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Update SoundTouch to 1.9.0. 

It claimed to be 1.7.1 but it had a mixture from various
versions. It was hard to update as everything in the top directory
so I used upstream's way to organize files. I renamed include to
soundtouch since I did not want to #ifdef that for windows.
.
Wavfile.h is a private header so I used the private path instead of
moving the file over. This changed 3 files in the plugin folder.
This commit is contained in:
Miguel A. Colón Vélez 2015-08-22 10:16:27 -04:00
parent 9f2642a714
commit 09c8a41294
52 changed files with 3677 additions and 2606 deletions
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349
3rdparty/soundtouch/3dnow_win.cpp vendored
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@ -1,349 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Win32 version of the AMD 3DNow! optimized routines for AMD K6-2/Athlon
/// processors. All 3DNow! optimized functions have been gathered into this
/// single source code file, regardless to their class or original source code
/// file, in order to ease porting the library to other compiler and processor
/// platforms.
///
/// By the way; the performance gain depends heavily on the CPU generation: On
/// K6-2 these routines provided speed-up of even 2.4 times, while on Athlon the
/// difference to the original routines stayed at unremarkable 8%! Such a small
/// improvement on Athlon is due to 3DNow can perform only two operations in
/// parallel, and obviously also the Athlon FPU is doing a very good job with
/// the standard C floating point routines! Here these routines are anyway,
/// although it might not be worth the effort to convert these to GCC platform,
/// for Athlon CPU at least. The situation is different regarding the SSE
/// optimizations though, thanks to the four parallel operations of SSE that
/// already make a difference.
///
/// This file is to be compiled in Windows platform with Microsoft Visual C++
/// Compiler. Please see '3dnow_gcc.cpp' for the gcc compiler version for all
/// GNU platforms (if file supplied).
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support 3DNow! instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// perform a search with keywords "processor pack".
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
// File revision : $Revision: 4 $
//
// $Id: 3dnow_win.cpp 63 2009-02-21 16:00:14Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
#ifndef WIN32
#error "wrong platform - this source code file is exclusively for Win32 platform"
#endif
using namespace soundtouch;
#ifdef ALLOW_3DNOW
// 3DNow! routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'TDStretch3DNow'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
// Calculates cross correlation of two buffers
double TDStretch3DNow::calcCrossCorrStereo(const float *pV1, const float *pV2) const
{
int overlapLengthLocal = overlapLength;
float corr = 0;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
/*
c-pseudocode:
corr = 0;
for (i = 0; i < overlapLength / 4; i ++)
{
corr += pV1[0] * pV2[0];
pV1[1] * pV2[1];
pV1[2] * pV2[2];
pV1[3] * pV2[3];
pV1[4] * pV2[4];
pV1[5] * pV2[5];
pV1[6] * pV2[6];
pV1[7] * pV2[7];
pV1 += 8;
pV2 += 8;
}
*/
_asm
{
// give prefetch hints to CPU of what data are to be needed soonish.
// give more aggressive hints on pV1 as that changes more between different calls
// while pV2 stays the same.
prefetch [pV1]
prefetch [pV2]
prefetch [pV1 + 32]
mov eax, dword ptr pV2
mov ebx, dword ptr pV1
pxor mm0, mm0
mov ecx, overlapLengthLocal
shr ecx, 2 // div by four
loop1:
movq mm1, [eax]
prefetch [eax + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm1, [ebx]
prefetch [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
movq mm2, [eax + 8]
pfadd mm0, mm1
pfmul mm2, [ebx + 8]
movq mm3, [eax + 16]
pfadd mm0, mm2
pfmul mm3, [ebx + 16]
movq mm4, [eax + 24]
pfadd mm0, mm3
pfmul mm4, [ebx + 24]
add eax, 32
pfadd mm0, mm4
add ebx, 32
dec ecx
jnz loop1
// add halfs of mm0 together and return the result.
// note: mm1 is used as a dummy parameter only, we actually don't care about it's value
pfacc mm0, mm1
movd corr, mm0
femms
}
return corr;
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilter3DNow::FIRFilter3DNow() : FIRFilter()
{
filterCoeffsUnalign = NULL;
filterCoeffsAlign = NULL;
}
FIRFilter3DNow::~FIRFilter3DNow()
{
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = NULL;
filterCoeffsAlign = NULL;
}
// (overloaded) Calculates filter coefficients for 3DNow! routine
void FIRFilter3DNow::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for 3DNow!
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((uint)filterCoeffsUnalign + 15) & (uint)-16);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// 3DNow!-optimized version of the filter routine for stereo sound
uint FIRFilter3DNow::evaluateFilterStereo(float *dest, const float *src, uint numSamples) const
{
float *filterCoeffsLocal = filterCoeffsAlign;
uint count = (numSamples - length) & (uint)-2;
uint lengthLocal = length / 4;
assert(length != 0);
assert(count % 2 == 0);
/* original code:
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
filterCoeffsLocal = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * filterCoeffsLocal[0] +
ptr[2] * filterCoeffsLocal[2] +
ptr[4] * filterCoeffsLocal[4] +
ptr[6] * filterCoeffsLocal[6];
sumr1 += ptr[1] * filterCoeffsLocal[1] +
ptr[3] * filterCoeffsLocal[3] +
ptr[5] * filterCoeffsLocal[5] +
ptr[7] * filterCoeffsLocal[7];
suml2 += ptr[8] * filterCoeffsLocal[0] +
ptr[10] * filterCoeffsLocal[2] +
ptr[12] * filterCoeffsLocal[4] +
ptr[14] * filterCoeffsLocal[6];
sumr2 += ptr[9] * filterCoeffsLocal[1] +
ptr[11] * filterCoeffsLocal[3] +
ptr[13] * filterCoeffsLocal[5] +
ptr[15] * filterCoeffsLocal[7];
ptr += 16;
filterCoeffsLocal += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
_asm
{
mov eax, dword ptr dest
mov ebx, dword ptr src
mov edx, count
shr edx, 1
loop1:
// "outer loop" : during each round 2*2 output samples are calculated
prefetch [ebx] // give a prefetch hint to CPU what data are to be needed soonish
prefetch [filterCoeffsLocal] // give a prefetch hint to CPU what data are to be needed soonish
mov esi, ebx
mov edi, filterCoeffsLocal
pxor mm0, mm0
pxor mm1, mm1
mov ecx, lengthLocal
loop2:
// "inner loop" : during each round four FIR filter taps are evaluated for 2*2 output samples
movq mm2, [edi]
movq mm3, mm2
prefetch [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm2, [esi]
prefetch [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm3, [esi + 8]
movq mm4, [edi + 8]
movq mm5, mm4
pfadd mm0, mm2
pfmul mm4, [esi + 8]
pfadd mm1, mm3
pfmul mm5, [esi + 16]
movq mm2, [edi + 16]
movq mm6, mm2
pfadd mm0, mm4
pfmul mm2, [esi + 16]
pfadd mm1, mm5
pfmul mm6, [esi + 24]
movq mm3, [edi + 24]
movq mm7, mm3
pfadd mm0, mm2
pfmul mm3, [esi + 24]
pfadd mm1, mm6
pfmul mm7, [esi + 32]
add esi, 32
pfadd mm0, mm3
add edi, 32
pfadd mm1, mm7
dec ecx
jnz loop2
movq [eax], mm0
add ebx, 16
movq [eax + 8], mm1
add eax, 16
dec edx
jnz loop1
femms
}
return count;
}
#endif // ALLOW_3DNOW

458
3rdparty/soundtouch/COPYING.TXT vendored Normal file
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@ -0,0 +1,458 @@
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NO WARRANTY
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WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
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END OF TERMS AND CONDITIONS

71
3rdparty/soundtouch/Makefile.am vendored
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@ -1,71 +0,0 @@
## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am 138 2012-04-01 20:00:09Z oparviai $
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch 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.
##
## SoundTouch 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., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
include $(top_srcdir)/config/am_include.mk
# set to something if you want other stuff to be included in the distribution tarball
EXTRA_DIST=SoundTouch.dsp SoundTouch.dsw SoundTouch.sln SoundTouch.vcproj
noinst_HEADERS=AAFilter.h cpu_detect.h cpu_detect_x86.cpp FIRFilter.h RateTransposer.h TDStretch.h PeakFinder.h
lib_LTLIBRARIES=libSoundTouch.la
#
libSoundTouch_la_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp RateTransposer.cpp SoundTouch.cpp TDStretch.cpp cpu_detect_x86.cpp BPMDetect.cpp PeakFinder.cpp
# Compiler flags
AM_CXXFLAGS=-O3 -fcheck-new -I../../include
# Compile the files that need MMX and SSE individually.
libSoundTouch_la_LIBADD=libSoundTouchMMX.la libSoundTouchSSE.la
noinst_LTLIBRARIES=libSoundTouchMMX.la libSoundTouchSSE.la
libSoundTouchMMX_la_SOURCES=mmx_optimized.cpp
libSoundTouchSSE_la_SOURCES=sse_optimized.cpp
# We enable optimizations by default.
# If MMX is supported compile with -mmmx.
# Do not assume -msse is also supported.
if HAVE_MMX
libSoundTouchMMX_la_CXXFLAGS = -mmmx $(AM_CXXFLAGS)
else
libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
endif
# We enable optimizations by default.
# If SSE is supported compile with -msse.
if HAVE_SSE
libSoundTouchSSE_la_CXXFLAGS = -msse $(AM_CXXFLAGS)
else
libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
endif
# Let the user disable optimizations if he wishes to.
if !X86_OPTIMIZATIONS
libSoundTouchMMX_la_CXXFLAGS = $(AM_CXXFLAGS)
libSoundTouchSSE_la_CXXFLAGS = $(AM_CXXFLAGS)
endif
# other linking flags to add
# noinst_LTLIBRARIES = libSoundTouchOpt.la
# libSoundTouch_la_LIBADD = libSoundTouchOpt.la
# libSoundTouchOpt_la_SOURCES = mmx_optimized.cpp sse_optimized.cpp
# libSoundTouchOpt_la_CXXFLAGS = -O3 -msse -fcheck-new -I../../include

