/ChannelMaps.h"
#include <cmath>

#undef min
#undef max

// FreeSurround implementation
// DPL2FSDecoder::Init() must be called before using the decoder.
DPL2FSDecoder::DPL2FSDecoder() {
initialized = false;
buffer_empty = true;
}

DPL2FSDecoder::~DPL2FSDecoder() {
#pragma warning(suppress : 4150)
delete forward;
#pragma warning(suppress : 4150)
delete inverse;
}

void DPL2FSDecoder::Init(channel_setup chsetup, unsigned int blsize,
unsigned int sample_rate) {
if (!initialized) {
setup = chsetup;
N = blsize;
samplerate = sample_rate;

// Initialize the parameters
wnd = std::vector<double>(N);
inbuf = std::vector<float>(3 * N);
lt = std::vector<double>(N);
rt = std::vector<double>(N);
dst = std::vector<double>(N);
lf = std::vector<cplx>(N / 2 + 1);
rf = std::vector<cplx>(N / 2 + 1);
forward = kiss_fftr_alloc(N, 0, 0, 0);
inverse = kiss_fftr_alloc(N, 1, 0, 0);
C = static_cast<unsigned int>(chn_alloc[setup].size());

// Allocate per-channel buffers
outbuf.resize((N + N / 2) * C);
signal.resize(C, std::vector<cplx>(N));

// Init the window function
for (unsigned int k = 0; k < N; k++)
wnd[k] = sqrt(0.5 * (1 - cos(2 * pi * k / N)) / N);

// set default parameters
set_circular_wrap(90);
set_shift(0);
set_depth(1);
set_focus(0);
set_center_image(1);
set_front_separation(1);
set_rear_separation(1);
set_low_cutoff(40.0f / samplerate * 2);
set_high_cutoff(90.0f / samplerate * 2);
set_bass_redirection(false);

initialized = true;
}
}

// decode a stereo chunk, produces a multichannel chunk of the same size
// (lagged)
float *DPL2FSDecoder::decode(float *input) {
if (initialized) {
// append incoming data to the end of the input buffer
memcpy(&inbuf[N], &input[0], 8 * N);
// process first and second half, overlapped
buffered_decode(&inbuf[0]);
buffered_decode(&inbuf[N]);
// shift last half of the input to the beginning (for overlapping with a
// future block)
memcpy(&inbuf[0], &inbuf[2 * N], 4 * N);
buffer_empty = false;
return &outbuf[0];
}
return 0;
}

// flush the internal buffers
void DPL2FSDecoder::flush() {
memset(&outbuf[0], 0, outbuf.size() * 4);
memset(&inbuf[0], 0, inbuf.size() * 4);
buffer_empty = true;
}

// number of samples currently held in the buffer
unsigned int DPL2FSDecoder::buffered() { return buffer_empty ? 0 : N / 2; }

// set soundfield & rendering parameters
void DPL2FSDecoder::set_circular_wrap(float v) { circular_wrap = v; }
void DPL2FSDecoder::set_shift(float v) { shift = v; }
void DPL2FSDecoder::set_depth(float v) { depth = v; }
void DPL2FSDecoder::set_focus(float v) { focus = v; }
void DPL2FSDecoder::set_center_image(float v) { center_image = v; }
void DPL2FSDecoder::set_front_separation(float v) { front_separation = v; }
void DPL2FSDecoder::set_rear_separation(float v) { rear_separation = v; }
void DPL2FSDecoder::set_low_cutoff(float v) { lo_cut = v * (N / 2); }
void DPL2FSDecoder::set_high_cutoff(float v) { hi_cut = v * (N / 2); }
void DPL2FSDecoder::set_bass_redirection(bool v) { use_lfe = v; }

// helper functions
inline float DPL2FSDecoder::sqr(double x) { return static_cast<float>(x * x); }
inline double DPL2FSDecoder::amplitude(const cplx &x) {
return sqrt(sqr(x.real()) + sqr(x.imag()));
}
inline double DPL2FSDecoder::phase(const cplx &x) {
return atan2(x.imag(), x.real());
}
inline cplx DPL2FSDecoder::polar(double a, double p) {
return cplx(a * cos(p), a * sin(p));
}
inline float DPL2FSDecoder::min(double a, double b) {
return static_cast<float>(a < b ? a : b);
}
inline float DPL2FSDecoder::max(double a, double b) {
return static_cast<float>(a > b ? a : b);
}
inline float DPL2FSDecoder::clamp(double x) { return max(-1, min(1, x)); }
inline float DPL2FSDecoder::sign(double x) {
return static_cast<float>(x < 0 ? -1 : (x > 0 ? 1 : 0));
}
// get the distance of the soundfield edge, along a given angle
inline double DPL2FSDecoder::edgedistance(double a) {
return min(sqrt(1 + sqr(tan(a))), sqrt(1 + sqr(1 / tan(a))));
}
// get the index (and fractional offset!) in a piecewise-linear channel
// allocation grid
int DPL2FSDecoder::map_to_grid(double &x) {
double gp = ((x + 1) * 0.5) * (grid_res - 1),
i = min(grid_res - 2, floor(gp));
x = gp - i;
return static_cast<int>(i);
}

