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Merge pull request #5235 from LAGonauta/fs-dplii-decoder

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Change FFDShow DPL2 decoder to FreeSurround
This commit is contained in:
Connor McLaughlin 2019-02-14 18:38:12 +10:00 committed by GitHub
commit 326d72728c
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27 changed files with 3009 additions and 440 deletions
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5
CMakeLists.txt
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@ -587,6 +587,11 @@ else()
set(PNG png)
endif()
# Using static FreeSurround from Externals
# There is no system FreeSurround library.
message(STATUS "Using static FreeSurround from Externals")
add_subdirectory(Externals/FreeSurround)
if (APPLE)
message(STATUS "Using ed25519 from Externals")
add_subdirectory(Externals/ed25519)

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set(CMAKE_CXX_STANDARD 14)
set(CMAKE_CXX_STANDARD_REQUIRED ON)
set(CMAKE_CXX_EXTENSIONS OFF)
set(SRCS
source/ChannelMaps.cpp
source/KissFFT.cpp
source/KissFFTR.cpp
source/FreeSurroundDecoder.cpp
)
add_library(FreeSurround STATIC ${SRCS})
target_include_directories(FreeSurround PUBLIC include)
target_compile_options(FreeSurround PRIVATE -w)

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Externals/FreeSurround/FreeSurround.vcxproj vendored Normal file
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<?xml version="1.0" encoding="utf-8"?>
<Project DefaultTargets="Build" ToolsVersion="15.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
<ItemGroup Label="ProjectConfigurations">
<ProjectConfiguration Include="Debug|x64">
<Configuration>Debug</Configuration>
<Platform>x64</Platform>
</ProjectConfiguration>
<ProjectConfiguration Include="Release|x64">
<Configuration>Release</Configuration>
<Platform>x64</Platform>
</ProjectConfiguration>
</ItemGroup>
<PropertyGroup Label="Globals">
<ProjectGuid>{8498F2FA-5CA6-4169-9971-DE5B1FE6132C}</ProjectGuid>
</PropertyGroup>
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
<PropertyGroup Label="Configuration">
<ConfigurationType>StaticLibrary</ConfigurationType>
<PlatformToolset>v141</PlatformToolset>
<CharacterSet>Unicode</CharacterSet>
</PropertyGroup>
<PropertyGroup Condition="'$(Configuration)'=='Debug'" Label="Configuration">
<UseDebugLibraries>true</UseDebugLibraries>
</PropertyGroup>
<PropertyGroup Condition="'$(Configuration)'=='Release'" Label="Configuration">
<UseDebugLibraries>false</UseDebugLibraries>
</PropertyGroup>
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
<ImportGroup Label="ExtensionSettings">
</ImportGroup>
<ImportGroup Label="PropertySheets">
<Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
<Import Project="..\..\Source\VSProps\Base.props" />
<Import Project="..\..\Source\VSProps\ClDisableAllWarnings.props" />
</ImportGroup>
<PropertyGroup Label="UserMacros" />
<ItemGroup>
<ClInclude Include="include\FreeSurround\ChannelMaps.h" />
<ClInclude Include="include\FreeSurround\FreeSurroundDecoder.h" />
<ClInclude Include="include\FreeSurround\KissFFT.h" />
<ClInclude Include="include\FreeSurround\KissFFTR.h" />
<ClInclude Include="include\FreeSurround\_KissFFTGuts.h" />
</ItemGroup>
<ItemGroup>
<ClCompile Include="source\ChannelMaps.cpp" />
<ClCompile Include="source\FreeSurroundDecoder.cpp" />
<ClCompile Include="source\KissFFT.cpp" />
<ClCompile Include="source\KissFFTR.cpp" />
</ItemGroup>
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
<ImportGroup Label="ExtensionTargets">
</ImportGroup>
</Project>