186
3rdparty/soundtouch/README.html vendored
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@ -8,15 +8,13 @@
<meta name="author" content="Olli Parviainen">
<meta name="description"
content="Readme file for SoundTouch audio processing library">
<meta name="GENERATOR" content="Microsoft FrontPage 4.0">
<meta name="ProgId" content="FrontPage.Editor.Document">
<style> <!-- .normal { font-family: Arial }
--></style>
</head>
<body class="normal">
<hr>
<h1>SoundTouch audio processing library v1.7.1</h1>
<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2012 </p>
<h1>SoundTouch audio processing library v1.9</h1>
<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2015</p>
<hr>
<h2>1. Introduction </h2>
<p>SoundTouch is an open-source audio processing library that allows
@ -33,26 +31,22 @@ same time </li>
</ul>
<h3>1.1 Contact information </h3>
<p>Author email: oparviai 'at' iki.fi </p>
<p>SoundTouch WWW page: <a href="http://www.surina.net/soundtouch">http://www.surina.net/soundtouch</a></p>
<p>SoundTouch WWW page: <a href="http://soundtouch.surina.net">http://soundtouch.surina.net</a></p>
<hr>
<h2>2. Compiling SoundTouch</h2>
<p>Before compiling, notice that you can choose the sample data format
if it's desirable to use floating point sample data instead of 16bit
integers. See section "sample data format" for more information.</p>
<p>Before compiling, notice that you can choose the sample data format if it's
desirable to use floating point sample data instead of 16bit integers. See
section &quot;sample data format&quot; for more information.</p>
<p>Also notice that SoundTouch can use OpenMP instructions for parallel
computation to accelerate the runtime processing speed in multi-core systems,
however, these improvements need to be separately enabled before compiling. See
OpenMP notes in Chapter 3 below.</p>
<h3>2.1. Building in Microsoft Windows</h3>
<p>Project files for Microsoft Visual C++ 6.0 and Visual C++ .NET are
supplied with the source code package.<br>
<p>Project files for Microsoft Visual C++ are supplied with the source
code package. Go to Microsoft WWW page to download
<a href="http://www.visualstudio.com/en-US/products/visual-studio-express-vs">
Microsoft Visual Studio Express version for free</a>.
</p>
<p> Please notice that SoundTouch library uses processor-specific
optimizations for Pentium III and AMD processors. Visual Studio .NET
and later versions supports the required instructions by default, but
Visual Studio 6.0 requires a processor pack upgrade to be installed in
order to support these optimizations. The processor pack upgrade can be
downloaded from Microsoft site at this URL:</p>
<p><a href="http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx">http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx</a></p>
<p>If the above URL is unavailable or removed, go to <a
href="http://msdn.microsoft.com/"> http://msdn.microsoft.com</a> and
perform a search with keywords "processor pack". </p>
<p>To build the binaries with Visual C++ compiler, either run
"make-win.bat" script, or open the appropriate project files in source
code directories with Visual Studio. The final executable will appear
@ -61,6 +55,22 @@ instead of the make-win.bat script, directories bin and lib may need to
be created manually to the SoundTouch package root for the final
executables. The make-win.bat script creates these directories
automatically. </p>
<p><strong>OpenMP NOTE</strong>: If activating the OpenMP parallel computing in
the compilation, the target program will require additional vcomp dll library to
properly run. In Visual C++ 9.0 these libraries can be found in the following
folders.</p>
<ul>
<li>x86 32bit: C:\Program Files (x86)\Microsoft Visual Studio
9.0\VC\redist\x86\Microsoft.VC90.OPENMP\vcomp90.dll</li>
<li>x64 64bit: C:\Program Files (x86)\Microsoft Visual Studio
9.0\VC\redist\amd64\Microsoft.VC90.OPENMP\vcomp90.dll</li>
</ul>
<p>In Visual Studio 2008, a SP1 version may be required for these libraries. In
other VC++ versions the required library will be expectedly found in similar
&quot;redist&quot; location.</p>
<p>Notice that as minor demonstration of a &quot;dll hell&quot; phenomenon both the 32-bit
and 64-bit version of vcomp90.dll have the same filename but different contents,
thus choose the proper version to allow the program start.</p>
<h3>2.2. Building in Gnu platforms</h3>
<p>The SoundTouch library compiles in practically any platform
supporting GNU compiler (GCC) tools. SoundTouch requires GCC version 4.3 or later.</p>
@ -92,7 +102,9 @@ Notice that "configure" file is not available before running the
<pre>make -</pre>
</td>
<td>
<p>Builds the SoundTouch library &amp; SoundStretch utility.</p>
<p>Builds the SoundTouch library &amp; SoundStretch utility. You can
optionally add &quot;-j&quot; switch after &quot;make&quot; to speed up the compilation in
multi-core systems.</p>
</td>
</tr>
<tr valign="top">
@ -133,7 +145,7 @@ directly and remove the following definition:<blockquote>
<pre>#define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1</pre>
</blockquote>
<h4><b>2.2.3 Compiling Shared Library / DLL version</b></h4>
<h4><b>2.2.3 Compiling Shared Library / DLL version in Cygwin</b></h4>
<p>
The GNU compilation does not automatically create a shared-library version of
SoundTouch (.so or .dll). If such is desired, then you can create it as follows
@ -147,7 +159,15 @@ sstrip SoundTouch.dll</pre>
<h3>2.1. Building in Android</h3>
<p>Android compilation instructions are within the
source code package, see file &quot;<b>source/Android-lib/README-SoundTouch-Android.html</b>&quot;
in the package.</p>
in the source code package. </p>
<p>The Android compilation automatically builds separate .so library binaries
for ARM, X86 and MIPS processor architectures. For optimal device support,
include all these .so library binaries into the Android .apk application
package, so the target Android device can automatically choose the proper
library binary version to use.</p>
<p>The <strong>source/Android-lib</strong> folder includes also an Android
example application that processes WAV audio files using SoundTouch library in
Android devices.</p>
<hr>
<h2>3. About implementation &amp; Usage tips <h3>3.1. Supported sample data formats</h3>
@ -343,28 +363,55 @@ function with parameter&nbsp; id of SETTING_USE_QUICKSEEK and value
<p>setSetting(SETTING_USE_QUICKSEEK, 1);</p>
</blockquote>
<p><strong>CPU-specific optimizations:</strong></p>
<p>Intel x86 specific SIMD optimizations are implemented using compiler
intrinsics, providing about a 3x processing speedup for x86 compatible
processors vs. non-SIMD implementation:</p>
<ul>
<li> Intel MMX optimized routines are used with compatible CPUs when
16bit integer sample type is used. MMX optimizations are available both
in Win32 and Gnu/x86 platforms. Compatible processors are Intel
PentiumMMX and later; AMD K6-2, Athlon and later. </li>
<li> Intel SSE optimized routines are used with compatible CPUs when
floating point sample type is used. SSE optimizations are currently
implemented for Win32 platform only. Processors compatible with SSE
extension are Intel processors starting from Pentium-III, and AMD
processors starting from Athlon XP. </li>
<li> AMD 3DNow! optimized routines are used with compatible CPUs when
floating point sample type is used, but SSE extension isn't supported .
3DNow! optimizations are currently implemented for Win32 platform only.
These optimizations are used in AMD K6-2 and Athlon (classic) CPU's;
better performing SSE routines are used with AMD processor starting
from Athlon XP. </li>
<li> Intel MMX optimized routines are used with x86 CPUs when 16bit integer
sample type is used</li>
<li> Intel SSE optimized routines are used with x86 CPUs when 32bit floating
point sample type is used</li>
</ul>
<h3>3.5 OpenMP parallel computation</h3>
<p>SoundTouch 1.9 onwards support running the algorithms parallel in several CPU
cores. Based on benchmark the experienced multi-core processing speed-up gain
ranges between +30% (on a high-spec dual-core x86 Windows PC) to 215% (on a moderately low-spec
quad-core ARM of Raspberry Pi2).</p>
<p>The parallel computing support is implemented using OpenMP spec 3.0
instructions. These instructions are supported by Visual C++ 2008 and later, and
GCC v4.2 and later. Compilers that do not supporting OpenMP will ignore these
optimizations and routines will still work properly. Possible warnings about
unknown #pragmas are related to OpenMP support and can be safely ignored.</p>
<p>The OpenMP improvements are disabled by default, and need to be enabled by
developer during compile-time. Reason for this is that parallel processing adds
moderate runtime overhead in managing the multi-threading, so it may not be
necessary nor desirable in all applications. For example real-time processing
that is not constrained by CPU power will not benefit of speed-up provided by
the parallel processing, in the contrary it may increase power consumption due
to the increased overhead.</p>
<p>However, applications that run on low-spec multi-core CPUs and may otherwise
have possibly constrained performance will benefit of the OpenMP improvements.
This include for example multi-core embedded devices.</p>
<p>OpenMP parallel computation can be enabled before compiling SoundTouch
library as follows:</p>
<ul>
<li><strong>Visual Studio</strong>: Open properties for the <strong>SoundTouch
</strong>sub-project, browse to <strong>C/C++</strong> and <strong>Language
</strong>settings. Set
there &quot;<strong>OpenMP support</strong>&quot; to &quot;<strong>Yes</strong>&quot;. Alternatively add
<strong>/openmp</strong> switch to command-line
parameters</li>
<li><strong>GNU</strong>: Run the configure script with &quot;<strong>./configure
--enable-openmp</strong>&quot; switch, then run make as usually</li>
<li><strong>Android</strong>: Add &quot;<strong>-fopenmp</strong>&quot; switches to compiler &amp; linker
options, see README-SoundTouch-Android.html in the source code package for
more detailed instructions.</li>
</ul>
<hr>
<h2><a name="SoundStretch"></a>4. SoundStretch audio processing utility
</h2>
<p>SoundStretch audio processing utility<br>
Copyright (c) Olli Parviainen 2002-2012</p>
Copyright (c) Olli Parviainen 2002-2015</p>
<p>SoundStretch is a simple command-line application that can change
tempo, pitch and playback rates of WAV sound files. This program is
intended primarily to demonstrate how the "SoundTouch" library can be
@ -510,9 +557,42 @@ and estimates the BPM rate:</p>
<blockquote>
<pre>soundstretch stdin -bpm</pre>
</blockquote>
<p><strong>Example 6</strong></p>
<p>The following command tunes song from original 440Hz tuning to 432Hz tuning:
this corresponds to lowering the pitch by -0.318 semitones:</p>
<blockquote>
<pre>soundstretch original.wav output.wav -pitch=-0.318</pre>
</blockquote>
<hr>
<h2>5. Change History</h2>
<h3>5.1. SoundTouch library Change History </h3>
<p><b>1.9:</b></p>
<ul>
<li>Added support for parallel computation support via OpenMP primitives for better performance in multicore systems.
Benchmarks show that achieved parallel processing speedup improvement
typically range from +30% (x86 dual-core) to +180% (ARM quad-core). The
OpenMP optimizations are disabled by default, see OpenMP notes above in this
readme file how to enabled these optimizations.</li>
<li>Android: Added support for Android devices featuring X86 and MIPS CPUs,
in addition to ARM CPUs.</li>
<li>Android: More versatile Android example application that processes WAV
audio files with SoundTouch library</li>
<li>Replaced Windows-like 'BOOL' types with native 'bool'</li>
<li>Changed documentation token to "dist_doc_DATA" in Makefile.am file</li>
<li>Miscellaneous small fixes and improvements</li>
</ul>
<p><b>1.8.0:</b></p>
<ul>
<li>Added support for multi-channel audio processing</li>
<li>Added support for <b>cubic</b> and <b>shannon</b> interpolation for rate and pitch shift effects besides
the original <b>linear</b> interpolation, to reduce aliasing at high frequencies due to interpolation.
Cubic interpolation is used as default for floating point processing, and linear interpolation for integer
processing.</li>
<li>Fixed bug in anti-alias filtering that limited stop-band attenuation to -10 dB instead of <-50dB, and
increased filter length from 32 to 64 taps to further reduce aliasing due to frequency folding.</li>
<li>Performance improvements in cross-correlation algorithm</li>
<li>Other bug and compatibility fixes</li>
</ul>
<p><b>1.7.1:</b></p>
<ul>
<li>Added files for Android compilation
@ -660,6 +740,11 @@ accessing the FIFOSampleBuffer class from external files. </li>
</ul>
<p>&nbsp;</p>
<h3>5.2. SoundStretch application Change History </h3>
<p><b>1.9:</b></p>
<ul>
<li>Added support for WAV file 'fact' information chunk.</li>
</ul>
<p><b>1.7.0:</b></p>
<ul>
<li>Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer
@ -718,40 +803,37 @@ switch "-bpm" </li>
<hr>
<h2>6. Acknowledgements </h2>
<p>Kudos for these people who have contributed to development or
submitted bugfixes since SoundTouch v1.3.1: </p>
submitted bugfixes:</p>
<ul>
<li> Arthur A</li>
<li> Richard Ash</li>
<li> Stanislav Brabec</li>
<li> Christian Budde</li>
<li> Chris Bryan</li>
<li> Jacek Caban</li>
<li> Brian Cameron</li>
<li> Jason Champion</li>
<li> David Clark</li>
<li> Patrick Colis</li>
<li> Miquel Colon</li>
<li> Justin Frankel</li>
<li> Jason Garland</li>
<li> Takashi Iwai</li>
<li> Yuval Naveh</li>
<li> Paulo Pizarro</li>
<li> Blaise Potard</li>
<li> RJ Ryan</li>
<li> Patrick Colis </li>
<li> Miquel Colon </li>
<li> Jim Credland</li>
<li> Sandro Cumerlato</li>
<li> Justin Frankel</li>
<li> Masa H.</li>
<li> Jason Garland</li>
<li> Takashi Iwai</li>
<li> Thomas Klausner</li>
<li> Mathias Möhl</li>
<li> Yuval Naveh</li>
<li> Paulo Pizarro</li>
<li> Blaise Potard</li>
<li> Michael Pruett</li>
<li> Rajeev Puran</li>
<li> RJ Ryan</li>
<li> John Sheehy</li>
<li> Tim Shuttleworth</li>
<li> Albert Sirvent</li>
<li> John Stumpo</li>
<li> Tim Shuttleworth</li>
<li> Katja Vetter</li>
</ul>
<p>Moral greetings to all other contributors and users also!</p>
@ -770,8 +852,8 @@ General Public License for more details.</p>
License along with this library; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA</p>
<hr><!--
$Id: README.html 168 2012-12-28 20:55:19Z oparviai $ -->
$Id: README.html 220 2015-05-18 17:39:26Z oparviai $ -->
<p>
<i>RREADME.html file updated on 28-Dec-2012</i></p>
<i>README.html file updated in May-2015</i></p>
</body>
</html>

626
3rdparty/soundtouch/RateTransposer.cpp vendored
View File

@ -1,626 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application)
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include "RateTransposer.h"
#include "AAFilter.h"
using namespace soundtouch;
/// A linear samplerate transposer class that uses integer arithmetics.
/// for the transposing.
class RateTransposerInteger : public RateTransposer
{
protected:
int iSlopeCount;
int iRate;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposerInteger();
virtual ~RateTransposerInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
};
/// A linear samplerate transposer class that uses floating point arithmetics
/// for the transposing.
class RateTransposerFloat : public RateTransposer
{
protected:
float fSlopeCount;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposerFloat();
virtual ~RateTransposerFloat();
};
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX/SSE/etc-capable CPU available or not.
void * RateTransposer::operator new(size_t s)
{
ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
return newInstance();
}
RateTransposer *RateTransposer::newInstance()
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
return ::new RateTransposerInteger;
#else
return ::new RateTransposerFloat;
#endif
}
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
numChannels = 2;
bUseAAFilter = TRUE;
fRate = 0;
// Instantiates the anti-alias filter with default tap length
// of 32
pAAFilter = new AAFilter(32);
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(BOOL newMode)
{
bUseAAFilter = newMode;
}
/// Returns nonzero if anti-alias filter is enabled.
BOOL RateTransposer::isAAFilterEnabled() const
{
return bUseAAFilter;
}
AAFilter *RateTransposer::getAAFilter()
{
return pAAFilter;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposer::setRate(float newRate)
{
double fCutoff;
fRate = newRate;
// design a new anti-alias filter
if (newRate > 1.0f)
{
fCutoff = 0.5f / newRate;
}
else
{
fCutoff = 0.5f * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
// Outputs as many samples of the 'outputBuffer' as possible, and if there's
// any room left, outputs also as many of the incoming samples as possible.
// The goal is to drive the outputBuffer empty.
//
// It's allowed for 'output' and 'input' parameters to point to the same
// memory position.
/*
void RateTransposer::flushStoreBuffer()
{
if (storeBuffer.isEmpty()) return;
outputBuffer.moveSamples(storeBuffer);
}
*/
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
processSamples(samples, nSamples);
}
// Transposes up the sample rate, causing the observed playback 'rate' of the
// sound to decrease
void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
{
uint count, sizeTemp, num;
// If the parameter 'uRate' value is smaller than 'SCALE', first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// First check that there's enough room in 'storeBuffer'
// (+16 is to reserve some slack in the destination buffer)
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
// Transpose the samples, store the result into the end of "storeBuffer"
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
storeBuffer.putSamples(count);
// Apply the anti-alias filter to samples in "store output", output the
// result to "dest"
num = storeBuffer.numSamples();
count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
storeBuffer.ptrBegin(), num, (uint)numChannels);
outputBuffer.putSamples(count);
// Remove the processed samples from "storeBuffer"
storeBuffer.receiveSamples(count);
}
// Transposes down the sample rate, causing the observed playback 'rate' of the
// sound to increase
void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
{
uint count, sizeTemp;
// If the parameter 'uRate' value is larger than 'SCALE', first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose.
// Add the new samples to the end of the storeBuffer
storeBuffer.putSamples(src, nSamples);
// Anti-alias filter the samples to prevent folding and output the filtered
// data to tempBuffer. Note : because of the FIR filter length, the
// filtering routine takes in 'filter_length' more samples than it outputs.
assert(tempBuffer.isEmpty());
sizeTemp = storeBuffer.numSamples();
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
if (count == 0) return;
// Remove the filtered samples from 'storeBuffer'
storeBuffer.receiveSamples(count);
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
outputBuffer.putSamples(count);
}
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{
uint count;
uint sizeReq;
if (nSamples == 0) return;
assert(pAAFilter);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == FALSE)
{
sizeReq = (uint)((float)nSamples / fRate + 1.0f);
count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
outputBuffer.putSamples(count);
return;
}
// Transpose with anti-alias filter
if (fRate < 1.0f)
{
upsample(src, nSamples);
}
else
{
downsample(src, nSamples);
}
}
// Transposes the sample rate of the given samples using linear interpolation.
// Returns the number of samples returned in the "dest" buffer
inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
if (numChannels == 2)
{
return transposeStereo(dest, src, nSamples);
}
else
{
return transposeMono(dest, src, nSamples);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels)
{
assert(nChannels > 0);
if (numChannels == nChannels) return;
assert(nChannels == 1 || nChannels == 2);
numChannels = nChannels;
storeBuffer.setChannels(numChannels);
tempBuffer.setChannels(numChannels);
outputBuffer.setChannels(numChannels);
// Inits the linear interpolation registers
resetRegisters();
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
storeBuffer.clear();
}
// Returns nonzero if there aren't any samples available for outputting.
int RateTransposer::isEmpty() const
{
int res;
res = FIFOProcessor::isEmpty();
if (res == 0) return 0;
return storeBuffer.isEmpty();
}
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerInteger - integer arithmetic implementation
//
/// fixed-point interpolation routine precision
#define SCALE 65536
// Constructor
RateTransposerInteger::RateTransposerInteger() : RateTransposer()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
RateTransposerInteger::resetRegisters();
RateTransposerInteger::setRate(1.0f);
}
RateTransposerInteger::~RateTransposerInteger()
{
}
void RateTransposerInteger::resetRegisters()
{
iSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int i, used;
LONG_SAMPLETYPE temp, vol1;
if (nSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= nSamples - 1) goto end;
}
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[used] * vol1 + iSlopeCount * src[used + 1];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[nSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Stereo' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int srcPos, i, used;
LONG_SAMPLETYPE temp, vol1;
if (nSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= nSamples - 1) goto end;
}
srcPos = 2 * used;
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += iRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * nSamples - 2];
sPrevSampleR = src[2 * nSamples - 1];
return i;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposerInteger::setRate(float newRate)
{
iRate = (int)(newRate * SCALE + 0.5f);
RateTransposer::setRate(newRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerFloat - floating point arithmetic implementation
//
//////////////////////////////////////////////////////////////////////////////
// Constructor
RateTransposerFloat::RateTransposerFloat() : RateTransposer()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
RateTransposerFloat::resetRegisters();
RateTransposerFloat::setRate(1.0f);
}
RateTransposerFloat::~RateTransposerFloat()
{
}
void RateTransposerFloat::resetRegisters()
{
fSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int i, used;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (fSlopeCount <= 1.0f)
{
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
i++;
fSlopeCount += fRate;
}
fSlopeCount -= 1.0f;
if (nSamples > 1)
{
while (1)
{
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
}
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
i++;
fSlopeCount += fRate;
}
}
end:
// Store the last sample for the next round
sPrevSampleL = src[nSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
{
unsigned int srcPos, i, used;
if (nSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (fSlopeCount <= 1.0f)
{
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
i++;
fSlopeCount += fRate;
}
// now always (iSlopeCount > 1.0f)
fSlopeCount -= 1.0f;
if (nSamples > 1)
{
while (1)
{
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
}
srcPos = 2 * used;
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos]
+ fSlopeCount * src[srcPos + 2]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1]
+ fSlopeCount * src[srcPos + 3]);
i++;
fSlopeCount += fRate;
}
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * nSamples - 2];
sPrevSampleR = src[2 * nSamples - 1];
return i;
}