// decode a block of data and overlap-add it into outbuf
void DPL2FSDecoder::buffered_decode(float *input) {
// demultiplex and apply window function
for (unsigned int k = 0; k < N; k++) {
lt[k] = wnd[k] * input[k * 2 + 0];
rt[k] = wnd[k] * input[k * 2 + 1];
}

// map into spectral domain
kiss_fftr(forward, &lt[0], (kiss_fft_cpx *)&lf[0]);
kiss_fftr(forward, &rt[0], (kiss_fft_cpx *)&rf[0]);

// compute multichannel output signal in the spectral domain
for (unsigned int f = 1; f < N / 2; f++) {
// get Lt/Rt amplitudes & phases
double ampL = amplitude(lf[f]), ampR = amplitude(rf[f]);
double phaseL = phase(lf[f]), phaseR = phase(rf[f]);
// calculate the amplitude & phase differences
double ampDiff =
clamp((ampL + ampR < epsilon) ? 0 : (ampR - ampL) / (ampR + ampL));
double phaseDiff = abs(phaseL - phaseR);
if (phaseDiff > pi)
phaseDiff = 2 * pi - phaseDiff;

// decode into x/y soundfield position
double x, y;
transform_decode(ampDiff, phaseDiff, x, y);
// add wrap control
transform_circular_wrap(x, y, circular_wrap);
// add shift control
y = clamp(y - shift);
// add depth control
y = clamp(1 - (1 - y) * depth);
// add focus control
transform_focus(x, y, focus);
// add crossfeed control
x = clamp(x *
(front_separation * (1 + y) / 2 + rear_separation * (1 - y) / 2));

// get total signal amplitude
double amp_total = sqrt(ampL * ampL + ampR * ampR);
// and total L/C/R signal phases
double phase_of[] = {
phaseL, atan2(lf[f].imag() + rf[f].imag(), lf[f].real() + rf[f].real()),
phaseR};
// compute 2d channel map indexes p/q and update x/y to fractional offsets
// in the map grid
int p = map_to_grid(x), q = map_to_grid(y);
// map position to channel volumes
for (unsigned int c = 0; c < C - 1; c++) {
// look up channel map at respective position (with bilinear
// interpolation) and build the
// signal
std::vector<float *> &a = chn_alloc[setup][c];
signal[c][f] = polar(
amp_total * ((1 - x) * (1 - y) * a[q][p] + x * (1 - y) * a[q][p + 1] +
(1 - x) * y * a[q + 1][p] + x * y * a[q + 1][p + 1]),
phase_of[1 + static_cast<int>(sign(chn_xsf[setup][c]))]);
}

// optionally redirect bass
if (use_lfe && f < hi_cut) {
// level of LFE channel according to normalized frequency
double lfe_level =
f < lo_cut ? 1
: 0.5 * (1 + cos(pi * (f - lo_cut) / (hi_cut - lo_cut)));
// assign LFE channel
signal[C - 1][f] = lfe_level * polar(amp_total, phase_of[1]);
// subtract the signal from the other channels
for (unsigned int c = 0; c < C - 1; c++)
signal[c][f] *= (1 - lfe_level);
}
}

// shift the last 2/3 to the first 2/3 of the output buffer
memcpy(&outbuf[0], &outbuf[C * N / 2], N * C * 4);
// and clear the rest
memset(&outbuf[C * N], 0, C * 4 * N / 2);
// backtransform each channel and overlap-add
for (unsigned int c = 0; c < C; c++) {
// back-transform into time domain
kiss_fftri(inverse, (kiss_fft_cpx *)&signal[c][0], &dst[0]);
// add the result to the last 2/3 of the output buffer, windowed (and
// remultiplex)
for (unsigned int k = 0; k < N; k++)
outbuf[C * (k + N / 2) + c] += static_cast<float>(wnd[k] * dst[k]);
}
}