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<?xml version="1.0" encoding="utf-8"?>
<Project ToolsVersion="4.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
<ItemGroup>
<ClCompile Include="source\ChannelMaps.cpp">
<Filter>source</Filter>
</ClCompile>
<ClCompile Include="source\FreeSurroundDecoder.cpp">
<Filter>source</Filter>
</ClCompile>
<ClCompile Include="source\KissFFT.cpp">
<Filter>source</Filter>
</ClCompile>
<ClCompile Include="source\KissFFTR.cpp">
<Filter>source</Filter>
</ClCompile>
</ItemGroup>
<ItemGroup>
<ClInclude Include="include\FreeSurround\_KissFFTGuts.h">
<Filter>include</Filter>
</ClInclude>
<ClInclude Include="include\FreeSurround\ChannelMaps.h">
<Filter>include</Filter>
</ClInclude>
<ClInclude Include="include\FreeSurround\FreeSurroundDecoder.h">
<Filter>include</Filter>
</ClInclude>
<ClInclude Include="include\FreeSurround\KissFFT.h">
<Filter>include</Filter>
</ClInclude>
<ClInclude Include="include\FreeSurround\KissFFTR.h">
<Filter>include</Filter>
</ClInclude>
</ItemGroup>
<ItemGroup>
<Filter Include="include">
<UniqueIdentifier>{776ecb31-6d5e-489f-bac9-b91a1b202345}</UniqueIdentifier>
</Filter>
<Filter Include="source">
<UniqueIdentifier>{11345325-d67c-4a21-b2e9-c7c6c8cfc8b4}</UniqueIdentifier>
</Filter>
</ItemGroup>
</Project>

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Externals/FreeSurround/include/FreeSurround/ChannelMaps.h vendored Normal file
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/*
Copyright (C) 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.
*/
#ifndef CHANNELMAPS_H
#define CHANNELMAPS_H
#include "FreeSurroundDecoder.h"
#include <map>
#include <vector>
const int grid_res = 21; // resolution of the lookup grid
// channel allocation maps (per setup)
typedef std::vector<std::vector<float *>> alloc_lut;
extern std::map<unsigned, alloc_lut> chn_alloc;
// channel metadata maps (per setup)
extern std::map<unsigned, std::vector<float>> chn_angle;
extern std::map<unsigned, std::vector<float>> chn_xsf;
extern std::map<unsigned, std::vector<float>> chn_ysf;
extern std::map<unsigned, std::vector<channel_id>> chn_id;
#endif