132
3rdparty/soundtouch/SoundTouch.vcxproj vendored
View File

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68
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745
3rdparty/soundtouch/WavFile.cpp vendored
View File

@ -1,745 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define _BIG_ENDIAN_ during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain,
/// but the reason for 'yet another' one is that those generic WAV reader
/// libraries are exhaustingly large and cumbersome! Wanted to have something
/// simpler here, i.e. something that's not already larger than rest of the
/// SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.cpp 63 2009-02-21 16:00:14Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdio.h>
#include <stdexcept>
#include <string>
#include <cstring>
#include <assert.h>
#include <limits.h>
#include "WavFile.h"
using namespace std;
static const char riffStr[] = "RIFF";
static const char waveStr[] = "WAVE";
static const char fmtStr[] = "fmt ";
static const char dataStr[] = "data";
//////////////////////////////////////////////////////////////////////////////
//
// Helper functions for swapping byte order to correctly read/write WAV files
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
// turn-on the conversion if it appears necessary.
//
// For example, Intel x86 is little-endian and doesn't require conversion,
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
#ifdef BYTE_ORDER
// In gcc compiler detect the byte order automatically
#if BYTE_ORDER == BIG_ENDIAN
// big-endian platform.
#define _BIG_ENDIAN_
#endif
#endif
#ifdef _BIG_ENDIAN_
// big-endian CPU, swap bytes in 16 & 32 bit words
// helper-function to swap byte-order of 32bit integer
static inline void _swap32(unsigned int &dwData)
{
dwData = ((dwData >> 24) & 0x000000FF) |
((dwData >> 8) & 0x0000FF00) |
((dwData << 8) & 0x00FF0000) |
((dwData << 24) & 0xFF000000);
}
// helper-function to swap byte-order of 16bit integer
static inline void _swap16(unsigned short &wData)
{
wData = ((wData >> 8) & 0x00FF) |
((wData << 8) & 0xFF00);
}
// helper-function to swap byte-order of buffer of 16bit integers
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumWords)
{
unsigned long i;
for (i = 0; i < dwNumWords; i ++)
{
_swap16(pData[i]);
}
}
#else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such
// dummy helper-function
static inline void _swap32(unsigned int &dwData)
{
// do nothing
}
// dummy helper-function
static inline void _swap16(unsigned short &wData)
{
// do nothing
}
// dummy helper-function
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumBytes)
{
// do nothing
}
#endif // BIG_ENDIAN
//////////////////////////////////////////////////////////////////////////////
//
// Class WavInFile
//
WavInFile::WavInFile(const char *fileName)
{
// Try to open the file for reading
fptr = fopen(fileName, "rb");
if (fptr == NULL)
{
// didn't succeed
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for reading.";
throw runtime_error(msg);
}
init();
}
WavInFile::WavInFile(FILE *file)
{
// Try to open the file for reading
fptr = file;
if (!file)
{
// didn't succeed
string msg = "Error : Unable to access input stream for reading";
throw runtime_error(msg);
}
init();
}
/// Init the WAV file stream
void WavInFile::init()
{
int hdrsOk;
// assume file stream is already open
assert(fptr);
// Read the file headers
hdrsOk = readWavHeaders();
if (hdrsOk != 0)
{
// Something didn't match in the wav file headers
string msg = "Input file is corrupt or not a WAV file";
throw runtime_error(msg);
}
if (header.format.fixed != 1)
{
string msg = "Input file uses unsupported encoding.";
throw runtime_error(msg);
}
dataRead = 0;
}
WavInFile::~WavInFile()
{
if (fptr) fclose(fptr);
fptr = NULL;
}
void WavInFile::rewind()
{
int hdrsOk;
fseek(fptr, 0, SEEK_SET);
hdrsOk = readWavHeaders();
assert(hdrsOk == 0);
dataRead = 0;
}
int WavInFile::checkCharTags() const
{
// header.format.fmt should equal to 'fmt '
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
// header.data.data_field should equal to 'data'
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
return 0;
}
int WavInFile::read(char *buffer, int maxElems)
{
int numBytes;
uint afterDataRead;
// ensure it's 8 bit format
if (header.format.bits_per_sample != 8)
{
throw runtime_error("Error: WavInFile::read(char*, int) works only with 8bit samples.");
}
assert(sizeof(char) == 1);
numBytes = maxElems;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
assert(buffer);
numBytes = fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
return numBytes;
}
int WavInFile::read(short *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
assert(buffer);
if (header.format.bits_per_sample == 8)
{
// 8 bit format
char *temp = new char[maxElems];
int i;
numElems = read(temp, maxElems);
// convert from 8 to 16 bit
for (i = 0; i < numElems; i ++)
{
buffer[i] = temp[i] << 8;
}
delete[] temp;
}
else
{
// 16 bit format
assert(header.format.bits_per_sample == 16);
assert(sizeof(short) == 2);
numBytes = maxElems * 2;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
numBytes = fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / 2;
// 16bit samples, swap byte order if necessary
_swap16Buffer((unsigned short *)buffer, numElems);
}
return numElems;
}
int WavInFile::read(float *buffer, int maxElems)
{
short *temp = new short[maxElems];
int num;
int i;
double fscale;
num = read(temp, maxElems);
fscale = 1.0 / 32768.0;
// convert to floats, scale to range [-1..+1[
for (i = 0; i < num; i ++)
{
buffer[i] = (float)(fscale * (double)temp[i]);
}
delete[] temp;
return num;
}
int WavInFile::eof() const
{
// return true if all data has been read or file eof has reached
return (dataRead == header.data.data_len || feof(fptr));
}
// test if character code is between a white space ' ' and little 'z'
static int isAlpha(char c)
{
return (c >= ' ' && c <= 'z') ? 1 : 0;
}
// test if all characters are between a white space ' ' and little 'z'
static int isAlphaStr(const char *str)
{
char c;
c = str[0];
while (c)
{
if (isAlpha(c) == 0) return 0;
str ++;
c = str[0];
}
return 1;
}
int WavInFile::readRIFFBlock()
{
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
// swap 32bit data byte order if necessary
_swap32((unsigned int &)header.riff.package_len);
// header.riff.riff_char should equal to 'RIFF');
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
// header.riff.wave should equal to 'WAVE'
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
return 0;
}
int WavInFile::readHeaderBlock()
{
char label[5];
string sLabel;
// lead label string
if (fread(label, 1, 4, fptr) !=4) return -1;
label[4] = 0;
if (isAlphaStr(label) == 0) return -1; // not a valid label
// Decode blocks according to their label
if (strcmp(label, fmtStr) == 0)
{
int nLen, nDump;
// 'fmt ' block
memcpy(header.format.fmt, fmtStr, 4);
// read length of the format field
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((unsigned int &)nLen); // int format_len;
header.format.format_len = nLen;
// calculate how much length differs from expected
nDump = nLen - ((int)sizeof(header.format) - 8);
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.format) - 8;
}
// read data
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap16((unsigned short &)header.format.fixed); // short int fixed;
_swap16((unsigned short &)header.format.channel_number); // short int channel_number;
_swap32((unsigned int &)header.format.sample_rate); // int sample_rate;
_swap32((unsigned int &)header.format.byte_rate); // int byte_rate;
_swap16((unsigned short &)header.format.byte_per_sample); // short int byte_per_sample;
_swap16((unsigned short &)header.format.bits_per_sample); // short int bits_per_sample;
// if format_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, dataStr) == 0)
{
// 'data' block
memcpy(header.data.data_field, dataStr, 4);
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((unsigned int &)header.data.data_len);
return 1;
}
else
{
uint len, i;
uint temp;
// unknown block
// read length
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
// scan through the block
for (i = 0; i < len; i ++)
{
if (fread(&temp, 1, 1, fptr) != 1) return -1;
if (feof(fptr)) return -1; // unexpected eof
}
}
return 0;
}
int WavInFile::readWavHeaders()
{
int res;
memset(&header, 0, sizeof(header));
res = readRIFFBlock();
if (res) return 1;
// read header blocks until data block is found
do
{
// read header blocks
res = readHeaderBlock();
if (res < 0) return 1; // error in file structure
} while (res == 0);
// check that all required tags are legal
return checkCharTags();
}
uint WavInFile::getNumChannels() const
{
return header.format.channel_number;
}
uint WavInFile::getNumBits() const
{
return header.format.bits_per_sample;
}
uint WavInFile::getBytesPerSample() const
{
return getNumChannels() * getNumBits() / 8;
}
uint WavInFile::getSampleRate() const
{
return header.format.sample_rate;
}
uint WavInFile::getDataSizeInBytes() const
{
return header.data.data_len;
}
uint WavInFile::getNumSamples() const
{
if (header.format.byte_per_sample == 0) return 0;
return header.data.data_len / (unsigned short)header.format.byte_per_sample;
}
uint WavInFile::getLengthMS() const
{
uint numSamples;
uint sampleRate;
numSamples = getNumSamples();
sampleRate = getSampleRate();
assert(numSamples < UINT_MAX / 1000);
return (1000 * numSamples / sampleRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavOutFile
//
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = fopen(fileName, "wb");
if (fptr == NULL)
{
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for writing.";
//pmsg = msg.c_str;
throw runtime_error(msg);
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = file;
if (fptr == NULL)
{
string msg = "Error : Unable to access output file stream.";
throw runtime_error(msg);
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::~WavOutFile()
{
finishHeader();
if (fptr) fclose(fptr);
fptr = NULL;
}
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
{
// fill in the 'riff' part..
// copy string 'RIFF' to riff_char
memcpy(&(header.riff.riff_char), riffStr, 4);
// package_len unknown so far
header.riff.package_len = 0;
// copy string 'WAVE' to wave
memcpy(&(header.riff.wave), waveStr, 4);
// fill in the 'format' part..
// copy string 'fmt ' to fmt
memcpy(&(header.format.fmt), fmtStr, 4);
header.format.format_len = 0x10;
header.format.fixed = 1;
header.format.channel_number = (short)channels;
header.format.sample_rate = (int)sampleRate;
header.format.bits_per_sample = (short)bits;
header.format.byte_per_sample = (short)(bits * channels / 8);
header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
header.format.sample_rate = (int)sampleRate;
// fill in the 'data' part..
// copy string 'data' to data_field
memcpy(&(header.data.data_field), dataStr, 4);
// data_len unknown so far
header.data.data_len = 0;
}
void WavOutFile::finishHeader()
{
// supplement the file length into the header structure
header.riff.package_len = bytesWritten + 36;
header.data.data_len = bytesWritten;
writeHeader();
}
void WavOutFile::writeHeader()
{
WavHeader hdrTemp;
int res;
// swap byte order if necessary
hdrTemp = header;
_swap32((unsigned int &)hdrTemp.riff.package_len);
_swap32((unsigned int &)hdrTemp.format.format_len);
_swap16((unsigned short &)hdrTemp.format.fixed);
_swap16((unsigned short &)hdrTemp.format.channel_number);
_swap32((unsigned int &)hdrTemp.format.sample_rate);
_swap32((unsigned int &)hdrTemp.format.byte_rate);
_swap16((unsigned short &)hdrTemp.format.byte_per_sample);
_swap16((unsigned short &)hdrTemp.format.bits_per_sample);
_swap32((unsigned int &)hdrTemp.data.data_len);
// write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET);
res = fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
if (res != 1)
{
throw runtime_error("Error while writing to a wav file.");
}
// jump back to the end of the file
fseek(fptr, 0, SEEK_END);
}
void WavOutFile::write(const char *buffer, int numElems)
{
int res;
if (header.format.bits_per_sample != 8)
{
throw runtime_error("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
}
assert(sizeof(char) == 1);
res = fwrite(buffer, 1, numElems, fptr);
if (res != numElems)
{
throw runtime_error("Error while writing to a wav file.");
}
bytesWritten += numElems;
}
void WavOutFile::write(const short *buffer, int numElems)
{
int res;
// 16 bit samples
if (numElems < 1) return; // nothing to do
if (header.format.bits_per_sample == 8)
{
int i;
char *temp = new char[numElems];
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i ++)
{
temp[i] = buffer[i] >> 8;
}
// write in 8bit format
write(temp, numElems);
delete[] temp;
}
else
{
// 16bit format
unsigned short *pTemp = new unsigned short[numElems];
assert(header.format.bits_per_sample == 16);
// allocate temp buffer to swap byte order if necessary
memcpy(pTemp, buffer, numElems * 2);
_swap16Buffer(pTemp, numElems);
res = fwrite(pTemp, 2, numElems, fptr);
delete[] pTemp;
if (res != numElems)
{
throw runtime_error("Error while writing to a wav file.");
}
bytesWritten += 2 * numElems;
}
}
void WavOutFile::write(const float *buffer, int numElems)
{
int i;
short *temp = new short[numElems];
int iTemp;
// convert to 16 bit integer
for (i = 0; i < numElems; i ++)
{
// convert to integer
iTemp = (int)(32768.0f * buffer[i]);
// saturate
if (iTemp < -32768) iTemp = -32768;
if (iTemp > 32767) iTemp = 32767;
temp[i] = (short)iTemp;
}
write(temp, numElems);
delete[] temp;
}

28
3rdparty/soundtouch/build.sh vendored
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@ -1,28 +0,0 @@
#!/bin/sh
curdir=`pwd`
echo -----------------
echo Building SoundTouch
echo -----------------
if [ $# -gt 0 ] && [ $1 = "all" ]
then
aclocal
automake -a
autoconf
./configure
make clean
make install
else
make $@
fi
if [ $? -ne 0 ]
then
exit 1
fi
#cp libZeroSPU2*.so* ${PCSX2PLUGINS}

37
3rdparty/soundtouch/configure.ac vendored
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@ -1,37 +0,0 @@
# -*- Autoconf -*-
# Process this file with autoconf to produce a configure script.
#AC_PREREQ([2.63])
AC_INIT([FULL-PACKAGE-NAME], [VERSION], [BUG-REPORT-ADDRESS])
AM_INIT_AUTOMAKE
AC_CONFIG_SRCDIR([BPMDetect.h])
# Checks for programs.
AC_PROG_CXX
AC_PROG_CC
AC_PROG_RANLIB
CFLAGS=
CPPFLAGS=
CXXFLAGS=
CCASFLAGS=
CFLAGS+=" -m32 "
CPPFLAGS+=" -m32 "
CXXFLAGS+=" -m32 "
CCASFLAGS+=" -m32 "
# Checks for header files.
AC_CHECK_HEADERS([limits.h memory.h stdlib.h string.h])
# Checks for typedefs, structures, and compiler characteristics.
AC_C_INLINE
AC_C_RESTRICT
AC_TYPE_SIZE_T
AC_HEADER_STDBOOL
# Checks for library functions.
AC_CHECK_FUNCS([memmove memset])
AC_CONFIG_FILES([Makefile])
AC_OUTPUT