// transform amp/phase difference space into x/y soundfield space
void DPL2FSDecoder::transform_decode(double a, double p, double &x, double &y) {
x = clamp(1.0047 * a + 0.46804 * a * p * p * p - 0.2042 * a * p * p * p * p +
0.0080586 * a * p * p * p * p * p * p * p -
0.0001526 * a * p * p * p * p * p * p * p * p * p * p -
0.073512 * a * a * a * p - 0.2499 * a * a * a * p * p * p * p +
0.016932 * a * a * a * p * p * p * p * p * p * p -
0.00027707 * a * a * a * p * p * p * p * p * p * p * p * p * p +
0.048105 * a * a * a * a * a * p * p * p * p * p * p * p -
0.0065947 * a * a * a * a * a * p * p * p * p * p * p * p * p * p *
p +
0.0016006 * a * a * a * a * a * p * p * p * p * p * p * p * p * p *
p * p -
0.0071132 * a * a * a * a * a * a * a * p * p * p * p * p * p * p *
p * p +
0.0022336 * a * a * a * a * a * a * a * p * p * p * p * p * p * p *
p * p * p * p -
0.0004804 * a * a * a * a * a * a * a * p * p * p * p * p * p * p *
p * p * p * p * p);
y = clamp(
0.98592 - 0.62237 * p + 0.077875 * p * p - 0.0026929 * p * p * p * p * p +
0.4971 * a * a * p - 0.00032124 * a * a * p * p * p * p * p * p +
9.2491e-006 * a * a * a * a * p * p * p * p * p * p * p * p * p * p +
0.051549 * a * a * a * a * a * a * a * a +
1.0727e-014 * a * a * a * a * a * a * a * a * a * a);
}

// apply a circular_wrap transformation to some position
void DPL2FSDecoder::transform_circular_wrap(double &x, double &y,
double refangle) {
if (refangle == 90)
return;
refangle = refangle * pi / 180;
double baseangle = 90 * pi / 180;
// translate into edge-normalized polar coordinates
double ang = atan2(x, y), len = sqrt(x * x + y * y);
len = len / edgedistance(ang);
// apply circular_wrap transform
if (abs(ang) < baseangle / 2)
// angle falls within the front region (to be enlarged)
ang *= refangle / baseangle;
else
// angle falls within the rear region (to be shrunken)
ang = pi - (-(((refangle - 2 * pi) * (pi - abs(ang)) * sign(ang)) /
(2 * pi - baseangle)));
// translate back into soundfield position
len = len * edgedistance(ang);
x = clamp(sin(ang) * len);
y = clamp(cos(ang) * len);
}

// apply a focus transformation to some position
void DPL2FSDecoder::transform_focus(double &x, double &y, double focus) {
if (focus == 0)
return;
// translate into edge-normalized polar coordinates
double ang = atan2(x, y),
len = clamp(sqrt(x * x + y * y) / edgedistance(ang));
// apply focus
len = focus > 0 ? 1 - pow(1 - len, 1 + focus * 20) : pow(len, 1 - focus * 20);
// back-transform into euclidian soundfield position
len = len * edgedistance(ang);
x = clamp(sin(ang) * len);
y = clamp(cos(ang) * len);
}
444 Externals/FreeSurround/source/KissFFT.cpp
View file

Large diffs are not rendered by default.

185 Externals/FreeSurround/source/KissFFTR.cpp
View file
@@ -0,0 +1,185 @@
/*
Copyright (c) 2003-2004, Mark Borgerding
All rights reserved.
Redistribution and use in source and binary forms, with or without modification,
are permitted
provided that the following conditions are met:
* Redistributions of source code must retain the above copyright notice,
this list of conditions
and the following disclaimer.
* Redistributions in binary form must reproduce the above copyright notice,
this list of
conditions and the following disclaimer in the documentation and/or other
materials provided with
the distribution.
* Neither the author nor the names of any contributors may be used to
endorse or promote
products derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND
FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
OWNER OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY,
OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER
IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
ARISING IN ANY WAY OUT OF
THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/

#include "FreeSurround/KissFFTR.h"
#include "FreeSurround/_KissFFTGuts.h"

struct kiss_fftr_state {
kiss_fft_cfg substate;
kiss_fft_cpx *tmpbuf;
kiss_fft_cpx *super_twiddles;
#ifdef USE_SIMD
void *pad;
#endif
};

kiss_fftr_cfg kiss_fftr_alloc(int nfft, int inverse_fft, void *mem,
size_t *lenmem) {
int i;
kiss_fftr_cfg st = NULL;
size_t subsize = 65536 * 4, memneeded = 0;

if (nfft & 1) {
fprintf(stderr, "Real FFT optimization must be even.\n");
return NULL;
}
nfft >>= 1;

kiss_fft_alloc(nfft, inverse_fft, NULL, &subsize);
memneeded = sizeof(struct kiss_fftr_state) + subsize +
sizeof(kiss_fft_cpx) * (nfft * 3 / 2);

if (lenmem == NULL) {
st = (kiss_fftr_cfg) new char[memneeded];
} else {
if (*lenmem >= memneeded)
st = (kiss_fftr_cfg)mem;
*lenmem = memneeded;
}
if (!st)
return NULL;

st->substate = (kiss_fft_cfg)(st + 1); /*just beyond kiss_fftr_state struct */
st->tmpbuf = (kiss_fft_cpx *)(((char *)st->substate) + subsize);
st->super_twiddles = st->tmpbuf + nfft;
kiss_fft_alloc(nfft, inverse_fft, st->substate, &subsize);

for (i = 0; i < nfft / 2; ++i) {
double phase =
-3.14159265358979323846264338327 * ((double)(i + 1) / nfft + .5);
if (inverse_fft)
phase *= -1;
kf_cexp(st->super_twiddles + i, phase);
}
return st;
}

void kiss_fftr(kiss_fftr_cfg st, const kiss_fft_scalar *timedata,
kiss_fft_cpx *freqdata) {
/* input buffer timedata is stored row-wise */
int k, ncfft;
kiss_fft_cpx fpnk, fpk, f1k, f2k, tw, tdc;

if (st->substate->inverse) {
fprintf(stderr, "kiss fft usage error: improper alloc\n");
exit(1);
}

ncfft = st->substate->nfft;