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Externals/FreeSurround/include/FreeSurround/FreeSurroundDecoder.h vendored Normal file
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// 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.
#ifndef FREESURROUND_DECODER_H
#define FREESURROUND_DECODER_H
#include "KissFFTR.h"
#include <complex>
#include <vector>
typedef std::complex<double> cplx;
// Identifiers for the supported output channels (from front to back, left to
// right). The ordering here also determines the ordering of interleaved
// samples in the output signal.
typedef enum channel_id {
ci_none = 0,
ci_front_left = 1 << 1,
ci_front_center_left = 1 << 2,
ci_front_center = 1 << 3,
ci_front_center_right = 1 << 4,
ci_front_right = 1 << 5,
ci_side_front_left = 1 << 6,
ci_side_front_right = 1 << 7,
ci_side_center_left = 1 << 8,
ci_side_center_right = 1 << 9,
ci_side_back_left = 1 << 10,
ci_side_back_right = 1 << 11,
ci_back_left = 1 << 12,
ci_back_center_left = 1 << 13,
ci_back_center = 1 << 14,
ci_back_center_right = 1 << 15,
ci_back_right = 1 << 16,
ci_lfe = 1 << 31
} channel_id;
// The supported output channel setups. A channel setup is defined by the set
// of channels that are present. Here is a graphic of the cs_5point1 setup:
// http://en.wikipedia.org/wiki/File:5_1_channels_(surround_sound)_label.svg
typedef enum channel_setup {
cs_5point1 = ci_front_left | ci_front_center | ci_front_right | ci_back_left |
ci_back_right | ci_lfe,
cs_7point1 = ci_front_left | ci_front_center | ci_front_right |
ci_side_center_left | ci_side_center_right | ci_back_left |
ci_back_right | ci_lfe
} channel_setup;
// The FreeSurround decoder.
class DPL2FSDecoder {
public:
// Create an instance of the decoder.
// @param setup The output channel setup -- determines the number of output
// channels and their place in the sound field.
// @param blocksize Granularity at which data is processed by the decode()
// function. Must be a power of two and should correspond to ca. 10ms worth
// of single-channel samples (default is 4096 for 44.1Khz data). Do not make
// it shorter or longer than 5ms to 20ms since the granularity at which
// locations are decoded changes with this.
DPL2FSDecoder();
~DPL2FSDecoder();
void Init(channel_setup setup = cs_5point1, unsigned int blocksize = 4096,
unsigned int samplerate = 48000);
// Decode a chunk of stereo sound. The output is delayed by half of the
// blocksize. This function is the only one needed for straightforward
// decoding.
// @param input Contains exactly blocksize (multiplexed) stereo samples, i.e.
// 2*blocksize numbers.
// @return A pointer to an internal buffer of exactly blocksize (multiplexed)
// multichannel samples. The actual number of values depends on the number of
// output channels in the chosen channel setup.
float *decode(float *input);
// Flush the internal buffer.
void flush();
// set soundfield & rendering parameters
// for more information, see full FreeSurround source code
void set_circular_wrap(float v);
void set_shift(float v);
void set_depth(float v);
void set_focus(float v);
void set_center_image(float v);
void set_front_separation(float v);
void set_rear_separation(float v);
void set_low_cutoff(float v);
void set_high_cutoff(float v);
void set_bass_redirection(bool v);
// number of samples currently held in the buffer
unsigned int buffered();
private:
// constants
const float pi = 3.141592654f;
const float epsilon = 0.000001f;
// number of samples per input/output block, number of output channels
unsigned int N, C;
unsigned int samplerate;
// the channel setup
channel_setup setup;
bool initialized;
// parameters
// angle of the front soundstage around the listener (90\B0=default)
float circular_wrap;
// forward/backward offset of the soundstage
float shift;
// backward extension of the soundstage
float depth;
// localization of the sound events
float focus;
// presence of the center speaker
float center_image;
// front stereo separation
float front_separation;
// rear stereo separation
float rear_separation;
// LFE cutoff frequencies
float lo_cut, hi_cut;
// whether to use the LFE channel
bool use_lfe;
// FFT data structures
// left total, right total (source arrays), time-domain destination buffer
// array
std::vector<double> lt, rt, dst;
// left total / right total in frequency domain
std::vector<cplx> lf, rf;
// FFT buffers
kiss_fftr_cfg forward, inverse;
// buffers
// whether the buffer is currently empty or dirty
bool buffer_empty;
// stereo input buffer (multiplexed)
std::vector<float> inbuf;
// multichannel output buffer (multiplexed)
std::vector<float> outbuf;
// the window function, precomputed
std::vector<double> wnd;
// the signal to be constructed in every channel, in the frequency domain
// instantiate the decoder with a given channel setup and processing block
// size (in samples)
std::vector<std::vector<cplx>> signal;
// helper functions
inline float sqr(double x);
inline double amplitude(const cplx &x);
inline double phase(const cplx &x);
inline cplx polar(double a, double p);
inline float min(double a, double b);
inline float max(double a, double b);
inline float clamp(double x);
inline float sign(double x);
// get the distance of the soundfield edge, along a given angle
inline double edgedistance(double a);
// get the index (and fractional offset!) in a piecewise-linear channel
// allocation grid
int map_to_grid(double &x);
// decode a block of data and overlap-add it into outbuf
void buffered_decode(float *input);
// transform amp/phase difference space into x/y soundfield space
void transform_decode(double a, double p, double &x, double &y);
// apply a circular_wrap transformation to some position
void transform_circular_wrap(double &x, double &y, double refangle);
// apply a focus transformation to some position
void transform_focus(double &x, double &y, double focus);
};
#endif

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Externals/FreeSurround/include/FreeSurround/KissFFT.h vendored Normal file
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#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

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#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

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Externals/FreeSurround/include/FreeSurround/_KissFFTGuts.h vendored Normal file
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/*
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