134
3rdparty/soundtouch/cpu_detect_x86_gcc.cpp vendored
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@ -1,134 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Generic version of the x86 CPU extension detection routine.
///
/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
/// for the Microsoft compiler version.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86_gcc.cpp 131 2011-09-02 18:56:11Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
#if (!(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS) || !(__GNUC__))
return 0; // always disable extensions on non-x86 platforms.
#else
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
asm volatile(
#ifndef __x86_64__
// Check if 'cpuid' instructions is available by toggling eflags bit 21.
// Skip this for x86-64 as they always have cpuid while stack manipulation
// differs from 16/32bit ISA.
"\n\txor %%esi, %%esi" // clear %%esi = result register
"\n\tpushf" // save eflags to stack
"\n\tmovl (%%esp), %%eax" // load eax from stack (with eflags)
"\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
"\n\txor $0x00200000, %%eax" // toggle bit 21
"\n\tmovl %%eax, (%%esp)" // store toggled eflags to stack
"\n\tpopf" // load eflags from stack
"\n\tpushf" // save updated eflags to stack
"\n\tmovl (%%esp), %%eax" // load eax from stack
"\n\tpopf" // pop stack to restore esp
"\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
"\n\tcmp %%ecx, %%eax" // compare to original eflags values
"\n\tjz end" // jumps to 'end' if cpuid not present
#endif // __x86_64__
// cpuid instruction available, test for presence of mmx instructions
"\n\tmovl $1, %%eax"
"\n\tcpuid"
"\n\ttest $0x00800000, %%edx"
"\n\tjz end" // branch if MMX not available
"\n\tor $0x01, %%esi" // otherwise add MMX support bit
"\n\ttest $0x02000000, %%edx"
"\n\tjz test3DNow" // branch if SSE not available
"\n\tor $0x08, %%esi" // otherwise add SSE support bit
"\n\ttest3DNow:"
// test for precense of AMD extensions
"\n\tmov $0x80000000, %%eax"
"\n\tcpuid"
"\n\tcmp $0x80000000, %%eax"
"\n\tjbe end" // branch if no AMD extensions detected
// test for precense of 3DNow! extension
"\n\tmov $0x80000001, %%eax"
"\n\tcpuid"
"\n\ttest $0x80000000, %%edx"
"\n\tjz end" // branch if 3DNow! not detected
"\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
"\n\tend:"
"\n\tmov %%esi, %0"
: "=r" (res)
: /* no inputs */
: "%edx", "%eax", "%ecx", "%esi" );
return res & ~_dwDisabledISA;
#endif
}

137
3rdparty/soundtouch/cpu_detect_x86_win.cpp vendored
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@ -1,137 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Win32 version of the x86 CPU detect routine.
///
/// This file is to be compiled in Windows platform with Microsoft Visual C++
/// Compiler. Please see 'cpu_detect_x86_gcc.cpp' for the gcc compiler version
/// for all GNU platforms.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-07-17 07:58:40 -0300 (dom, 17 jul 2011) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86_win.cpp 127 2011-07-17 10:58:40Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
#ifndef _M_X64
// 32bit compilation, detect CPU capabilities with inline assembler.
__asm
{
; check if 'cpuid' instructions is available by toggling eflags bit 21
;
xor esi, esi ; clear esi = result register
pushfd ; save eflags to stack
mov eax,dword ptr [esp] ; load eax from stack (with eflags)
mov ecx, eax ; save the original eflags values to ecx
xor eax, 0x00200000 ; toggle bit 21
mov dword ptr [esp],eax ; store toggled eflags to stack
popfd ; load eflags from stack
pushfd ; save updated eflags to stack
mov eax,dword ptr [esp] ; load eax from stack
popfd ; pop stack to restore stack pointer
xor edx, edx ; clear edx for defaulting no mmx
cmp eax, ecx ; compare to original eflags values
jz end ; jumps to 'end' if cpuid not present
; cpuid instruction available, test for presence of mmx instructions
mov eax, 1
cpuid
test edx, 0x00800000
jz end ; branch if MMX not available
or esi, SUPPORT_MMX ; otherwise add MMX support bit
test edx, 0x02000000
jz test3DNow ; branch if SSE not available
or esi, SUPPORT_SSE ; otherwise add SSE support bit
test3DNow:
; test for precense of AMD extensions
mov eax, 0x80000000
cpuid
cmp eax, 0x80000000
jbe end ; branch if no AMD extensions detected
; test for precense of 3DNow! extension
mov eax, 0x80000001
cpuid
test edx, 0x80000000
jz end ; branch if 3DNow! not detected
or esi, SUPPORT_3DNOW ; otherwise add 3DNow support bit
end:
mov res, esi
}
#else
// Visual C++ 64bit compilation doesn't support inline assembler. However,
// all x64 compatible CPUs support MMX & SSE extensions.
res = SUPPORT_MMX | SUPPORT_SSE | SUPPORT_SSE2;
#endif
return res & ~_dwDisabledISA;
}

1
3rdparty/soundtouch/depcomp vendored
View File

@ -1 +0,0 @@
/usr/share/automake-1.10/depcomp

1
3rdparty/soundtouch/install-sh vendored
View File

@ -1 +0,0 @@
/usr/share/automake-1.10/install-sh

1
3rdparty/soundtouch/missing vendored
View File

@ -1 +0,0 @@
/usr/share/automake-1.10/missing

2
3rdparty/soundtouch/BPMDetect.h → 3rdparty/soundtouch/soundtouch/BPMDetect.h vendored
View File

@ -26,7 +26,7 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-08-30 16:53:44 -0300 (qui, 30 ago 2012) $
// Last changed : $Date: 2012-08-30 19:53:44 +0000 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $

10
3rdparty/soundtouch/FIFOSampleBuffer.h → 3rdparty/soundtouch/soundtouch/FIFOSampleBuffer.h vendored
View File

@ -15,10 +15,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 16:29:53 -0300 (qua, 13 jun 2012) $
// Last changed : $Date: 2014-01-05 21:40:22 +0000 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h 143 2012-06-13 19:29:53Z oparviai $
// $Id: FIFOSampleBuffer.h 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -162,6 +162,12 @@ public:
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(int numChannels);
/// Get number of channels
int getChannels()
{
return channels;
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;

2
3rdparty/soundtouch/FIFOSamplePipe.h → 3rdparty/soundtouch/soundtouch/FIFOSamplePipe.h vendored
View File

@ -17,7 +17,7 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 16:29:53 -0300 (qua, 13 jun 2012) $
// Last changed : $Date: 2012-06-13 19:29:53 +0000 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $

24
3rdparty/soundtouch/STTypes.h → 3rdparty/soundtouch/soundtouch/STTypes.h vendored
View File

@ -8,10 +8,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-12-28 12:53:56 -0200 (sex, 28 dez 2012) $
// Last changed : $Date: 2015-05-18 15:25:07 +0000 (Mon, 18 May 2015) $
// File revision : $Revision: 3 $
//
// $Id: STTypes.h 162 2012-12-28 14:53:56Z oparviai $
// $Id: STTypes.h 215 2015-05-18 15:25:07Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -60,16 +60,6 @@ typedef unsigned long ulong;
#include "soundtouch_config.h"
#endif
#ifndef _WINDEF_
// if these aren't defined already by Windows headers, define now
typedef int BOOL;
#define FALSE 0
#define TRUE 1
#endif // _WINDEF_
namespace soundtouch
{
@ -78,7 +68,14 @@ namespace soundtouch
//#undef SOUNDTOUCH_INTEGER_SAMPLES
//#undef SOUNDTOUCH_FLOAT_SAMPLES
#if (defined(__SOFTFP__))
/// If following flag is defined, always uses multichannel processing
/// routines also for mono and stero sound. This is for routine testing
/// purposes; output should be same with either routines, yet disabling
/// the dedicated mono/stereo processing routines will result in slower
/// runtime performance so recommendation is to keep this off.
// #define USE_MULTICH_ALWAYS
#if (defined(__SOFTFP__) && defined(ANDROID))
// For Android compilation: Force use of Integer samples in case that
// compilation uses soft-floating point emulation - soft-fp is way too slow
#undef SOUNDTOUCH_FLOAT_SAMPLES
@ -175,6 +172,7 @@ namespace soundtouch
#else
// use c++ standard exceptions
#include <stdexcept>
#include <string>
#define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
#endif

14
3rdparty/soundtouch/SoundTouch.h → 3rdparty/soundtouch/soundtouch/SoundTouch.h vendored
View File

@ -41,10 +41,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-12-28 17:32:59 -0200 (sex, 28 dez 2012) $
// Last changed : $Date: 2015-05-18 15:28:41 +0000 (Mon, 18 May 2015) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.h 163 2012-12-28 19:32:59Z oparviai $
// $Id: SoundTouch.h 216 2015-05-18 15:28:41Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -79,10 +79,10 @@ namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.7.1"
#define SOUNDTOUCH_VERSION "1.9.0"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (10701)
#define SOUNDTOUCH_VERSION_ID (10900)
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
@ -160,7 +160,7 @@ private:
float virtualPitch;
/// Flag: Has sample rate been set?
BOOL bSrateSet;
bool bSrateSet;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
@ -247,8 +247,8 @@ public:
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'TRUE' if the setting was succesfully changed
BOOL setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
/// \return 'true' if the setting was succesfully changed
bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);

7
3rdparty/soundtouch/soundtouch_config.h vendored
View File

@ -1,7 +0,0 @@
#ifndef SOUNDTOUCH_CONFIG_H_INCLUDED
#define SOUNDTOUCH_CONFIG_H_INCLUDED
#endif // SOUNDTOUCH_CONFIG_H_INCLUDED

997
3rdparty/soundtouch/source/SoundStretch/WavFile.cpp vendored Normal file
View File