/*perform the parallel fft of two real signals packed in real,imag*/
kiss_fft(st->substate, (const kiss_fft_cpx *)timedata, st->tmpbuf);
/* The real part of the DC element of the frequency spectrum in st->tmpbuf
* contains the sum of the even-numbered elements of the input time sequence
* The imag part is the sum of the odd-numbered elements
*
* The sum of tdc.r and tdc.i is the sum of the input time sequence.
* yielding DC of input time sequence
* The difference of tdc.r - tdc.i is the sum of the input (dot product)
* [1,-1,1,-1...
* yielding Nyquist bin of input time sequence
*/

tdc.r = st->tmpbuf[0].r;
tdc.i = st->tmpbuf[0].i;
C_FIXDIV(tdc, 2);
CHECK_OVERFLOW_OP(tdc.r, +, tdc.i);
CHECK_OVERFLOW_OP(tdc.r, -, tdc.i);
freqdata[0].r = tdc.r + tdc.i;
freqdata[ncfft].r = tdc.r - tdc.i;
#ifdef USE_SIMD
freqdata[ncfft].i = freqdata[0].i = _mm_set1_ps(0);
#else
freqdata[ncfft].i = freqdata[0].i = 0;
#endif

for (k = 1; k <= ncfft / 2; ++k) {
fpk = st->tmpbuf[k];
fpnk.r = st->tmpbuf[ncfft - k].r;
fpnk.i = -st->tmpbuf[ncfft - k].i;
C_FIXDIV(fpk, 2);
C_FIXDIV(fpnk, 2);

C_ADD(f1k, fpk, fpnk);
C_SUB(f2k, fpk, fpnk);
C_MUL(tw, f2k, st->super_twiddles[k - 1]);

freqdata[k].r = HALF_OF(f1k.r + tw.r);
freqdata[k].i = HALF_OF(f1k.i + tw.i);
freqdata[ncfft - k].r = HALF_OF(f1k.r - tw.r);
freqdata[ncfft - k].i = HALF_OF(tw.i - f1k.i);
}
}

void kiss_fftri(kiss_fftr_cfg st, const kiss_fft_cpx *freqdata,
kiss_fft_scalar *timedata) {
/* input buffer timedata is stored row-wise */
int k, ncfft;

if (st->substate->inverse == 0) {
fprintf(stderr, "kiss fft usage error: improper alloc\n");
exit(1);
}

ncfft = st->substate->nfft;

st->tmpbuf[0].r = freqdata[0].r + freqdata[ncfft].r;
st->tmpbuf[0].i = freqdata[0].r - freqdata[ncfft].r;
C_FIXDIV(st->tmpbuf[0], 2);

for (k = 1; k <= ncfft / 2; ++k) {
kiss_fft_cpx fk, fnkc, fek, fok, tmp;
fk = freqdata[k];
fnkc.r = freqdata[ncfft - k].r;
fnkc.i = -freqdata[ncfft - k].i;
C_FIXDIV(fk, 2);
C_FIXDIV(fnkc, 2);