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/*
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);
}

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/*
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.
*/
#include "FreeSurround/_KissFFTGuts.h"
/* The guts header contains all the multiplication and addition macros that are
defined for
fixed or floating point complex numbers. It also delares the kf_ internal
functions.
*/
static void kf_bfly2(kiss_fft_cpx *Fout, const size_t fstride,
const kiss_fft_cfg st, int m) {
kiss_fft_cpx *Fout2;
kiss_fft_cpx *tw1 = st->twiddles;
kiss_fft_cpx t;
Fout2 = Fout + m;
do {
C_FIXDIV(*Fout, 2);
C_FIXDIV(*Fout2, 2);
C_MUL(t, *Fout2, *tw1);
tw1 += fstride;
C_SUB(*Fout2, *Fout, t);
C_ADDTO(*Fout, t);
++Fout2;
++Fout;
} while (--m);
}
static void kf_bfly4(kiss_fft_cpx *Fout, const size_t fstride,
const kiss_fft_cfg st, const size_t m) {
kiss_fft_cpx *tw1, *tw2, *tw3;
kiss_fft_cpx scratch[6];
size_t k = m;
const size_t m2 = 2 * m;
const size_t m3 = 3 * m;
tw3 = tw2 = tw1 = st->twiddles;
do {
C_FIXDIV(*Fout, 4);
C_FIXDIV(Fout[m], 4);
C_FIXDIV(Fout[m2], 4);
C_FIXDIV(Fout[m3], 4);
C_MUL(scratch[0], Fout[m], *tw1);
C_MUL(scratch[1], Fout[m2], *tw2);
C_MUL(scratch[2], Fout[m3], *tw3);
C_SUB(scratch[5], *Fout, scratch[1]);
C_ADDTO(*Fout, scratch[1]);
C_ADD(scratch[3], scratch[0], scratch[2]);
C_SUB(scratch[4], scratch[0], scratch[2]);
C_SUB(Fout[m2], *Fout, scratch[3]);
tw1 += fstride;
tw2 += fstride * 2;
tw3 += fstride * 3;
C_ADDTO(*Fout, scratch[3]);
if (st->inverse) {
Fout[m].r = scratch[5].r - scratch[4].i;
Fout[m].i = scratch[5].i + scratch[4].r;
Fout[m3].r = scratch[5].r + scratch[4].i;
Fout[m3].i = scratch[5].i - scratch[4].r;
} else {
Fout[m].r = scratch[5].r + scratch[4].i;
Fout[m].i = scratch[5].i - scratch[4].r;
Fout[m3].r = scratch[5].r - scratch[4].i;
Fout[m3].i = scratch[5].i + scratch[4].r;
}
++Fout;
} while (--k);
}
static void kf_bfly3(kiss_fft_cpx *Fout, const size_t fstride,
const kiss_fft_cfg st, size_t m) {
size_t k = m;
const size_t m2 = 2 * m;
kiss_fft_cpx *tw1, *tw2;
kiss_fft_cpx scratch[5];
kiss_fft_cpx epi3;
epi3 = st->twiddles[fstride * m];
tw1 = tw2 = st->twiddles;
do {
C_FIXDIV(*Fout, 3);
C_FIXDIV(Fout[m], 3);
C_FIXDIV(Fout[m2], 3);
C_MUL(scratch[1], Fout[m], *tw1);
C_MUL(scratch[2], Fout[m2], *tw2);
C_ADD(scratch[3], scratch[1], scratch[2]);
C_SUB(scratch[0], scratch[1], scratch[2]);
tw1 += fstride;
tw2 += fstride * 2;
Fout[m].r = Fout->r - HALF_OF(scratch[3].r);
Fout[m].i = Fout->i - HALF_OF(scratch[3].i);
C_MULBYSCALAR(scratch[0], epi3.i);
C_ADDTO(*Fout, scratch[3]);
Fout[m2].r = Fout[m].r + scratch[0].i;
Fout[m2].i = Fout[m].i - scratch[0].r;
Fout[m].r -= scratch[0].i;
Fout[m].i += scratch[0].r;
++Fout;
} while (--k);
}
static void kf_bfly5(kiss_fft_cpx *Fout, const size_t fstride,
const kiss_fft_cfg st, int m) {
kiss_fft_cpx *Fout0, *Fout1, *Fout2, *Fout3, *Fout4;
int u;
kiss_fft_cpx scratch[13];
kiss_fft_cpx *twiddles = st->twiddles;
kiss_fft_cpx *tw;
kiss_fft_cpx ya, yb;
ya = twiddles[fstride * m];
yb = twiddles[fstride * 2 * m];
Fout0 = Fout;
Fout1 = Fout0 + m;
Fout2 = Fout0 + 2 * m;