@ -0,0 +1,997 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define _BIG_ENDIAN_ during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain,
/// but the reason for 'yet another' one is that those generic WAV reader
/// libraries are exhaustingly large and cumbersome! Wanted to have something
/// simpler here, i.e. something that's not already larger than rest of the
/// SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-10-05 16:20:24 +0000 (Sun, 05 Oct 2014) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.cpp 200 2014-10-05 16:20:24Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdio.h>
#include <string>
#include <sstream>
#include <cstring>
#include <assert.h>
#include <limits.h>
#include "WavFile.h"
#include "STTypes.h"
using namespace std;
static const char riffStr[] = "RIFF";
static const char waveStr[] = "WAVE";
static const char fmtStr[] = "fmt ";
static const char factStr[] = "fact";
static const char dataStr[] = "data";
//////////////////////////////////////////////////////////////////////////////
//
// Helper functions for swapping byte order to correctly read/write WAV files
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
// turn-on the conversion if it appears necessary.
//
// For example, Intel x86 is little-endian and doesn't require conversion,
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
#ifdef BYTE_ORDER
// In gcc compiler detect the byte order automatically
#if BYTE_ORDER == BIG_ENDIAN
// big-endian platform.
#define _BIG_ENDIAN_
#endif
#endif
#ifdef _BIG_ENDIAN_
// big-endian CPU, swap bytes in 16 & 32 bit words
// helper-function to swap byte-order of 32bit integer
static inline int _swap32(int &dwData)
{
dwData = ((dwData >> 24) & 0x000000FF) |
((dwData >> 8) & 0x0000FF00) |
((dwData << 8) & 0x00FF0000) |
((dwData << 24) & 0xFF000000);
return dwData;
}
// helper-function to swap byte-order of 16bit integer
static inline short _swap16(short &wData)
{
wData = ((wData >> 8) & 0x00FF) |
((wData << 8) & 0xFF00);
return wData;
}
// helper-function to swap byte-order of buffer of 16bit integers
static inline void _swap16Buffer(short *pData, int numWords)
{
int i;
for (i = 0; i < numWords; i ++)
{
pData[i] = _swap16(pData[i]);
}
}
#else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such
// dummy helper-function
static inline int _swap32(int &dwData)
{
// do nothing
return dwData;
}
// dummy helper-function
static inline short _swap16(short &wData)
{
// do nothing
return wData;
}
// dummy helper-function
static inline void _swap16Buffer(short *pData, int numBytes)
{
// do nothing
}
#endif // BIG_ENDIAN
//////////////////////////////////////////////////////////////////////////////
//
// Class WavFileBase
//
WavFileBase::WavFileBase()
{
convBuff = NULL;
convBuffSize = 0;
}
WavFileBase::~WavFileBase()
{
delete[] convBuff;
convBuffSize = 0;
}
/// Get pointer to conversion buffer of at min. given size
void *WavFileBase::getConvBuffer(int sizeBytes)
{
if (convBuffSize < sizeBytes)
{
delete[] convBuff;
convBuffSize = (sizeBytes + 15) & -8; // round up to following 8-byte bounday
convBuff = new char[convBuffSize];
}
return convBuff;
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavInFile
//
WavInFile::WavInFile(const char *fileName)
{
// Try to open the file for reading
fptr = fopen(fileName, "rb");
if (fptr == NULL)
{
// didn't succeed
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for reading.";
ST_THROW_RT_ERROR(msg.c_str());
}
init();
}
WavInFile::WavInFile(FILE *file)
{
// Try to open the file for reading
fptr = file;
if (!file)
{
// didn't succeed
string msg = "Error : Unable to access input stream for reading";
ST_THROW_RT_ERROR(msg.c_str());
}
init();
}
/// Init the WAV file stream
void WavInFile::init()
{
int hdrsOk;
// assume file stream is already open
assert(fptr);
// Read the file headers
hdrsOk = readWavHeaders();
if (hdrsOk != 0)
{
// Something didn't match in the wav file headers
string msg = "Input file is corrupt or not a WAV file";
ST_THROW_RT_ERROR(msg.c_str());
}
/* Ignore 'fixed' field value as 32bit signed linear data can have other value than 1.
if (header.format.fixed != 1)
{
string msg = "Input file uses unsupported encoding.";
ST_THROW_RT_ERROR(msg.c_str());
}
*/
dataRead = 0;
}
WavInFile::~WavInFile()
{
if (fptr) fclose(fptr);
fptr = NULL;
}
void WavInFile::rewind()
{
int hdrsOk;
fseek(fptr, 0, SEEK_SET);
hdrsOk = readWavHeaders();
assert(hdrsOk == 0);
dataRead = 0;
}
int WavInFile::checkCharTags() const
{
// header.format.fmt should equal to 'fmt '
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
// header.data.data_field should equal to 'data'
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
return 0;
}
int WavInFile::read(unsigned char *buffer, int maxElems)
{
int numBytes;
uint afterDataRead;
// ensure it's 8 bit format
if (header.format.bits_per_sample != 8)
{
ST_THROW_RT_ERROR("Error: WavInFile::read(char*, int) works only with 8bit samples.");
}
assert(sizeof(char) == 1);
numBytes = maxElems;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
assert(buffer);
numBytes = (int)fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
return numBytes;
}
int WavInFile::read(short *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
assert(buffer);
switch (header.format.bits_per_sample)
{
case 8:
{
// 8 bit format
unsigned char *temp = (unsigned char*)getConvBuffer(maxElems);
int i;
numElems = read(temp, maxElems);
// convert from 8 to 16 bit
for (i = 0; i < numElems; i ++)
{
buffer[i] = (short)(((short)temp[i] - 128) * 256);
}
break;
}
case 16:
{
// 16 bit format
assert(sizeof(short) == 2);
numBytes = maxElems * 2;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
numBytes = (int)fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / 2;
// 16bit samples, swap byte order if necessary
_swap16Buffer((short *)buffer, numElems);
break;
}
default:
{
stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
};
return numElems;
}
/// Read data in float format. Notice that when reading in float format
/// 8/16/24/32 bit sample formats are supported
int WavInFile::read(float *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
int bytesPerSample;
assert(buffer);
bytesPerSample = header.format.bits_per_sample / 8;
if ((bytesPerSample < 1) || (bytesPerSample > 4))
{
stringstream ss;
ss << "\nOnly 8/16/24/32 bit sample WAV files supported. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
numBytes = maxElems * bytesPerSample;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = (int)header.data.data_len - (int)dataRead;
assert(numBytes >= 0);
}
// read raw data into temporary buffer
char *temp = (char*)getConvBuffer(numBytes);
numBytes = (int)fread(temp, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / bytesPerSample;
// swap byte ordert & convert to float, depending on sample format
switch (bytesPerSample)
{
case 1:
{
unsigned char *temp2 = (unsigned char*)temp;
double conv = 1.0 / 128.0;
for (int i = 0; i < numElems; i ++)
{
buffer[i] = (float)(temp2[i] * conv - 1.0);
}
break;
}
case 2:
{
short *temp2 = (short*)temp;
double conv = 1.0 / 32768.0;
for (int i = 0; i < numElems; i ++)
{
short value = temp2[i];
buffer[i] = (float)(_swap16(value) * conv);
}
break;
}
case 3:
{
char *temp2 = (char *)temp;
double conv = 1.0 / 8388608.0;
for (int i = 0; i < numElems; i ++)
{
int value = *((int*)temp2);
value = _swap32(value) & 0x00ffffff; // take 24 bits
value |= (value & 0x00800000) ? 0xff000000 : 0; // extend minus sign bits
buffer[i] = (float)(value * conv);
temp2 += 3;
}
break;
}
case 4:
{
int *temp2 = (int *)temp;
double conv = 1.0 / 2147483648.0;
assert(sizeof(int) == 4);
for (int i = 0; i < numElems; i ++)
{
int value = temp2[i];
buffer[i] = (float)(_swap32(value) * conv);
}
break;
}
}
return numElems;
}
int WavInFile::eof() const
{
// return true if all data has been read or file eof has reached
return (dataRead == header.data.data_len || feof(fptr));
}
// test if character code is between a white space ' ' and little 'z'
static int isAlpha(char c)
{
return (c >= ' ' && c <= 'z') ? 1 : 0;
}
// test if all characters are between a white space ' ' and little 'z'
static int isAlphaStr(const char *str)
{
char c;
c = str[0];
while (c)
{
if (isAlpha(c) == 0) return 0;
str ++;
c = str[0];
}
return 1;
}
int WavInFile::readRIFFBlock()
{
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
// swap 32bit data byte order if necessary
_swap32((int &)header.riff.package_len);
// header.riff.riff_char should equal to 'RIFF');
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
// header.riff.wave should equal to 'WAVE'
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
return 0;
}
int WavInFile::readHeaderBlock()
{
char label[5];
string sLabel;
// lead label string
if (fread(label, 1, 4, fptr) !=4) return -1;
label[4] = 0;
if (isAlphaStr(label) == 0) return -1; // not a valid label
// Decode blocks according to their label
if (strcmp(label, fmtStr) == 0)
{
int nLen, nDump;
// 'fmt ' block
memcpy(header.format.fmt, fmtStr, 4);
// read length of the format field
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32(nLen); // int format_len;
header.format.format_len = nLen;
// calculate how much length differs from expected
nDump = nLen - ((int)sizeof(header.format) - 8);
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.format) - 8;
}
// read data
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap16(header.format.fixed); // short int fixed;
_swap16(header.format.channel_number); // short int channel_number;
_swap32((int &)header.format.sample_rate); // int sample_rate;
_swap32((int &)header.format.byte_rate); // int byte_rate;
_swap16(header.format.byte_per_sample); // short int byte_per_sample;
_swap16(header.format.bits_per_sample); // short int bits_per_sample;
// if format_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, factStr) == 0)
{
int nLen, nDump;
// 'fact' block
memcpy(header.fact.fact_field, factStr, 4);
// read length of the fact field
if (fread(&nLen, sizeof(int), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32(nLen); // int fact_len;
header.fact.fact_len = nLen;
// calculate how much length differs from expected
nDump = nLen - ((int)sizeof(header.fact) - 8);
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.fact) - 8;
}
// read data
if (fread(&(header.fact.fact_sample_len), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((int &)header.fact.fact_sample_len); // int sample_length;
// if fact_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, dataStr) == 0)
{
// 'data' block
memcpy(header.data.data_field, dataStr, 4);
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
// swap byte order if necessary
_swap32((int &)header.data.data_len);
return 1;
}
else
{
uint len, i;
uint temp;
// unknown block
// read length
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
// scan through the block
for (i = 0; i < len; i ++)
{
if (fread(&temp, 1, 1, fptr) != 1) return -1;
if (feof(fptr)) return -1; // unexpected eof
}
}
return 0;
}
int WavInFile::readWavHeaders()
{
int res;
memset(&header, 0, sizeof(header));
res = readRIFFBlock();
if (res) return 1;
// read header blocks until data block is found
do
{
// read header blocks
res = readHeaderBlock();
if (res < 0) return 1; // error in file structure
} while (res == 0);
// check that all required tags are legal
return checkCharTags();
}
uint WavInFile::getNumChannels() const
{
return header.format.channel_number;
}
uint WavInFile::getNumBits() const
{
return header.format.bits_per_sample;
}
uint WavInFile::getBytesPerSample() const
{
return getNumChannels() * getNumBits() / 8;
}
uint WavInFile::getSampleRate() const
{
return header.format.sample_rate;
}
uint WavInFile::getDataSizeInBytes() const
{
return header.data.data_len;
}
uint WavInFile::getNumSamples() const
{
if (header.format.byte_per_sample == 0) return 0;
if (header.format.fixed > 1) return header.fact.fact_sample_len;
return header.data.data_len / (unsigned short)header.format.byte_per_sample;
}
uint WavInFile::getLengthMS() const
{
double numSamples;
double sampleRate;
numSamples = (double)getNumSamples();
sampleRate = (double)getSampleRate();
return (uint)(1000.0 * numSamples / sampleRate + 0.5);
}
/// Returns how many milliseconds of audio have so far been read from the file
uint WavInFile::getElapsedMS() const
{
return (uint)(1000.0 * (double)dataRead / (double)header.format.byte_rate);
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavOutFile
//
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = fopen(fileName, "wb");
if (fptr == NULL)
{
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for writing.";
//pmsg = msg.c_str;
ST_THROW_RT_ERROR(msg.c_str());
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = file;
if (fptr == NULL)
{
string msg = "Error : Unable to access output file stream.";
ST_THROW_RT_ERROR(msg.c_str());
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
}
WavOutFile::~WavOutFile()
{
finishHeader();
if (fptr) fclose(fptr);
fptr = NULL;
}
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
{
// fill in the 'riff' part..
// copy string 'RIFF' to riff_char
memcpy(&(header.riff.riff_char), riffStr, 4);
// package_len unknown so far
header.riff.package_len = 0;
// copy string 'WAVE' to wave
memcpy(&(header.riff.wave), waveStr, 4);
// fill in the 'format' part..
// copy string 'fmt ' to fmt
memcpy(&(header.format.fmt), fmtStr, 4);
header.format.format_len = 0x10;
header.format.fixed = 1;
header.format.channel_number = (short)channels;
header.format.sample_rate = (int)sampleRate;
header.format.bits_per_sample = (short)bits;
header.format.byte_per_sample = (short)(bits * channels / 8);
header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
header.format.sample_rate = (int)sampleRate;
// fill in the 'fact' part...
memcpy(&(header.fact.fact_field), factStr, 4);
header.fact.fact_len = 4;
header.fact.fact_sample_len = 0;
// fill in the 'data' part..
// copy string 'data' to data_field
memcpy(&(header.data.data_field), dataStr, 4);
// data_len unknown so far
header.data.data_len = 0;
}
void WavOutFile::finishHeader()
{
// supplement the file length into the header structure
header.riff.package_len = bytesWritten + sizeof(WavHeader) - sizeof(WavRiff) + 4;
header.data.data_len = bytesWritten;
header.fact.fact_sample_len = bytesWritten / header.format.byte_per_sample;
writeHeader();
}
void WavOutFile::writeHeader()
{
WavHeader hdrTemp;
int res;
// swap byte order if necessary
hdrTemp = header;
_swap32((int &)hdrTemp.riff.package_len);
_swap32((int &)hdrTemp.format.format_len);
_swap16((short &)hdrTemp.format.fixed);
_swap16((short &)hdrTemp.format.channel_number);
_swap32((int &)hdrTemp.format.sample_rate);
_swap32((int &)hdrTemp.format.byte_rate);
_swap16((short &)hdrTemp.format.byte_per_sample);
_swap16((short &)hdrTemp.format.bits_per_sample);
_swap32((int &)hdrTemp.data.data_len);
_swap32((int &)hdrTemp.fact.fact_len);
_swap32((int &)hdrTemp.fact.fact_sample_len);
// write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET);
res = (int)fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
if (res != 1)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
// jump back to the end of the file
fseek(fptr, 0, SEEK_END);
}
void WavOutFile::write(const unsigned char *buffer, int numElems)
{
int res;
if (header.format.bits_per_sample != 8)
{
ST_THROW_RT_ERROR("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
}
assert(sizeof(char) == 1);
res = (int)fwrite(buffer, 1, numElems, fptr);
if (res != numElems)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += numElems;
}
void WavOutFile::write(const short *buffer, int numElems)
{
int res;
// 16 bit samples
if (numElems < 1) return; // nothing to do
switch (header.format.bits_per_sample)
{
case 8:
{
int i;
unsigned char *temp = (unsigned char *)getConvBuffer(numElems);
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i ++)
{
temp[i] = (unsigned char)(buffer[i] / 256 + 128);
}
// write in 8bit format
write(temp, numElems);
break;
}
case 16:
{
// 16bit format
// use temp buffer to swap byte order if necessary
short *pTemp = (short *)getConvBuffer(numElems * sizeof(short));
memcpy(pTemp, buffer, numElems * 2);
_swap16Buffer(pTemp, numElems);
res = (int)fwrite(pTemp, 2, numElems, fptr);
if (res != numElems)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += 2 * numElems;
break;
}
default:
{
stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
ss << (int)header.format.bits_per_sample;
ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str());
}
}
}
/// Convert from float to integer and saturate
inline int saturate(float fvalue, float minval, float maxval)
{
if (fvalue > maxval)
{
fvalue = maxval;
}
else if (fvalue < minval)
{
fvalue = minval;
}
return (int)fvalue;
}
void WavOutFile::write(const float *buffer, int numElems)
{
int numBytes;
int bytesPerSample;
if (numElems == 0) return;
bytesPerSample = header.format.bits_per_sample / 8;
numBytes = numElems * bytesPerSample;
short *temp = (short*)getConvBuffer(numBytes);
switch (bytesPerSample)
{
case 1:
{
unsigned char *temp2 = (unsigned char *)temp;
for (int i = 0; i < numElems; i ++)
{
temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
}
break;
}
case 2:
{
short *temp2 = (short *)temp;
for (int i = 0; i < numElems; i ++)
{
short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
temp2[i] = _swap16(value);
}
break;
}
case 3:
{
char *temp2 = (char *)temp;
for (int i = 0; i < numElems; i ++)
{
int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
*((int*)temp2) = _swap32(value);
temp2 += 3;
}
break;
}
case 4:
{
int *temp2 = (int *)temp;
for (int i = 0; i < numElems; i ++)
{
int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
temp2[i] = _swap32(value);
}
break;
}
default:
assert(false);
}
int res = (int)fwrite(temp, 1, numBytes, fptr);
if (res != numBytes)
{
ST_THROW_RT_ERROR("Error while writing to a wav file.");
}
bytesWritten += numBytes;
}

49
3rdparty/soundtouch/WavFile.h → 3rdparty/soundtouch/source/SoundStretch/WavFile.h vendored
View File

@ -16,10 +16,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
// Last changed : $Date: 2014-10-05 16:20:24 +0000 (Sun, 05 Oct 2014) $
// File revision : $Revision: 4 $
//
// $Id: WavFile.h 63 2009-02-21 16:00:14Z oparviai $
// $Id: WavFile.h 200 2014-10-05 16:20:24Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -75,6 +75,14 @@ typedef struct
short bits_per_sample;
} WavFormat;
/// WAV audio file 'fact' section header
typedef struct
{
char fact_field[4];
int fact_len;
uint fact_sample_len;
} WavFact;
/// WAV audio file 'data' section header
typedef struct
{
@ -88,19 +96,40 @@ typedef struct
{
WavRiff riff;
WavFormat format;
WavFact fact;
WavData data;
} WavHeader;
/// Base class for processing WAV audio files.
class WavFileBase
{
private:
/// Conversion working buffer;
char *convBuff;
int convBuffSize;
protected:
WavFileBase();
virtual ~WavFileBase();
/// Get pointer to conversion buffer of at min. given size
void *getConvBuffer(int sizeByte);
};
/// Class for reading WAV audio files.
class WavInFile
class WavInFile : protected WavFileBase
{
private:
/// File pointer.
FILE *fptr;
/// Position within the audio stream
long position;
/// Counter of how many bytes of sample data have been read from the file.
uint dataRead;
long dataRead;
/// WAV header information
WavHeader header;
@ -158,12 +187,17 @@ public:
/// Get the audio file length in milliseconds
uint getLengthMS() const;
/// Returns how many milliseconds of audio have so far been read from the file
///
/// \return elapsed duration in milliseconds
uint getElapsedMS() const;
/// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
/// Reads given number of elements from the file or if end-of-file reached, as many
/// elements as are left in the file.
///
/// \return Number of 8-bit integers read from the file.
int read(char *buffer, int maxElems);
int read(unsigned char *buffer, int maxElems);
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
/// of elements from the file or if end-of-file reached, as many elements as are
@ -177,6 +211,7 @@ public:
/// Reads audio samples from the WAV file to floating point format, converting
/// sample values to range [-1,1[. Reads given number of elements from the file
/// or if end-of-file reached, as many elements as are left in the file.
/// Notice that reading in float format supports 8/16/24/32bit sample formats.
///
/// \return Number of elements read from the file.
int read(float *buffer, ///< Pointer to buffer where to read data.
@ -192,7 +227,7 @@ public:
/// Class for writing WAV audio files.
class WavOutFile
class WavOutFile : protected WavFileBase
{
private:
/// Pointer to the WAV file
@ -230,7 +265,7 @@ public:
/// Write data to WAV file. This function works only with 8bit samples.
/// Throws a 'runtime_error' exception if writing to file fails.
void write(const char *buffer, ///< Pointer to sample data buffer.
void write(const unsigned char *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);

66
3rdparty/soundtouch/AAFilter.cpp → 3rdparty/soundtouch/source/SoundTouch/AAFilter.cpp vendored
View File

@ -12,10 +12,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2009-01-11 09:34:24 -0200 (dom, 11 jan 2009) $
// Last changed : $Date: 2014-01-05 21:40:22 +0000 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -52,6 +52,30 @@ using namespace soundtouch;
#define PI 3.141592655357989
#define TWOPI (2 * PI)
// define this to save AA filter coefficients to a file
// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS 1
#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
#include <stdio.h>
static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
{
FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
if (fptr == NULL) return;
for (int i = 0; i < len; i ++)
{
double temp = coeffs[i];
fprintf(fptr, "%lf\n", temp);
}
fclose(fptr);
}
#else
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
@ -99,7 +123,7 @@ void AAFilter::calculateCoeffs()
{
uint i;
double cntTemp, temp, tempCoeff,h, w;
double fc2, wc;
double wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;
@ -112,8 +136,7 @@ void AAFilter::calculateCoeffs()
work = new double[length];
coeffs = new SAMPLETYPE[length];
fc2 = 2.0 * cutoffFreq;
wc = PI * fc2;
wc = 2.0 * PI * cutoffFreq;
tempCoeff = TWOPI / (double)length;
sum = 0;
@ -124,7 +147,7 @@ void AAFilter::calculateCoeffs()
temp = cntTemp * wc;
if (temp != 0)
{
h = fc2 * sin(temp) / temp; // sinc function
h = sin(temp) / temp; // sinc function
}
else
{
@ -153,17 +176,21 @@ void AAFilter::calculateCoeffs()
for (i = 0; i < length; i ++)
{
// scale & round to nearest integer
temp = work[i] * scaleCoeff;
//#if SOUNDTOUCH_INTEGER_SAMPLES
// scale & round to nearest integer
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
//#endif
coeffs[i] = (SAMPLETYPE)temp;
}
// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
pFIR->setCoefficients(coeffs, length, 14);
_DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
delete[] work;
delete[] coeffs;
}
@ -178,6 +205,31 @@ uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples
}
/// Applies the filter to the given src & dest pipes, so that processed amount of
/// samples get removed from src, and produced amount added to dest
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
{
SAMPLETYPE *pdest;
const SAMPLETYPE *psrc;
uint numSrcSamples;
uint result;
int numChannels = src.getChannels();
assert(numChannels == dest.getChannels());
numSrcSamples = src.numSamples();
psrc = src.ptrBegin();
pdest = dest.ptrEnd(numSrcSamples);
result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
src.receiveSamples(result);
dest.putSamples(result);
return result;
}
uint AAFilter::getLength() const
{
return pFIR->getLength();