C_ADD(fek, fk, fnkc);
C_SUB(tmp, fk, fnkc);
C_MUL(fok, tmp, st->super_twiddles[k - 1]);
C_ADD(st->tmpbuf[k], fek, fok);
C_SUB(st->tmpbuf[ncfft - k], fek, fok);
#ifdef USE_SIMD
st->tmpbuf[ncfft - k].i *= _mm_set1_ps(-1.0);
#else
st->tmpbuf[ncfft - k].i *= -1;
#endif
}
kiss_fft(st->substate, st->tmpbuf, (kiss_fft_cpx *)timedata);
}
7 Source/Core/AudioCommon/AudioCommon.vcxproj
View file
@@ -40,11 +40,11 @@
<ClCompile Include="AudioStretcher.cpp" />
<ClCompile Include="CubebStream.cpp" />
<ClCompile Include="CubebUtils.cpp" />
<ClCompile Include="DPL2Decoder.cpp" />
<ClCompile Include="Mixer.cpp" />
<ClCompile Include="NullSoundStream.cpp" />
<ClCompile Include="OpenALStream.cpp" />
<ClCompile Include="WASAPIStream.cpp" />
<ClCompile Include="SurroundDecoder.cpp" />
<ClCompile Include="WaveFile.cpp" />
<ClCompile Include="XAudio2Stream.cpp" />
<ClCompile Include="XAudio2_7Stream.cpp">
@@ -57,14 +57,14 @@
<ClInclude Include="AudioStretcher.h" />
<ClInclude Include="CubebStream.h" />
<ClInclude Include="CubebUtils.h" />
<ClInclude Include="DPL2Decoder.h" />
<ClInclude Include="Mixer.h" />
<ClInclude Include="NullSoundStream.h" />
<ClInclude Include="OpenALStream.h" />
<ClInclude Include="OpenSLESStream.h" />
<ClInclude Include="PulseAudioStream.h" />
<ClInclude Include="SoundStream.h" />
<ClInclude Include="WASAPIStream.h" />
<ClInclude Include="SurroundDecoder.h" />
<ClInclude Include="WaveFile.h" />
<ClInclude Include="XAudio2Stream.h" />
<ClInclude Include="XAudio2_7Stream.h" />
@@ -79,6 +79,9 @@
<ProjectReference Include="$(CoreDir)Common\Common.vcxproj">
<Project>{2e6c348c-c75c-4d94-8d1e-9c1fcbf3efe4}</Project>
</ProjectReference>
<ProjectReference Include="$(ExternalsDir)FreeSurround\FreeSurround.vcxproj">
<Project>{8498f2fa-5ca6-4169-9971-de5b1fe6132c}</Project>
</ProjectReference>
</ItemGroup>
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
<ImportGroup Label="ExtensionTargets">
4 Source/Core/AudioCommon/AudioCommon.vcxproj.filters
View file
@@ -9,7 +9,6 @@
<ClCompile Include="AudioCommon.cpp" />
<ClCompile Include="AudioStretcher.cpp" />
<ClCompile Include="CubebUtils.cpp" />
<ClCompile Include="DPL2Decoder.cpp" />
<ClCompile Include="Mixer.cpp" />
<ClCompile Include="WaveFile.cpp" />
<ClCompile Include="NullSoundStream.cpp">
@@ -30,12 +29,12 @@
<ClCompile Include="WASAPIStream.cpp">
<Filter>SoundStreams</Filter>
</ClCompile>
<ClCompile Include="SurroundDecoder.cpp" />
</ItemGroup>
<ItemGroup>
<ClInclude Include="AudioCommon.h" />
<ClInclude Include="AudioStretcher.h" />
<ClInclude Include="CubebUtils.h" />
<ClInclude Include="DPL2Decoder.h" />
<ClInclude Include="Mixer.h" />
<ClInclude Include="WaveFile.h" />
<ClInclude Include="NullSoundStream.h">
@@ -68,6 +67,7 @@
<ClInclude Include="WASAPIStream.h">
<Filter>SoundStreams</Filter>
</ClInclude>
<ClInclude Include="SurroundDecoder.h" />
</ItemGroup>
<ItemGroup>
<Text Include="CMakeLists.txt" />
4 Source/Core/AudioCommon/CMakeLists.txt
View file
@@ -3,8 +3,8 @@ add_library(audiocommon
AudioStretcher.cpp
CubebStream.cpp
CubebUtils.cpp
DPL2Decoder.cpp
Mixer.cpp
SurroundDecoder.cpp
NullSoundStream.cpp
WaveFile.cpp
)
@@ -69,4 +69,4 @@ if(WIN32)
endif()
endif()

target_link_libraries(audiocommon PRIVATE cubeb SoundTouch)
target_link_libraries(audiocommon PRIVATE cubeb SoundTouch FreeSurround)
1 Source/Core/AudioCommon/CubebStream.cpp
View file
@@ -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"
364 Source/Core/AudioCommon/DPL2Decoder.cpp
View file

This file was deleted.

8 Source/Core/AudioCommon/DPL2Decoder.h
View file

This file was deleted.

26 Source/Core/AudioCommon/Mixer.cpp
View file
@@ -7,7 +7,6 @@
#include <cmath>
#include <cstring>

#include "AudioCommon/DPL2Decoder.h"
#include "Common/ChunkFile.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
@@ -16,10 +15,10 @@
#include "Core/ConfigManager.h"

Mixer::Mixer(unsigned int BackendSampleRate)
: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate)
: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate),
m_surround_decoder(BackendSampleRate, SURROUND_BLOCK_SIZE)
{
INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
DPL2Reset();
}

Mixer::~Mixer()
@@ -167,20 +166,23 @@ unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
if (!num_samples)
return 0;

memset(samples, 0, num_samples * 6 * sizeof(float));
memset(samples, 0, num_samples * SURROUND_CHANNELS * sizeof(float));

// Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads and
// writes.
unsigned int available_samples = Mix(m_scratch_buffer.data(), num_samples);
for (size_t i = 0; i < static_cast<size_t>(available_samples) * 2; ++i)
size_t needed_frames = m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);

// Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads
// and writes.
size_t available_frames = Mix(m_scratch_buffer.data(), static_cast<u32>(needed_frames));
if (available_frames != needed_frames)
{
m_float_conversion_buffer[i] =
m_scratch_buffer[i] / static_cast<float>(std::numeric_limits<short>::max());
ERROR_LOG(AUDIO, "Error decoding surround frames.");
return 0;
}

DPL2Decode(m_float_conversion_buffer.data(), available_samples, samples);
m_surround_decoder.PutFrames(m_scratch_buffer.data(), needed_frames);
m_surround_decoder.ReceiveFrames(samples, num_samples);

return available_samples;
return num_samples;
}

void Mixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples)
6 Source/Core/AudioCommon/Mixer.h
View file
@@ -8,6 +8,7 @@
#include <atomic>

#include "AudioCommon/AudioStretcher.h"
#include "AudioCommon/SurroundDecoder.h"
#include "AudioCommon/WaveFile.h"
#include "Common/CommonTypes.h"

@@ -52,6 +53,9 @@ class Mixer final
static constexpr float CONTROL_FACTOR = 0.2f;
static constexpr u32 CONTROL_AVG = 32; // In freq_shift per FIFO size offset

const unsigned int SURROUND_CHANNELS = 6;
const unsigned int SURROUND_BLOCK_SIZE = 512;

class MixerFifo final
{
public:
@@ -86,8 +90,8 @@ class Mixer final

bool m_is_stretching = false;
AudioCommon::AudioStretcher m_stretcher;
AudioCommon::SurroundDecoder m_surround_decoder;
std::array<short, MAX_SAMPLES * 2> m_scratch_buffer;
std::array<float, MAX_SAMPLES * 2> m_float_conversion_buffer;

WaveFileWriter m_wave_writer_dtk;
WaveFileWriter m_wave_writer_dsp;
40 Source/Core/AudioCommon/OpenALStream.cpp
View file
@@ -246,12 +246,6 @@ void OpenALStream::SoundLoop()
frames_per_buffer = OAL_MAX_FRAMES;
}

// DPL2 needs a minimum number of samples to work (FWRDURATION)
if (use_surround && frames_per_buffer < 240)
{
frames_per_buffer = 240;
}

INFO_LOG(AUDIO, "Using %d buffers, each with %d audio frames for a total of %d.", OAL_BUFFERS,
frames_per_buffer, frames_per_buffer * OAL_BUFFERS);

@@ -312,15 +306,6 @@ void OpenALStream::SoundLoop()
if (rendered_frames < min_frames)
continue;

// zero-out the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output. Sadly there is not a 5.0
// AL_FORMAT_50CHN32 to make this super-explicit.
// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
for (u32 i = 0; i < rendered_frames; ++i)
{
dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
}

if (float32_capable)
{
palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN32, dpl2.data(),
@@ -332,14 +317,11 @@ void OpenALStream::SoundLoop()

for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
{
// For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1.
// Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
// fix the decoder or implement a limiter.
dpl2[i] = dpl2[i] * (INT64_C(1) << 31);
if (dpl2[i] > INT_MAX)
surround_int32[i] = INT_MAX;
else if (dpl2[i] < INT_MIN)
surround_int32[i] = INT_MIN;
dpl2[i] = dpl2[i] * std::numeric_limits<int>::max();
if (dpl2[i] > std::numeric_limits<int>::max())
surround_int32[i] = std::numeric_limits<int>::max();
else if (dpl2[i] < std::numeric_limits<int>::min())
surround_int32[i] = std::numeric_limits<int>::min();
else
surround_int32[i] = static_cast<int>(dpl2[i]);
}
@@ -353,13 +335,13 @@ void OpenALStream::SoundLoop()

for (u32 i = 0; i < rendered_frames * SURROUND_CHANNELS; ++i)
{
dpl2[i] = dpl2[i] * (1 << 15);
if (dpl2[i] > SHRT_MAX)
surround_short[i] = SHRT_MAX;
else if (dpl2[i] < SHRT_MIN)
surround_short[i] = SHRT_MIN;
dpl2[i] = dpl2[i] * std::numeric_limits<short>::max();
if (dpl2[i] > std::numeric_limits<short>::max())
surround_short[i] = std::numeric_limits<short>::max();
else if (dpl2[i] < std::numeric_limits<short>::min())
surround_short[i] = std::numeric_limits<short>::min();
else
surround_short[i] = static_cast<int>(dpl2[i]);
surround_short[i] = static_cast<short>(dpl2[i]);
}

palBufferData(m_buffers[next_buffer], AL_FORMAT_51CHN16, surround_short.data(),
26 Source/Core/AudioCommon/PulseAudioStream.cpp
View file
@@ -22,7 +22,7 @@ PulseAudio::PulseAudio() : m_thread(), m_run_thread()
bool PulseAudio::Init()
{
m_stereo = !SConfig::GetInstance().bDPL2Decoder;
m_channels = m_stereo ? 2 : 5; // will tell PA we use a Stereo or 5.0 channel setup
m_channels = m_stereo ? 2 : 6; // will tell PA we use a Stereo or 5.0 channel setup