Fout3 = Fout0 + 3 * m;
Fout4 = Fout0 + 4 * m;
tw = st->twiddles;
for (u = 0; u < m; ++u) {
C_FIXDIV(*Fout0, 5);
C_FIXDIV(*Fout1, 5);
C_FIXDIV(*Fout2, 5);
C_FIXDIV(*Fout3, 5);
C_FIXDIV(*Fout4, 5);
scratch[0] = *Fout0;
C_MUL(scratch[1], *Fout1, tw[u * fstride]);
C_MUL(scratch[2], *Fout2, tw[2 * u * fstride]);
C_MUL(scratch[3], *Fout3, tw[3 * u * fstride]);
C_MUL(scratch[4], *Fout4, tw[4 * u * fstride]);
C_ADD(scratch[7], scratch[1], scratch[4]);
C_SUB(scratch[10], scratch[1], scratch[4]);
C_ADD(scratch[8], scratch[2], scratch[3]);
C_SUB(scratch[9], scratch[2], scratch[3]);
Fout0->r += scratch[7].r + scratch[8].r;
Fout0->i += scratch[7].i + scratch[8].i;
scratch[5].r =
scratch[0].r + S_MUL(scratch[7].r, ya.r) + S_MUL(scratch[8].r, yb.r);
scratch[5].i =
scratch[0].i + S_MUL(scratch[7].i, ya.r) + S_MUL(scratch[8].i, yb.r);
scratch[6].r = S_MUL(scratch[10].i, ya.i) + S_MUL(scratch[9].i, yb.i);
scratch[6].i = -S_MUL(scratch[10].r, ya.i) - S_MUL(scratch[9].r, yb.i);
C_SUB(*Fout1, scratch[5], scratch[6]);
C_ADD(*Fout4, scratch[5], scratch[6]);
scratch[11].r =
scratch[0].r + S_MUL(scratch[7].r, yb.r) + S_MUL(scratch[8].r, ya.r);
scratch[11].i =
scratch[0].i + S_MUL(scratch[7].i, yb.r) + S_MUL(scratch[8].i, ya.r);
scratch[12].r = -S_MUL(scratch[10].i, yb.i) + S_MUL(scratch[9].i, ya.i);
scratch[12].i = S_MUL(scratch[10].r, yb.i) - S_MUL(scratch[9].r, ya.i);
C_ADD(*Fout2, scratch[11], scratch[12]);
C_SUB(*Fout3, scratch[11], scratch[12]);
++Fout0;
++Fout1;
++Fout2;
++Fout3;
++Fout4;
}
}
/* perform the butterfly for one stage of a mixed radix FFT */
static void kf_bfly_generic(kiss_fft_cpx *Fout, const size_t fstride,
const kiss_fft_cfg st, int m, int p) {
int u, k, q1, q;
kiss_fft_cpx *twiddles = st->twiddles;
kiss_fft_cpx t;
int Norig = st->nfft;
kiss_fft_cpx *scratch =
(kiss_fft_cpx *)KISS_FFT_TMP_ALLOC(sizeof(kiss_fft_cpx) * p);
for (u = 0; u < m; ++u) {
k = u;
for (q1 = 0; q1 < p; ++q1) {
scratch[q1] = Fout[k];
C_FIXDIV(scratch[q1], p);
k += m;
}
k = u;
for (q1 = 0; q1 < p; ++q1) {
int twidx = 0;
Fout[k] = scratch[0];
for (q = 1; q < p; ++q) {
twidx += static_cast<int>(fstride) * k;
if (twidx >= Norig)
twidx -= Norig;
C_MUL(t, scratch[q], twiddles[twidx]);
C_ADDTO(Fout[k], t);
}
k += m;
}
}
KISS_FFT_TMP_FREE(scratch);
}
static void kf_work(kiss_fft_cpx *Fout, const kiss_fft_cpx *f,
const size_t fstride, int in_stride, int *factors,
const kiss_fft_cfg st) {
kiss_fft_cpx *Fout_beg = Fout;
const int p = *factors++; /* the radix */
const int m = *factors++; /* stage's fft length/p */
const kiss_fft_cpx *Fout_end = Fout + p * m;
#ifdef _OPENMP
// use openmp extensions at the
// top-level (not recursive)
if (fstride == 1 && p <= 5) {
int k;
// execute the p different work units in different threads
#pragma omp parallel for
for (k = 0; k < p; ++k)
kf_work(Fout + k * m, f + fstride * in_stride * k, fstride * p, in_stride,
factors, st);
// all threads have joined by this point
switch (p) {
case 2:
kf_bfly2(Fout, fstride, st, m);
break;
case 3:
kf_bfly3(Fout, fstride, st, m);
break;
case 4:
kf_bfly4(Fout, fstride, st, m);
break;
case 5:
kf_bfly5(Fout, fstride, st, m);
break;
default:
kf_bfly_generic(Fout, fstride, st, m, p);
break;
}
return;
}
#endif
if (m == 1) {
do {
*Fout = *f;
f += fstride * in_stride;
} while (++Fout != Fout_end);
} else {
do {
// recursive call:
// DFT of size m*p performed by doing
// p instances of smaller DFTs of size m,
// each one takes a decimated version of the input
kf_work(Fout, f, fstride * p, in_stride, factors, st);
f += fstride * in_stride;
} while ((Fout += m) != Fout_end);
}
Fout = Fout_beg;
// recombine the p smaller DFTs
switch (p) {
case 2:
kf_bfly2(Fout, fstride, st, m);
break;
case 3:
kf_bfly3(Fout, fstride, st, m);
break;
case 4:
kf_bfly4(Fout, fstride, st, m);
break;
case 5:
kf_bfly5(Fout, fstride, st, m);
break;
default:
kf_bfly_generic(Fout, fstride, st, m, p);
break;
}
}
/* facbuf is populated by p1,m1,p2,m2, ...
where
p[i] * m[i] = m[i-1]
m0 = n */
static void kf_factor(int n, int *facbuf) {
int p = 4;
double floor_sqrt;
floor_sqrt = floor(sqrt((double)n));
/*factor out powers of 4, powers of 2, then any remaining primes */
do {
while (n % p) {
switch (p) {
case 4:
p = 2;
break;
case 2:
p = 3;
break;
default:
p += 2;
break;
}
if (p > floor_sqrt)
p = n; /* no more factors, skip to end */
}
n /= p;
*facbuf++ = p;
*facbuf++ = n;
} while (n > 1);
}
/*
*
* User-callable function to allocate all necessary storage space for the fft.
*
* The return value is a contiguous block of memory, allocated with malloc. As
* such,
* It can be freed with free(), rather than a kiss_fft-specific function.
* */
kiss_fft_cfg kiss_fft_alloc(int nfft, int inverse_fft, void *mem,
size_t *lenmem) {
kiss_fft_cfg st = NULL;
size_t memneeded = sizeof(struct kiss_fft_state) +
sizeof(kiss_fft_cpx) * (nfft - 1); /* twiddle factors*/
if (lenmem == NULL) {
st = (kiss_fft_cfg) new char[memneeded];
} else {
if (mem != NULL && *lenmem >= memneeded)
st = (kiss_fft_cfg)mem;
*lenmem = memneeded;
}
if (st) {
int i;
st->nfft = nfft;
st->inverse = inverse_fft;
for (i = 0; i < nfft; ++i) {
const double pi =
3.141592653589793238462643383279502884197169399375105820974944;
double phase = -2 * pi * i / nfft;
if (st->inverse)
phase *= -1;
kf_cexp(st->twiddles + i, phase);
}
kf_factor(nfft, st->factors);
}
return st;
}
void kiss_fft_stride(kiss_fft_cfg st, const kiss_fft_cpx *fin,
kiss_fft_cpx *fout, int in_stride) {
if (fin == fout) {
// NOTE: this is not really an in-place FFT algorithm.
// It just performs an out-of-place FFT into a temp buffer
kiss_fft_cpx *tmpbuf =
(kiss_fft_cpx *)KISS_FFT_TMP_ALLOC(sizeof(kiss_fft_cpx) * st->nfft);
kf_work(tmpbuf, fin, 1, in_stride, st->factors, st);
memcpy(fout, tmpbuf, sizeof(kiss_fft_cpx) * st->nfft);
KISS_FFT_TMP_FREE(tmpbuf);
} else {
kf_work(fout, fin, 1, in_stride, st->factors, st);
}
}
void kiss_fft(kiss_fft_cfg cfg, const kiss_fft_cpx *fin, kiss_fft_cpx *fout) {
kiss_fft_stride(cfg, fin, fout, 1);
}
void kiss_fft_cleanup(void) {
// nothing needed any more
}
int kiss_fft_next_fast_size(int n) {
while (1) {
int m = n;
while ((m % 2) == 0)
m /= 2;
while ((m % 3) == 0)
m /= 3;
while ((m % 5) == 0)
m /= 5;
if (m <= 1)
break; /* n is completely factorable by twos, threes, and fives */
n++;
}
return n;
}