13
3rdparty/soundtouch/AAFilter.h → 3rdparty/soundtouch/source/SoundTouch/AAFilter.h vendored
View File

@ -13,10 +13,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 14:26:55 -0200 (dom, 10 fev 2008) $
// Last changed : $Date: 2014-01-07 19:41:23 +0000 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
// $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -45,6 +45,7 @@
#define AAFilter_H
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
namespace soundtouch
{
@ -84,6 +85,14 @@ public:
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
/// Applies the filter to the given src & dest pipes, so that processed amount of
/// samples get removed from src, and produced amount added to dest
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(FIFOSampleBuffer &dest,
FIFOSampleBuffer &src) const;
};
}

5
3rdparty/soundtouch/BPMDetect.cpp → 3rdparty/soundtouch/source/SoundTouch/BPMDetect.cpp vendored
View File

@ -26,10 +26,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-08-30 16:45:25 -0300 (qui, 30 ago 2012) $
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.cpp 149 2012-08-30 19:45:25Z oparviai $
// $Id: BPMDetect.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -226,6 +226,7 @@ void BPMDetect::updateXCorr(int process_samples)
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
pBuffer = buffer->ptrBegin();
#pragma omp parallel for
for (offs = windowStart; offs < windowLen; offs ++)
{
LONG_SAMPLETYPE sum;

2
3rdparty/soundtouch/FIFOSampleBuffer.cpp → 3rdparty/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp vendored
View File

@ -15,7 +15,7 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
// Last changed : $Date: 2012-11-08 18:53:01 +0000 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $

105
3rdparty/soundtouch/FIRFilter.cpp → 3rdparty/soundtouch/source/SoundTouch/FIRFilter.cpp vendored
View File

@ -11,10 +11,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.cpp 131 2011-09-02 18:56:11Z oparviai $
// $Id: FIRFilter.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -72,8 +72,7 @@ FIRFilter::~FIRFilter()
// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE suml, sumr;
int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
@ -87,9 +86,12 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
end = 2 * (numSamples - length);
#pragma omp parallel for
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE suml, sumr;
uint i;
suml = sumr = 0;
ptr = src + j;
@ -130,28 +132,31 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE sum;
int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = numSamples - length;
#pragma omp parallel for
for (j = 0; j < end; j ++)
{
const SAMPLETYPE *pSrc = src + j;
LONG_SAMPLETYPE sum;
uint i;
sum = 0;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
sum += src[i + 0] * filterCoeffs[i + 0] +
src[i + 1] * filterCoeffs[i + 1] +
src[i + 2] * filterCoeffs[i + 2] +
src[i + 3] * filterCoeffs[i + 3];
sum += pSrc[i + 0] * filterCoeffs[i + 0] +
pSrc[i + 1] * filterCoeffs[i + 1] +
pSrc[i + 2] * filterCoeffs[i + 2] +
pSrc[i + 3] * filterCoeffs[i + 3];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor;
@ -161,12 +166,67 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
sum *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
src ++;
}
return end;
}
uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
{
int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
assert(numChannels < 16);
end = numChannels * (numSamples - length);
#pragma omp parallel for
for (j = 0; j < end; j += numChannels)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE sums[16];
uint c, i;
for (c = 0; c < numChannels; c ++)
{
sums[c] = 0;
}
ptr = src + j;
for (i = 0; i < length; i ++)
{
SAMPLETYPE coef=filterCoeffs[i];
for (c = 0; c < numChannels; c ++)
{
sums[c] += ptr[0] * coef;
ptr ++;
}
}
for (c = 0; c < numChannels; c ++)
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sums[c] >>= resultDivFactor;
#else
sums[c] *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j+c] = (SAMPLETYPE)sums[c];
}
}
return numSamples - length;
}
// Set filter coeffiecients and length.
//
// Throws an exception if filter length isn't divisible by 8
@ -199,18 +259,27 @@ uint FIRFilter::getLength() const
//
// Note : The amount of outputted samples is by value of 'filter_length'
// smaller than the amount of input samples.
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
{
assert(numChannels == 1 || numChannels == 2);
assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
if (numChannels == 2)
#ifndef USE_MULTICH_ALWAYS
if (numChannels == 1)
{
return evaluateFilterMono(dest, src, numSamples);
}
else if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
} else {
return evaluateFilterMono(dest, src, numSamples);
}
else
#endif // USE_MULTICH_ALWAYS
{
assert(numChannels > 0);
return evaluateFilterMulti(dest, src, numSamples, numChannels);
}
}

7
3rdparty/soundtouch/FIRFilter.h → 3rdparty/soundtouch/source/SoundTouch/FIRFilter.h vendored
View File

@ -11,10 +11,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-02-13 17:13:57 -0200 (dom, 13 fev 2011) $
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h 104 2011-02-13 19:13:57Z oparviai $
// $Id: FIRFilter.h 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -71,6 +71,7 @@ protected:
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
public:
FIRFilter();
@ -90,7 +91,7 @@ public:
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
uint numChannels);
uint getLength() const;

200
3rdparty/soundtouch/source/SoundTouch/InterpolateCubic.cpp vendored Normal file
View File

@ -0,0 +1,200 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Cubic interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stddef.h>
#include <math.h>
#include "InterpolateCubic.h"
#include "STTypes.h"
using namespace soundtouch;
// cubic interpolation coefficients
static const float _coeffs[]=
{ -0.5f, 1.0f, -0.5f, 0.0f,
1.5f, -2.5f, 0.0f, 1.0f,
-1.5f, 2.0f, 0.5f, 0.0f,
0.5f, -0.5f, 0.0f, 0.0f};
InterpolateCubic::InterpolateCubic()
{
fract = 0;
}
void InterpolateCubic::resetRegisters()
{
fract = 0;
}
/// Transpose mono audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateCubic::transposeMono(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 4;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
float out;
const float x3 = 1.0f;
const float x2 = (float)fract; // x
const float x1 = x2*x2; // x^2
const float x0 = x1*x2; // x^3
float y0, y1, y2, y3;
assert(fract < 1.0);
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
pdest[i] = (SAMPLETYPE)out;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 4;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
const float x3 = 1.0f;
const float x2 = (float)fract; // x
const float x1 = x2*x2; // x^2
const float x0 = x1*x2; // x^3
float y0, y1, y2, y3;
float out0, out1;
assert(fract < 1.0);
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
pdest[2*i] = (SAMPLETYPE)out0;
pdest[2*i+1] = (SAMPLETYPE)out1;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += 2*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose multi-channel audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 4;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
const float x3 = 1.0f;
const float x2 = (float)fract; // x
const float x1 = x2*x2; // x^2
const float x0 = x1*x2; // x^3
float y0, y1, y2, y3;
assert(fract < 1.0);
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
for (int c = 0; c < numChannels; c ++)
{
float out;
out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
pdest[0] = (SAMPLETYPE)out;
pdest ++;
}
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += numChannels*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Cubic interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateCubic.h 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateCubic_H_
#define _InterpolateCubic_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
class InterpolateCubic : public TransposerBase
{
protected:
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
float fract;
public:
InterpolateCubic();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Linear interpolation algorithm.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateLinear.cpp 180 2014-01-06 19:16:02Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <assert.h>
#include <stdlib.h>
#include "InterpolateLinear.h"
using namespace soundtouch;
//////////////////////////////////////////////////////////////////////////////
//
// InterpolateLinearInteger - integer arithmetic implementation
//
/// fixed-point interpolation routine precision
#define SCALE 65536
// Constructor
InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
resetRegisters();
setRate(1.0f);
}
void InterpolateLinearInteger::resetRegisters()
{
iFract = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
LONG_SAMPLETYPE temp;
assert(iFract < SCALE);
temp = (SCALE - iFract) * src[0] + iFract * src[1];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iFract += iRate;
int iWhole = iFract / SCALE;
iFract -= iWhole * SCALE;
srcCount += iWhole;
src += iWhole;
}
srcSamples = srcCount;
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Stereo' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
LONG_SAMPLETYPE temp0;
LONG_SAMPLETYPE temp1;
assert(iFract < SCALE);
temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
dest[0] = (SAMPLETYPE)(temp0 / SCALE);
dest[1] = (SAMPLETYPE)(temp1 / SCALE);
dest += 2;
i++;
iFract += iRate;
int iWhole = iFract / SCALE;
iFract -= iWhole * SCALE;
srcCount += iWhole;
src += 2*iWhole;
}
srcSamples = srcCount;
return i;
}
int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
LONG_SAMPLETYPE temp, vol1;
assert(iFract < SCALE);
vol1 = (SCALE - iFract);
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + iFract * src[c + numChannels];
dest[0] = (SAMPLETYPE)(temp / SCALE);
dest ++;
}
i++;
iFract += iRate;
int iWhole = iFract / SCALE;
iFract -= iWhole * SCALE;
srcCount += iWhole;
src += iWhole * numChannels;
}
srcSamples = srcCount;
return i;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void InterpolateLinearInteger::setRate(float newRate)
{
iRate = (int)(newRate * SCALE + 0.5f);
TransposerBase::setRate(newRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// InterpolateLinearFloat - floating point arithmetic implementation
//
//////////////////////////////////////////////////////////////////////////////
// Constructor
InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
resetRegisters();
setRate(1.0f);
}
void InterpolateLinearFloat::resetRegisters()
{
fract = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out;
assert(fract < 1.0);
out = (1.0 - fract) * src[0] + fract * src[1];
dest[i] = (SAMPLETYPE)out;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
src += whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out0, out1;
assert(fract < 1.0);
out0 = (1.0 - fract) * src[0] + fract * src[2];
out1 = (1.0 - fract) * src[1] + fract * src[3];
dest[2*i] = (SAMPLETYPE)out0;
dest[2*i+1] = (SAMPLETYPE)out1;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
src += 2*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
float temp, vol1;
vol1 = (1.0f- fract);
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + fract * src[c + numChannels];
*dest = (SAMPLETYPE)temp;
dest ++;
}
i++;
fract += rate;
int iWhole = (int)fract;
fract -= iWhole;
srcCount += iWhole;
src += iWhole * numChannels;
}
srcSamples = srcCount;
return i;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Linear interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateLinear.h 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateLinear_H_
#define _InterpolateLinear_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
/// Linear transposer class that uses integer arithmetics
class InterpolateLinearInteger : public TransposerBase
{
protected:
int iFract;
int iRate;
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
public:
InterpolateLinearInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
};
/// Linear transposer class that uses floating point arithmetics
class InterpolateLinearFloat : public TransposerBase
{
protected:
float fract;
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
public:
InterpolateLinearFloat();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
/// with kaiser window.
///
/// Notice. This algorithm is remarkably much heavier than linear or cubic
/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
/// for experimental purposes
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateShannon.cpp 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include "InterpolateShannon.h"
#include "STTypes.h"
using namespace soundtouch;
/// Kaiser window with beta = 2.0
/// Values scaled down by 5% to avoid overflows
static const double _kaiser8[8] =
{
0.41778693317814,
0.64888025049173,
0.83508562409944,
0.93887857733412,
0.93887857733412,
0.83508562409944,
0.64888025049173,
0.41778693317814
};
InterpolateShannon::InterpolateShannon()
{
fract = 0;
}
void InterpolateShannon::resetRegisters()
{
fract = 0;
}
#define PI 3.1415926536
#define sinc(x) (sin(PI * (x)) / (PI * (x)))
/// Transpose mono audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeMono(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 8;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out;
assert(fract < 1.0);
out = psrc[0] * sinc(-3.0 - fract) * _kaiser8[0];
out += psrc[1] * sinc(-2.0 - fract) * _kaiser8[1];
out += psrc[2] * sinc(-1.0 - fract) * _kaiser8[2];
if (fract < 1e-6)
{
out += psrc[3] * _kaiser8[3]; // sinc(0) = 1
}
else
{
out += psrc[3] * sinc(- fract) * _kaiser8[3];
}
out += psrc[4] * sinc( 1.0 - fract) * _kaiser8[4];
out += psrc[5] * sinc( 2.0 - fract) * _kaiser8[5];
out += psrc[6] * sinc( 3.0 - fract) * _kaiser8[6];
out += psrc[7] * sinc( 4.0 - fract) * _kaiser8[7];
pdest[i] = (SAMPLETYPE)out;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 8;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out0, out1, w;
assert(fract < 1.0);
w = sinc(-3.0 - fract) * _kaiser8[0];
out0 = psrc[0] * w; out1 = psrc[1] * w;
w = sinc(-2.0 - fract) * _kaiser8[1];
out0 += psrc[2] * w; out1 += psrc[3] * w;
w = sinc(-1.0 - fract) * _kaiser8[2];
out0 += psrc[4] * w; out1 += psrc[5] * w;
w = _kaiser8[3] * ((fract < 1e-5) ? 1.0 : sinc(- fract)); // sinc(0) = 1
out0 += psrc[6] * w; out1 += psrc[7] * w;
w = sinc( 1.0 - fract) * _kaiser8[4];
out0 += psrc[8] * w; out1 += psrc[9] * w;
w = sinc( 2.0 - fract) * _kaiser8[5];
out0 += psrc[10] * w; out1 += psrc[11] * w;
w = sinc( 3.0 - fract) * _kaiser8[6];
out0 += psrc[12] * w; out1 += psrc[13] * w;
w = sinc( 4.0 - fract) * _kaiser8[7];
out0 += psrc[14] * w; out1 += psrc[15] * w;
pdest[2*i] = (SAMPLETYPE)out0;
pdest[2*i+1] = (SAMPLETYPE)out1;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += 2*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeMulti(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
// not implemented
assert(false);
return 0;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
/// with kaiser window.
///
/// Notice. This algorithm is remarkably much heavier than linear or cubic
/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
/// for experimental purposes
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateShannon.h 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateShannon_H_
#define _InterpolateShannon_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
class InterpolateShannon : public TransposerBase
{
protected:
void resetRegisters();
int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
float fract;
public:
InterpolateShannon();
};
}
#endif

16
3rdparty/soundtouch/PeakFinder.cpp → 3rdparty/soundtouch/source/SoundTouch/PeakFinder.cpp vendored
View File

@ -11,10 +11,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-12-28 17:52:47 -0200 (sex, 28 dez 2012) $
// Last changed : $Date: 2015-05-18 15:22:02 +0000 (Mon, 18 May 2015) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.cpp 164 2012-12-28 19:52:47Z oparviai $
// $Id: PeakFinder.cpp 213 2015-05-18 15:22:02Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -192,11 +192,21 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);
groundLevel = 0.5f * (data[gp1] + data[gp2]);
peakLevel = data[peakpos];
if (gp1 == gp2)
{
// avoid rounding errors when all are equal
assert(gp1 == peakpos);
cutLevel = groundLevel = peakLevel;
} else {
// get average of the ground levels
groundLevel = 0.5f * (data[gp1] + data[gp2]);
// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
}
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);

2
3rdparty/soundtouch/PeakFinder.h → 3rdparty/soundtouch/source/SoundTouch/PeakFinder.h vendored
View File

@ -9,7 +9,7 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-12-30 18:33:46 -0200 (sex, 30 dez 2011) $
// Last changed : $Date: 2011-12-30 20:33:46 +0000 (Fri, 30 Dec 2011) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $

302
3rdparty/soundtouch/source/SoundTouch/RateTransposer.cpp vendored Normal file
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@ -0,0 +1,302 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application)
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include "RateTransposer.h"
#include "InterpolateLinear.h"
#include "InterpolateCubic.h"
#include "InterpolateShannon.h"
#include "AAFilter.h"
using namespace soundtouch;
// Define default interpolation algorithm here
TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
bUseAAFilter = true;
// Instantiates the anti-alias filter
pAAFilter = new AAFilter(64);
pTransposer = TransposerBase::newInstance();
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
delete pTransposer;
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(bool newMode)
{
bUseAAFilter = newMode;
}
/// Returns nonzero if anti-alias filter is enabled.
bool RateTransposer::isAAFilterEnabled() const
{
return bUseAAFilter;
}
AAFilter *RateTransposer::getAAFilter()
{
return pAAFilter;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposer::setRate(float newRate)
{
double fCutoff;
pTransposer->setRate(newRate);
// design a new anti-alias filter
if (newRate > 1.0f)
{
fCutoff = 0.5f / newRate;
}
else
{
fCutoff = 0.5f * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
processSamples(samples, nSamples);
}
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{
uint count;
if (nSamples == 0) return;
// Store samples to input buffer
inputBuffer.putSamples(src, nSamples);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == false)
{
count = pTransposer->transpose(outputBuffer, inputBuffer);
return;
}
assert(pAAFilter);
// Transpose with anti-alias filter
if (pTransposer->rate < 1.0f)
{
// If the parameter 'Rate' value is smaller than 1, first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// Transpose the samples, store the result to end of "midBuffer"
pTransposer->transpose(midBuffer, inputBuffer);
// Apply the anti-alias filter for transposed samples in midBuffer
pAAFilter->evaluate(outputBuffer, midBuffer);
}
else
{
// If the parameter 'Rate' value is larger than 1, first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose.
// Apply the anti-alias filter for samples in inputBuffer
pAAFilter->evaluate(midBuffer, inputBuffer);
// Transpose the AA-filtered samples in "midBuffer"
pTransposer->transpose(outputBuffer, midBuffer);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels)
{
assert(nChannels > 0);
if (pTransposer->numChannels == nChannels) return;
pTransposer->setChannels(nChannels);
inputBuffer.setChannels(nChannels);
midBuffer.setChannels(nChannels);
outputBuffer.setChannels(nChannels);
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
midBuffer.clear();
inputBuffer.clear();
}
// Returns nonzero if there aren't any samples available for outputting.
int RateTransposer::isEmpty() const
{
int res;
res = FIFOProcessor::isEmpty();
if (res == 0) return 0;
return inputBuffer.isEmpty();
}
//////////////////////////////////////////////////////////////////////////////
//
// TransposerBase - Base class for interpolation
//
// static function to set interpolation algorithm
void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
{
TransposerBase::algorithm = a;
}
// Transposes the sample rate of the given samples using linear interpolation.
// Returns the number of samples returned in the "dest" buffer
int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
{
int numSrcSamples = src.numSamples();
int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
int numOutput;
SAMPLETYPE *psrc = src.ptrBegin();
SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
#ifndef USE_MULTICH_ALWAYS
if (numChannels == 1)
{
numOutput = transposeMono(pdest, psrc, numSrcSamples);
}
else if (numChannels == 2)
{
numOutput = transposeStereo(pdest, psrc, numSrcSamples);
}
else
#endif // USE_MULTICH_ALWAYS
{
assert(numChannels > 0);
numOutput = transposeMulti(pdest, psrc, numSrcSamples);
}
dest.putSamples(numOutput);
src.receiveSamples(numSrcSamples);
return numOutput;
}
TransposerBase::TransposerBase()
{
numChannels = 0;
rate = 1.0f;
}
TransposerBase::~TransposerBase()
{
}
void TransposerBase::setChannels(int channels)
{
numChannels = channels;
resetRegisters();
}
void TransposerBase::setRate(float newRate)
{
rate = newRate;
}
// static factory function
TransposerBase *TransposerBase::newInstance()
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
return ::new InterpolateLinearInteger;
#else
switch (algorithm)
{
case LINEAR:
return new InterpolateLinearFloat;
case CUBIC:
return new InterpolateCubic;
case SHANNON:
return new InterpolateShannon;
default:
assert(false);
return NULL;
}
#endif
}

90
3rdparty/soundtouch/RateTransposer.h → 3rdparty/soundtouch/source/SoundTouch/RateTransposer.h vendored
View File

@ -14,10 +14,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2009-02-21 13:00:14 -0300 (sáb, 21 fev 2009) $
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
// $Id: RateTransposer.h 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -55,50 +55,71 @@
namespace soundtouch
{
/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
class TransposerBase
{
public:
enum ALGORITHM {
LINEAR = 0,
CUBIC,
SHANNON
};
protected:
virtual void resetRegisters() = 0;
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
virtual int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
static ALGORITHM algorithm;
public:
float rate;
int numChannels;
TransposerBase();
virtual ~TransposerBase();
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
virtual void setRate(float newRate);
virtual void setChannels(int channels);
// static factory function
static TransposerBase *newInstance();
// static function to set interpolation algorithm
static void setAlgorithm(ALGORITHM a);
};
/// A common linear samplerate transposer class.
///
/// Note: Use function "RateTransposer::newInstance()" to create a new class
/// instance instead of the "new" operator; that function automatically
/// chooses a correct implementation depending on if integer or floating
/// arithmetics are to be used.
class RateTransposer : public FIFOProcessor
{
protected:
/// Anti-alias filter object
AAFilter *pAAFilter;
float fRate;
int numChannels;
TransposerBase *pTransposer;
/// Buffer for collecting samples to feed the anti-alias filter between
/// two batches
FIFOSampleBuffer storeBuffer;
FIFOSampleBuffer inputBuffer;
/// Buffer for keeping samples between transposing & anti-alias filter
FIFOSampleBuffer tempBuffer;
FIFOSampleBuffer midBuffer;
/// Output sample buffer
FIFOSampleBuffer outputBuffer;
BOOL bUseAAFilter;
bool bUseAAFilter;
virtual void resetRegisters() = 0;
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
inline uint transpose(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
void downsample(const SAMPLETYPE *src,
uint numSamples);
void upsample(const SAMPLETYPE *src,
uint numSamples);
/// Transposes sample rate by applying anti-alias filter to prevent folding.
/// Returns amount of samples returned in the "dest" buffer.
@ -107,34 +128,33 @@ protected:
void processSamples(const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposer();
virtual ~RateTransposer();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetics.
static void *operator new(size_t s);
// static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// integer ot floating point arithmetics are to be used.
static RateTransposer *newInstance();
// static RateTransposer *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
FIFOSamplePipe *getStore() { return &storeBuffer; };
// FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter();
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(BOOL newMode);
void enableAAFilter(bool newMode);
/// Returns nonzero if anti-alias filter is enabled.
BOOL isAAFilterEnabled() const;
bool isAAFilterEnabled() const;
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.

45
3rdparty/soundtouch/SoundTouch.cpp → 3rdparty/soundtouch/source/SoundTouch/SoundTouch.cpp vendored
View File

@ -41,10 +41,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 16:29:53 -0300 (qua, 13 jun 2012) $
// Last changed : $Date: 2014-10-08 15:26:57 +0000 (Wed, 08 Oct 2014) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.cpp 143 2012-06-13 19:29:53Z oparviai $
// $Id: SoundTouch.cpp 201 2014-10-08 15:26:57Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -97,7 +97,7 @@ SoundTouch::SoundTouch()
{
// Initialize rate transposer and tempo changer instances
pRateTransposer = RateTransposer::newInstance();
pRateTransposer = new RateTransposer();
pTDStretch = TDStretch::newInstance();
setOutPipe(pTDStretch);
@ -111,7 +111,7 @@ SoundTouch::SoundTouch()
calcEffectiveRateAndTempo();
channels = 0;
bSrateSet = FALSE;
bSrateSet = false;
}
@ -143,10 +143,11 @@ uint SoundTouch::getVersionId()
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (numChannels != 1 && numChannels != 2)
/*if (numChannels != 1 && numChannels != 2)
{
ST_THROW_RT_ERROR("Illegal number of channels");
}
//ST_THROW_RT_ERROR("Illegal number of channels");
return;
}*/
channels = numChannels;
pRateTransposer->setChannels((int)numChannels);
pTDStretch->setChannels((int)numChannels);
@ -254,7 +255,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
pTDStretch->moveSamples(*pRateTransposer->getStore());
// deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
@ -282,7 +283,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
// Sets sample rate.
void SoundTouch::setSampleRate(uint srate)
{
bSrateSet = TRUE;
bSrateSet = true;
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters((int)srate);
}
@ -292,7 +293,7 @@ void SoundTouch::setSampleRate(uint srate)
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
if (bSrateSet == FALSE)
if (bSrateSet == false)
{
ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
}
@ -347,7 +348,7 @@ void SoundTouch::flush()
int i;
int nUnprocessed;
int nOut;
SAMPLETYPE buff[64*2]; // note: allocate 2*64 to cater 64 sample frames of stereo sound
SAMPLETYPE *buff = new SAMPLETYPE[64 * channels];
// check how many samples still await processing, and scale
// that by tempo & rate to get expected output sample count
@ -377,6 +378,8 @@ void SoundTouch::flush()
}
}
delete[] buff;
// Clear working buffers
pRateTransposer->clear();
pTDStretch->clearInput();
@ -387,7 +390,7 @@ void SoundTouch::flush()
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
BOOL SoundTouch::setSetting(int settingId, int value)
bool SoundTouch::setSetting(int settingId, int value)
{
int sampleRate, sequenceMs, seekWindowMs, overlapMs;
@ -398,36 +401,36 @@ BOOL SoundTouch::setSetting(int settingId, int value)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
return TRUE;
pRateTransposer->enableAAFilter((value != 0) ? true : false);
return true;
case SETTING_AA_FILTER_LENGTH :
// sets anti-alias filter length
pRateTransposer->getAAFilter()->setLength(value);
return TRUE;
return true;
case SETTING_USE_QUICKSEEK :
// enables / disables tempo routine quick seeking algorithm
pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
return TRUE;
pTDStretch->enableQuickSeek((value != 0) ? true : false);
return true;
case SETTING_SEQUENCE_MS:
// change time-stretch sequence duration parameter
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
return TRUE;
return true;
case SETTING_SEEKWINDOW_MS:
// change time-stretch seek window length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
return TRUE;
return true;
case SETTING_OVERLAP_MS:
// change time-stretch overlap length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
return TRUE;
return true;
default :
return FALSE;
return false;
}
}

238
3rdparty/soundtouch/TDStretch.cpp → 3rdparty/soundtouch/source/SoundTouch/TDStretch.cpp vendored
View File

@ -13,10 +13,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
// Last changed : $Date: 2015-02-22 15:07:12 +0000 (Sun, 22 Feb 2015) $
// File revision : $Revision: 1.12 $
//
// $Id: TDStretch.cpp 160 2012-11-08 18:53:01Z oparviai $
// $Id: TDStretch.cpp 205 2015-02-22 15:07:12Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -51,8 +51,6 @@
#include "cpu_detect.h"
#include "TDStretch.h"
#include <stdio.h>
using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y))
@ -86,15 +84,15 @@ static const short _scanOffsets[5][24]={
TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
{
bQuickSeek = FALSE;
bQuickSeek = false;
channels = 2;
pMidBuffer = NULL;
pMidBufferUnaligned = NULL;
overlapLength = 0;
bAutoSeqSetting = TRUE;
bAutoSeekSetting = TRUE;
bAutoSeqSetting = true;
bAutoSeekSetting = true;
// outDebt = 0;
skipFract = 0;
@ -134,23 +132,23 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
if (aSequenceMS > 0)
{
this->sequenceMs = aSequenceMS;
bAutoSeqSetting = FALSE;
bAutoSeqSetting = false;
}
else if (aSequenceMS == 0)
{
// if zero, use automatic setting
bAutoSeqSetting = TRUE;
bAutoSeqSetting = true;
}
if (aSeekWindowMS > 0)
{
this->seekWindowMs = aSeekWindowMS;
bAutoSeekSetting = FALSE;
bAutoSeekSetting = false;
}
else if (aSeekWindowMS == 0)
{
// if zero, use automatic setting
bAutoSeekSetting = TRUE;
bAutoSeekSetting = true;
}
calcSeqParameters();
@ -159,7 +157,6 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
// set tempo to recalculate 'sampleReq'
setTempo(tempo);
}
@ -212,7 +209,7 @@ void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
void TDStretch::clearMidBuffer()
{
memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
memset(pMidBuffer, 0, channels * sizeof(SAMPLETYPE) * overlapLength);
}
@ -234,14 +231,14 @@ void TDStretch::clear()
// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
// to enable
void TDStretch::enableQuickSeek(BOOL enable)
void TDStretch::enableQuickSeek(bool enable)
{
bQuickSeek = enable;
}
// Returns nonzero if the quick seeking algorithm is enabled.
BOOL TDStretch::isQuickSeekEnabled() const
bool TDStretch::isQuickSeekEnabled() const
{
return bQuickSeek;
}
@ -265,13 +262,22 @@ int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
// of 'ovlPos'.
inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const
{
if (channels == 2)
#ifndef USE_MULTICH_ALWAYS
if (channels == 1)
{
// mono sound.
overlapMono(pOutput, pInput + ovlPos);
}
else if (channels == 2)
{
// stereo sound
overlapStereo(pOutput, pInput + 2 * ovlPos);
} else {
// mono sound.
overlapMono(pOutput, pInput + ovlPos);
}
else
#endif // USE_MULTICH_ALWAYS
{
assert(channels > 0);
overlapMulti(pOutput, pInput + channels * ovlPos);
}
}
@ -286,30 +292,50 @@ inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, ui
int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
{
int bestOffs;
double bestCorr, corr;
double bestCorr;
int i;
double norm;
bestCorr = FLT_MIN;
bestOffs = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
for (i = 0; i < seekLength; i ++)
bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
#pragma omp parallel for
for (i = 1; i < seekLength; i ++)
{
// Calculates correlation value for the mixing position corresponding
// to 'i'
corr = calcCrossCorr(refPos + channels * i, pMidBuffer);
double corr;
// Calculates correlation value for the mixing position corresponding to 'i'
#ifdef _OPENMP
// in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't
// iterate the loop in sequential order
corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm);
#else
// In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same
// as "calcCrossCorr", but saves time by reusing & updating previously stored
// "norm" value
corr = calcCrossCorrAccumulate(refPos + channels * i, pMidBuffer, norm);
#endif
// heuristic rule to slightly favour values close to mid of the range
double tmp = (double)(2 * i - seekLength) / (double)seekLength;
corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
// Checks for the highest correlation value
if (corr > bestCorr)
{
// For optimal performance, enter critical section only in case that best value found.
// in such case repeat 'if' condition as it's possible that parallel execution may have
// updated the bestCorr value in the mean time
#pragma omp critical
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = i;
}
}
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
@ -346,12 +372,13 @@ int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
j = 0;
while (_scanOffsets[scanCount][j])
{
double norm;
tempOffset = corrOffset + _scanOffsets[scanCount][j];
if (tempOffset >= seekLength) break;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer);
corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer, norm);
// heuristic rule to slightly favour values close to mid of the range
double tmp = (double)(2 * tempOffset - seekLength) / seekLength;
corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
@ -458,11 +485,15 @@ void TDStretch::setChannels(int numChannels)
{
assert(numChannels > 0);
if (channels == numChannels) return;
assert(numChannels == 1 || numChannels == 2);
// assert(numChannels == 1 || numChannels == 2);
channels = numChannels;
inputBuffer.setChannels(channels);
outputBuffer.setChannels(channels);
// re-init overlap/buffer
overlapLength=0;
setParameters(sampleRate);
}
@ -498,7 +529,6 @@ void TDStretch::processNominalTempo()
}
*/
#include <stdio.h>
// Processes as many processing frames of the samples 'inputBuffer', store
// the result into 'outputBuffer'
@ -588,7 +618,7 @@ void TDStretch::acceptNewOverlapLength(int newOverlapLength)
{
delete[] pMidBufferUnaligned;
pMidBufferUnaligned = new SAMPLETYPE[overlapLength * 2 + 16 / sizeof(SAMPLETYPE)];
pMidBufferUnaligned = new SAMPLETYPE[overlapLength * channels + 16 / sizeof(SAMPLETYPE)];
// ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency
pMidBuffer = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(pMidBufferUnaligned);
@ -666,6 +696,27 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
}
}
// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi'
// version of the routine.
void TDStretch::overlapMulti(SAMPLETYPE *poutput, const SAMPLETYPE *input) const
{
SAMPLETYPE m1=(SAMPLETYPE)0;
SAMPLETYPE m2;
int i=0;
for (m2 = (SAMPLETYPE)overlapLength; m2; m2 --)
{
for (int c = 0; c < channels; c ++)
{
poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength;
i++;
}
m1++;
}
}
// Calculates the x having the closest 2^x value for the given value
static int _getClosest2Power(double value)
{
@ -699,32 +750,72 @@ void TDStretch::calculateOverlapLength(int aoverlapMs)
}
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare) const
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const
{
long corr;
long norm;
long lnorm;
int i;
corr = norm = 0;
corr = lnorm = 0;
// Same routine for stereo and mono. For stereo, unroll loop for better
// efficiency and gives slightly better resolution against rounding.
// For mono it same routine, just unrolls loop by factor of 4
for (i = 0; i < channels * overlapLength; i += 4)
{
corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1] +
mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
corr += (mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
norm += (mixingPos[i] * mixingPos[i] +
mixingPos[i + 1] * mixingPos[i + 1] +
mixingPos[i + 2] * mixingPos[i + 2] +
lnorm += (mixingPos[i] * mixingPos[i] +
mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
lnorm += (mixingPos[i + 2] * mixingPos[i + 2] +
mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits;
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
if (norm == 0) norm = 1; // to avoid div by zero
return (double)corr / sqrt((double)norm);
norm = (double)lnorm;
return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
}
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const
{
long corr;
long lnorm;
int i;
// cancel first normalizer tap from previous round
lnorm = 0;
for (i = 1; i <= channels; i ++)
{
lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBits;
}
corr = 0;
// Same routine for stereo and mono. For stereo, unroll loop for better
// efficiency and gives slightly better resolution against rounding.
// For mono it same routine, just unrolls loop by factor of 4
for (i = 0; i < channels * overlapLength; i += 4)
{
corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits; // notice: do intermediate division here to avoid integer overflow
corr += (mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
}
// update normalizer with last samples of this round
for (int j = 0; j < channels; j ++)
{
i --;
lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBits;
}
norm += (double)lnorm;
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
}
#endif // SOUNDTOUCH_INTEGER_SAMPLES
@ -760,6 +851,34 @@ void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
}
// Overlaps samples in 'midBuffer' with the samples in 'input'.
void TDStretch::overlapMulti(float *pOutput, const float *pInput) const
{
int i;
float fScale;
float f1;
float f2;
fScale = 1.0f / (float)overlapLength;
f1 = 0;
f2 = 1.0f;
i=0;
for (int i2 = 0; i2 < overlapLength; i2 ++)
{
// note: Could optimize this slightly by taking into account that always channels > 2
for (int c = 0; c < channels; c ++)
{
pOutput[i] = pInput[i] * f1 + pMidBuffer[i] * f2;
i++;
}
f1 += fScale;
f2 -= fScale;
}
}
/// Calculates overlapInMsec period length in samples.
void TDStretch::calculateOverlapLength(int overlapInMsec)
{
@ -776,7 +895,8 @@ void TDStretch::calculateOverlapLength(int overlapInMsec)
}
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) const
/// Calculate cross-correlation
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm) const
{
double corr;
double norm;
@ -801,8 +921,44 @@ double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) co
mixingPos[i + 3] * mixingPos[i + 3];
}
if (norm < 1e-9) norm = 1.0; // to avoid div by zero
return corr / sqrt(norm);
anorm = norm;
return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
}
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const
{
double corr;
int i;
corr = 0;
// cancel first normalizer tap from previous round
for (i = 1; i <= channels; i ++)
{
norm -= mixingPos[-i] * mixingPos[-i];
}
// Same routine for stereo and mono. For Stereo, unroll by factor of 2.
// For mono it's same routine yet unrollsd by factor of 4.
for (i = 0; i < channels * overlapLength; i += 4)
{
corr += mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1] +
mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3];
}
// update normalizer with last samples of this round
for (int j = 0; j < channels; j ++)
{
i --;
norm += mixingPos[i] * mixingPos[i];
}
return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
}
#endif // SOUNDTOUCH_FLOAT_SAMPLES