NOTICE_LOG(AUDIO, "PulseAudio backend using %d channels", m_channels);

@@ -96,12 +96,13 @@ bool PulseAudio::PulseInit()
m_bytespersample = sizeof(float);

channel_map_p = &channel_map; // explicit channel map:
channel_map.channels = 5;
channel_map.channels = 6;
channel_map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
channel_map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
channel_map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
channel_map.map[3] = PA_CHANNEL_POSITION_REAR_LEFT;
channel_map.map[4] = PA_CHANNEL_POSITION_REAR_RIGHT;
channel_map.map[3] = PA_CHANNEL_POSITION_LFE;
channel_map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
channel_map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
}
ss.channels = m_channels;
ss.rate = m_mixer->GetSampleRate();
@@ -185,22 +186,9 @@ void PulseAudio::WriteCallback(pa_stream* s, size_t length)
}
else
{
if (m_channels == 5) // Extract dpl2/5.0 Surround
if (m_channels == 6) // Extract dpl2/5.1 Surround
{
float floatbuffer_6chan[frames * 6];
m_mixer->MixSurround(floatbuffer_6chan, frames);

// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
// Discard the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output.
const int dpl2_to_5chan[] = {0, 1, 2, 4, 5};
for (int i = 0; i < frames; ++i)
{
for (int j = 0; j < m_channels; ++j)
{
((float*)buffer)[m_channels * i + j] = floatbuffer_6chan[6 * i + dpl2_to_5chan[j]];
}
}
m_mixer->MixSurround((float*)buffer, frames);
}
else
{
93 Source/Core/AudioCommon/SurroundDecoder.cpp
View file
@@ -0,0 +1,93 @@
// Copyright 2017 Dolphin Emulator Project
// Licensed under GPLv2+
// Refer to the license.txt file included.

#include <FreeSurround/FreeSurroundDecoder.h>
#include <limits>

#include "AudioCommon/SurroundDecoder.h"

namespace AudioCommon
{
constexpr size_t STEREO_CHANNELS = 2;
constexpr size_t SURROUND_CHANNELS = 6;

SurroundDecoder::SurroundDecoder(u32 sample_rate, u32 frame_block_size)
: m_sample_rate(sample_rate), m_frame_block_size(frame_block_size)
{
m_fsdecoder = std::make_unique<DPL2FSDecoder>();
m_fsdecoder->Init(cs_5point1, m_frame_block_size, m_sample_rate);
}

SurroundDecoder::~SurroundDecoder() = default;

void SurroundDecoder::Clear()
{
m_fsdecoder->flush();
m_decoded_fifo.clear();
}

// Currently only 6 channels are supported.
size_t SurroundDecoder::QueryFramesNeededForSurroundOutput(const size_t output_frames) const
{
if (m_decoded_fifo.size() < output_frames * SURROUND_CHANNELS)
{
// Output stereo frames needed to have at least the desired number of surround frames
size_t frames_needed = output_frames - m_decoded_fifo.size() / SURROUND_CHANNELS;
return frames_needed + m_frame_block_size - frames_needed % m_frame_block_size;
}

return 0;
}

// Receive and decode samples
void SurroundDecoder::PutFrames(const short* in, const size_t num_frames_in)
{
// Maybe check if it is really power-of-2?
s64 remaining_frames = static_cast<s64>(num_frames_in);
size_t frame_index = 0;

while (remaining_frames > 0)
{
// Convert to float
for (size_t i = 0, end = m_frame_block_size * STEREO_CHANNELS; i < end; ++i)
{
m_float_conversion_buffer[i] = in[i + frame_index * STEREO_CHANNELS] /
static_cast<float>(std::numeric_limits<short>::max());
}

// Decode
const float* dpl2_fs = m_fsdecoder->decode(m_float_conversion_buffer.data());

// Add to ring buffer and fix channel mapping
// Maybe modify FreeSurround to output the correct mapping?
// FreeSurround:
// FL | FC | FR | BL | BR | LFE
// Most backends:
// FL | FR | FC | LFE | BL | BR
for (size_t i = 0; i < m_frame_block_size; ++i)
{
m_decoded_fifo.push(dpl2_fs[i * SURROUND_CHANNELS + 0]); // LEFTFRONT
m_decoded_fifo.push(dpl2_fs[i * SURROUND_CHANNELS + 2]); // RIGHTFRONT
m_decoded_fifo.push(dpl2_fs[i * SURROUND_CHANNELS + 1]); // CENTREFRONT
m_decoded_fifo.push(dpl2_fs[i * SURROUND_CHANNELS + 5]); // sub/lfe
m_decoded_fifo.push(dpl2_fs[i * SURROUND_CHANNELS + 3]); // LEFTREAR
m_decoded_fifo.push(dpl2_fs[i * SURROUND_CHANNELS + 4]); // RIGHTREAR
}

remaining_frames = remaining_frames - static_cast<int>(m_frame_block_size);
frame_index = frame_index + m_frame_block_size;
}
}

void SurroundDecoder::ReceiveFrames(float* out, const size_t num_frames_out)
{
// Copy to output array with desired num_frames_out
for (size_t i = 0, num_samples_output = num_frames_out * SURROUND_CHANNELS;
i < num_samples_output; ++i)
{
out[i] = m_decoded_fifo.pop_front();
}
}