185
Externals/FreeSurround/source/KissFFTR.cpp vendored Normal file
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">
@ -57,7 +57,6 @@
<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" />
@ -65,6 +64,7 @@
<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

@ -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 <algorithm>
#include <cmath>
#include <cstdlib>
#include <cstring>
#include <functional>
#include <numeric>
#include <vector>
#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<float> fwrbuf_l, fwrbuf_r;
static float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
static std::vector<float> lf, rf, lr, rr, cf, cr;
static float LFE_buf[256];
static unsigned int lfe_pos;
static std::vector<float> filter_coefs_lfe;
static unsigned int len125;
template <class T>
static float DotProduct(int count, const T* buf, const std::vector<float>& coeffs, int offset)
{
return std::inner_product(buf, buf + count, coeffs.begin() + offset, T(0));
}
template <class T>
static T FIRFilter(const T* buf, int pos, int len, int count, const std::vector<float>& 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<float> Hamming(int n)
{
std::vector<float> w(n);
float k = static_cast<float>(2.0 * M_PI / (n - 1));
// Calculate window coefficients
for (int i = 0; i < n; i++)
w[i] = static_cast<float>(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<float> 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<float>(M_PI) * fc1; // Cutoff frequency in rad/s
const float k2 = 0.5f * static_cast<float>(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<float> 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<float>(i + 1) - k2;
w[end - i - 1] = w[n - end + i] =
static_cast<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 (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<float> CalculateCoefficients125HzLowpass(int rate)
{
len125 = 256;
float f = 125.0f / (rate / 2);
std::vector<float> 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<float>(M_SQRT1_2);
if (decode_rear)
{
_lr[kr] = _rr[kr] = (l_agc - r_agc) * static_cast<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]) * static_cast<float>(M_SQRT1_2);
lmr = (in[il] - in[ir]) * static_cast<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 * PassiveLock(*_adapt_lpr_gain);
lmr_agc = lmr * PassiveLock(*_adapt_lmr_gain);
_lf[k] = (lpr_agc + lmr_agc) * static_cast<float>(M_SQRT1_2);
_rf[k] = (lpr_agc - lmr_agc) * static_cast<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 = 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();
}

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

@ -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();

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 @@ private:
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 @@ private:
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 Normal file
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 Normal file
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}