24
3rdparty/soundtouch/TDStretch.h → 3rdparty/soundtouch/source/SoundTouch/TDStretch.h vendored
View File

@ -13,10 +13,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-04-01 16:49:30 -0300 (dom, 01 abr 2012) $
// Last changed : $Date: 2014-04-06 15:57:21 +0000 (Sun, 06 Apr 2014) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.h 137 2012-04-01 19:49:30Z oparviai $
// $Id: TDStretch.h 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -125,21 +125,22 @@ protected:
float skipFract;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
BOOL bQuickSeek;
bool bQuickSeek;
int sampleRate;
int sequenceMs;
int seekWindowMs;
int overlapMs;
BOOL bAutoSeqSetting;
BOOL bAutoSeekSetting;
bool bAutoSeqSetting;
bool bAutoSeekSetting;
void acceptNewOverlapLength(int newOverlapLength);
virtual void clearCrossCorrState();
void calculateOverlapLength(int overlapMs);
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm) const;
virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
@ -147,6 +148,7 @@ protected:
virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
void clearMidBuffer();
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
@ -193,10 +195,10 @@ public:
/// Enables/disables the quick position seeking algorithm. Zero to disable,
/// nonzero to enable
void enableQuickSeek(BOOL enable);
void enableQuickSeek(bool enable);
/// Returns nonzero if the quick seeking algorithm is enabled.
BOOL isQuickSeekEnabled() const;
bool isQuickSeekEnabled() const;
/// Sets routine control parameters. These control are certain time constants
/// defining how the sound is stretched to the desired duration.
@ -247,7 +249,8 @@ public:
class TDStretchMMX : public TDStretch
{
protected:
double calcCrossCorr(const short *mixingPos, const short *compare) const;
double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) const;
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) const;
virtual void overlapStereo(short *output, const short *input) const;
virtual void clearCrossCorrState();
};
@ -259,7 +262,8 @@ public:
class TDStretchSSE : public TDStretch
{
protected:
double calcCrossCorr(const float *mixingPos, const float *compare) const;
double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) const;
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) const;
};
#endif /// SOUNDTOUCH_ALLOW_SSE

2
3rdparty/soundtouch/cpu_detect.h → 3rdparty/soundtouch/source/SoundTouch/cpu_detect.h vendored
View File

@ -12,7 +12,7 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 14:26:55 -0200 (dom, 10 fev 2008) $
// Last changed : $Date: 2008-02-10 16:26:55 +0000 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $

9
3rdparty/soundtouch/cpu_detect_x86.cpp → 3rdparty/soundtouch/source/SoundTouch/cpu_detect_x86.cpp vendored
View File

@ -11,10 +11,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 16:44:37 -0200 (qui, 08 nov 2012) $
// Last changed : $Date: 2014-01-07 18:24:28 +0000 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86.cpp 159 2012-11-08 18:44:37Z oparviai $
// $Id: cpu_detect_x86.cpp 183 2014-01-07 18:24:28Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -42,6 +42,7 @@
#include "cpu_detect.h"
#include "STTypes.h"
#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
#if defined(__GNUC__) && defined(__i386__)
@ -50,13 +51,13 @@
#elif defined(_M_IX86)
// windows non-gcc
#include <intrin.h>
#endif
#define bit_MMX (1 << 23)
#define bit_SSE (1 << 25)
#define bit_SSE2 (1 << 26)
#endif
#endif
//////////////////////////////////////////////////////////////////////////////
//

102
3rdparty/soundtouch/mmx_optimized.cpp → 3rdparty/soundtouch/source/SoundTouch/mmx_optimized.cpp vendored
View File

@ -20,10 +20,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
// Last changed : $Date: 2015-02-22 15:10:38 +0000 (Sun, 22 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: mmx_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
// $Id: mmx_optimized.cpp 206 2015-02-22 15:10:38Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -68,7 +68,7 @@ using namespace soundtouch;
// Calculates cross correlation of two buffers
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm) const
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
@ -93,19 +93,19 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
// _mm_add_pi32 : 2*32bit add
// _m_psrad : 32bit right-shift
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
_mm_madd_pi16(pVec1[1], pVec2[1]));
temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),
_mm_madd_pi16(pVec1[1], pVec1[1]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
accu = _mm_add_pi32(accu, temp);
normaccu = _mm_add_pi32(normaccu, temp2);
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
_mm_madd_pi16(pVec1[3], pVec2[3]));
temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),
_mm_madd_pi16(pVec1[3], pVec1[3]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
accu = _mm_add_pi32(accu, temp);
normaccu = _mm_add_pi32(normaccu, temp2);
pVec1 += 4;
pVec2 += 4;
@ -125,14 +125,81 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
if (norm == 0) norm = 1; // to avoid div by zero
dnorm = (double)norm;
return (double)corr / sqrt((double)norm);
return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm);
// Note: Warning about the missing EMMS instruction is harmless
// as it'll be called elsewhere.
}
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm) const
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
__m64 accu;
long corr, lnorm;
int i;
// cancel first normalizer tap from previous round
lnorm = 0;
for (i = 1; i <= channels; i ++)
{
lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBits;
}
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBits);
accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
// during each round for improved CPU-level parallellization.
for (i = 0; i < channels * overlapLength / 16; i ++)
{
__m64 temp;
// dictionary of instructions:
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
// _mm_add_pi32 : 2*32bit add
// _m_psrad : 32bit right-shift
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
accu = _mm_add_pi32(accu, temp);
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
accu = _mm_add_pi32(accu, temp);
pVec1 += 4;
pVec2 += 4;
}
// copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
// and finally store the result into the variable "corr"
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
corr = _m_to_int(accu);
// Clear MMS state
_m_empty();
// update normalizer with last samples of this round
pV1 = (short *)pVec1;
for (int j = 1; j <= channels; j ++)
{
lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBits;
}
dnorm += (double)lnorm;
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
}
void TDStretchMMX::clearCrossCorrState()
{
@ -220,6 +287,7 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
FIRFilterMMX::FIRFilterMMX() : FIRFilter()
{
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}

31
3rdparty/soundtouch/sse_optimized.cpp → 3rdparty/soundtouch/source/SoundTouch/sse_optimized.cpp vendored
View File

@ -23,10 +23,10 @@
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 16:53:01 -0200 (qui, 08 nov 2012) $
// Last changed : $Date: 2015-02-21 21:24:29 +0000 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: sse_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
// $Id: sse_optimized.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
@ -71,7 +71,7 @@ using namespace soundtouch;
#include <math.h>
// Calculates cross correlation of two buffers
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm) const
{
int i;
const float *pVec1;
@ -141,11 +141,11 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
// return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
float *pvNorm = (float*)&vNorm;
double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
if (norm < 1e-9) norm = 1.0; // to avoid div by zero
float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
anorm = norm;
float *pvSum = (float*)&vSum;
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : norm);
/* This is approximately corresponding routine in C-language yet without normalization:
double corr, norm;
@ -182,6 +182,16 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const
}
double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm) const
{
// call usual calcCrossCorr function because SSE does not show big benefit of
// accumulating "norm" value, and also the "norm" rolling algorithm would get
// complicated due to SSE-specific alignment-vs-nonexact correlation rules.
return calcCrossCorr(pV1, pV2, norm);
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'FIRFilter'
@ -249,14 +259,17 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
#pragma omp parallel for
for (j = 0; j < count; j += 2)
{
const float *pSrc;
float *pDest;
const __m128 *pFil;
__m128 sum1, sum2;
uint i;
pSrc = (const float*)source; // source audio data
pSrc = (const float*)source + j * 2; // source audio data
pDest = dest + j * 2; // destination audio data
pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
sum1 = sum2 = _mm_setzero_ps();
@ -289,12 +302,10 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
// to sum the two hi- and lo-floats of these registers together.
// post-shuffle & add the filtered values and store to dest.
_mm_storeu_ps(dest, _mm_add_ps(
_mm_storeu_ps(pDest, _mm_add_ps(
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
));
source += 4;
dest += 4;
}
// Ideas for further improvement:

21
PCSX2_suite.sln
View File

@ -185,25 +185,32 @@ Global
{18E42F6F-3A62-41EE-B42F-79366C4F1E95}.Release|x64.Build.0 = Release SSE2|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.ActiveCfg = Debug|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.Build.0 = Debug|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.Build.0 = Debug|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.ActiveCfg = Devel|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.Build.0 = Devel|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.Build.0 = Devel|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|Win32.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|Win32.Build.0 = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|x64.ActiveCfg = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX|x64.Build.0 = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|Win32.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|Win32.Build.0 = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|x64.ActiveCfg = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release AVX2|x64.Build.0 = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|Win32.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|Win32.Build.0 = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|x64.ActiveCfg = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSE4|x64.Build.0 = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|Win32.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|Win32.Build.0 = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|x64.ActiveCfg = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release SSSE3|x64.Build.0 = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.Build.0 = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.Build.0 = Release|x64
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.ActiveCfg = Debug|Win32
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.Build.0 = Debug|Win32
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|x64.ActiveCfg = Debug|Win32

2
common/vsprops/3rdpartyDeps.props
View File

@ -6,7 +6,7 @@
</PropertyGroup>
<ItemDefinitionGroup>
<ClCompile>
<AdditionalIncludeDirectories>$(SvnRootDir)\3rdparty\;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
<AdditionalIncludeDirectories>$(SvnRootDir)\3rdparty\;$(SvnRootDir)\3rdparty\soundtouch\;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
</ClCompile>
<Link>
<AdditionalLibraryDirectories>$(SvnRootDir)\deps\$(Platform)\$(Configuration);%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

9
old_plugins.sln
View File

@ -97,13 +97,16 @@ Global
{5F78E90B-BD22-47B1-9CA5-7A80F4DF5EF3}.Release|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.ActiveCfg = Debug|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|Win32.Build.0 = Debug|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.ActiveCfg = Debug|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Debug|x64.Build.0 = Debug|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.ActiveCfg = Devel|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|Win32.Build.0 = Devel|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.ActiveCfg = Devel|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Devel|x64.Build.0 = Devel|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|Win32.Build.0 = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|Win32
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.ActiveCfg = Release|x64
{E9B51944-7E6D-4BCD-83F2-7BBD5A46182D}.Release|x64.Build.0 = Release|x64
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.ActiveCfg = Debug|Win32
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|Win32.Build.0 = Debug|Win32
{2F6C0388-20CB-4242-9F6C-A6EBB6A83F47}.Debug|x64.ActiveCfg = Debug|Win32

2
plugins/spu2-x/src/Wavedump_wav.cpp
View File

@ -19,7 +19,7 @@
#ifdef __linux__
#include "WavFile.h"
#else
#include "soundtouch/WavFile.h"
#include "soundtouch/source/SoundStretch/WavFile.h"
#endif
static WavOutFile* _new_WavOutFile( const char* destfile )

2
plugins/zerospu2/zerospu2.cpp
View File

@ -31,7 +31,7 @@
#ifdef __linux__
#include "WavFile.h"
#else
#include "soundtouch/WavFile.h"
#include "soundtouch/source/SoundStretch/WavFile.h"
#endif
char libraryName[256];

2
plugins/zerospu2/zeroworker.cpp
View File

@ -22,7 +22,7 @@
#ifdef __linux__
#include "WavFile.h"
#else
#include "soundtouch/WavFile.h"
#include "soundtouch/source/SoundStretch/WavFile.h"
#endif
s32 g_logsound = 0;