} // namespace AudioCommon
36 Source/Core/AudioCommon/SurroundDecoder.h
View file
@@ -0,0 +1,36 @@
// Copyright 2017 Dolphin Emulator Project
// Licensed under GPLv2+
// Refer to the license.txt file included.

#pragma once

#include <array>
#include <memory>

#include "Common/CommonTypes.h"
#include "Common/FixedSizeQueue.h"

class DPL2FSDecoder;

namespace AudioCommon
{
class SurroundDecoder
{
public:
explicit SurroundDecoder(u32 sample_rate, u32 frame_block_size);
~SurroundDecoder();
size_t QueryFramesNeededForSurroundOutput(const size_t output_frames) const;
void PutFrames(const short* in, const size_t num_frames_in);
void ReceiveFrames(float* out, const size_t num_frames_out);
void Clear();

private:
u32 m_sample_rate;
u32 m_frame_block_size;

std::unique_ptr<DPL2FSDecoder> m_fsdecoder;
std::array<float, 32768> m_float_conversion_buffer;
FixedSizeQueue<float, 32768> m_decoded_fifo;
};

} // AudioCommon
1 Source/VSProps/Base.props
View file
@@ -36,6 +36,7 @@
<AdditionalIncludeDirectories>$(ExternalsDir);%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(ExternalsDir)Bochs_disasm;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(ExternalsDir)cpp-optparse;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(ExternalsDir)FreeSurround\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(ExternalsDir)cubeb\include;$(ExternalsDir)cubeb\msvc;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(ExternalsDir)curl\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(ExternalsDir)enet\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
7 Source/dolphin-emu.sln
View file
@@ -83,6 +83,8 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "ed25519", "..\externals\ed2
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "Updater", "Core\Updater\Updater.vcxproj", "{E4BECBAB-9C6E-41AB-BB56-F9D70AB6BE03}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "FreeSurround", "..\Externals\FreeSurround\FreeSurround.vcxproj", "{8498F2FA-5CA6-4169-9971-DE5B1FE6132C}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "discord-rpc", "..\Externals\discord-rpc\src\discord-rpc.vcxproj", "{4482FD2A-EC43-3FFB-AC20-2E5C54B05EAD}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "minizip", "..\Externals\minizip\minizip.vcxproj", "{23114507-079A-4418-9707-CFA81A03CA99}"
@@ -247,6 +249,10 @@ Global
{E4BECBAB-9C6E-41AB-BB56-F9D70AB6BE03}.Debug|x64.Build.0 = Debug|x64
{E4BECBAB-9C6E-41AB-BB56-F9D70AB6BE03}.Release|x64.ActiveCfg = Release|x64
{E4BECBAB-9C6E-41AB-BB56-F9D70AB6BE03}.Release|x64.Build.0 = Release|x64
{8498F2FA-5CA6-4169-9971-DE5B1FE6132C}.Debug|x64.ActiveCfg = Debug|x64
{8498F2FA-5CA6-4169-9971-DE5B1FE6132C}.Debug|x64.Build.0 = Debug|x64
{8498F2FA-5CA6-4169-9971-DE5B1FE6132C}.Release|x64.ActiveCfg = Release|x64
{8498F2FA-5CA6-4169-9971-DE5B1FE6132C}.Release|x64.Build.0 = Release|x64
{4482FD2A-EC43-3FFB-AC20-2E5C54B05EAD}.Debug|x64.ActiveCfg = Debug|x64
{4482FD2A-EC43-3FFB-AC20-2E5C54B05EAD}.Debug|x64.Build.0 = Debug|x64
{4482FD2A-EC43-3FFB-AC20-2E5C54B05EAD}.Release|x64.ActiveCfg = Release|x64
@@ -296,6 +302,7 @@ Global
{38FEE76F-F347-484B-949C-B4649381CFFB} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{2C0D058E-DE35-4471-AD99-E68A2CAF9E18} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{5BDF4B91-1491-4FB0-BC27-78E9A8E97DC3} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{8498F2FA-5CA6-4169-9971-DE5B1FE6132C} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{4482FD2A-EC43-3FFB-AC20-2E5C54B05EAD} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{23114507-079A-4418-9707-CFA81A03CA99} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{4C3B2264-EA73-4A7B-9CFE-65B0FD635EBB} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}