mirror of https://github.com/PCSX2/pcsx2.git
3rdparty/soundtouch: Bump to v2.3.3
This commit is contained in:
parent
30e7de7555
commit
ebc3923b35
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@ -15,8 +15,8 @@
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<body class="normal">
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<hr>
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<h1>SoundTouch audio processing library v2.3.1</h1>
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<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2021</p>
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<h1>SoundTouch audio processing library v2.3.3</h1>
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<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2024</p>
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<hr>
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<h2>1. Introduction </h2>
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<p>SoundTouch is an open-source audio processing library that allows
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@ -35,7 +35,7 @@
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<p>Author email: oparviai 'at' iki.fi </p>
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<p>SoundTouch WWW page: <a href="http://soundtouch.surina.net">http://soundtouch.surina.net</a></p>
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<p>SoundTouch git repository: <a
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href="https://gitlab.com/soundtouch/soundtouch.git">https://gitlab.com/soundtouch/soundtouch.git</a></p>
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href="https://codeberg.org/soundtouch/soundtouch.git">https://codeberg.org/soundtouch/soundtouch.git</a></p>
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<hr>
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<h2>2. Compiling SoundTouch</h2>
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<p>Before compiling, notice that you can choose the sample data format if it's
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@ -131,10 +131,12 @@
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</table>
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<b>Compiling portable Shared Library / DLL version</b>
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<p> The GNU autotools compilation does not automatically create a shared-library version of
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SoundTouch (.so or .dll) that features position-independent code and C-language
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api that are more suitable for cross-language development than C++ libraries.</p>
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<p> Use script "make-gnu-dll-sh" to build a portable dynamic library version if such is desired.</p>
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<p> The GNU autotools compilation automatically builds an additional dynamic-link version
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of SoundTouch library that features position-independent code and "C"-style API that is
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more suitable for calling the SoundTouch routines from other programming languages.</p>
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<p>This dynamic-link library is built under source/SoundTouchDLL directory, whose
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subdirectories also comtain simple example apps that use the dynamic-link library.
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</p>
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<h4><b>2.2.2 Compiling with cmake</b></h4>
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<p>'cmake' build scripts are provided as an alternative to the autotools toolchain.</p>
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@ -145,6 +147,9 @@
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cmake .
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make -j
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make install</pre>
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<p>To list available build options:</p>
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<pre>
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cmake -LH</pre>
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<p>To compile the additional portable Shared Library / DLL version with the native C-language API:</p>
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<pre>
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cmake . -DSOUNDTOUCH_DLL=ON
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@ -448,7 +453,7 @@
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<h2><a name="SoundStretch"></a>4. SoundStretch audio processing utility
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</h2>
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<p>SoundStretch audio processing utility<br>
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Copyright (c) Olli Parviainen 2002-2015</p>
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Copyright (c) Olli Parviainen 2002-2024</p>
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<p>SoundStretch is a simple command-line application that can change
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tempo, pitch and playback rates of WAV sound files. This program is
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intended primarily to demonstrate how the "SoundTouch" library can be
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@ -603,6 +608,18 @@
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<hr>
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<h2>5. Change History</h2>
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<h3>5.1. SoundTouch library Change History </h3>
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<p><b>2.3.3:</b></p>
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<ul class="current">
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<li>Fixing compiler warnings, maintenance fixes to make/build files for various systems
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</li>
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</ul>
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<p><b>2.3.2:</b></p>
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<ul>
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<li>Improve autotools makefiles to build the `SoundTouchDLL` dynamic-link link library with
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C-style API. This library variation is easier to import and use from other programming
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languages than the default C++ library.
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</li>
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</ul>
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<p><b>2.3.1:</b></p>
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<ul>
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<li>Adjusted cmake build settings and header files that cmake installs</li>
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@ -865,11 +882,14 @@
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<li> Initial release</li>
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</ul>
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<h3>5.2. SoundStretch application Change History </h3>
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<p><b>2.3.3:</b></p>
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<ul class="current_soundstretch">
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<li>Added support for Asian / non-latin filenames in Windows. Gnu platform has supported them already earlier.</li>
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</ul>
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<p><b>1.9:</b></p>
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<ul>
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<li>Added support for WAV file 'fact' information chunk.</li>
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</ul>
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<p><b>1.7.0:</b></p>
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<ul>
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<li>Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer
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@ -966,6 +986,7 @@
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<li> Michael Pruett</li>
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<li> Rajeev Puran</li>
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<li> RJ Ryan</li>
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<li> Serge Sans Paille</li>
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<li> John Sheehy</li>
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<li> Tim Shuttleworth</li>
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<li> Albert Sirvent</li>
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@ -196,7 +196,7 @@ namespace soundtouch
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/// - "values" receive array of beat detection strengths
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/// - max_num indicates max.size of "pos" and "values" array.
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///
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/// You can query a suitable array sized by calling this with NULL in "pos" & "values".
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/// You can query a suitable array sized by calling this with nullptr in "pos" & "values".
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///
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/// \return number of beats in the arrays.
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int getBeats(float *pos, float *strength, int max_num);
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@ -88,11 +88,11 @@ public:
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void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
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)
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{
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int oNumSamples = other.numSamples();
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const uint oNumSamples = other.numSamples();
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putSamples(other.ptrBegin(), oNumSamples);
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other.receiveSamples(oNumSamples);
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};
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}
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/// Output samples from beginning of the sample buffer. Copies requested samples to
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/// output buffer and removes them from the sample buffer. If there are less than
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@ -144,8 +144,8 @@ protected:
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/// Sets output pipe.
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void setOutPipe(FIFOSamplePipe *pOutput)
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{
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assert(output == NULL);
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assert(pOutput != NULL);
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assert(output == nullptr);
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assert(pOutput != nullptr);
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output = pOutput;
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}
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@ -153,7 +153,7 @@ protected:
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/// 'setOutPipe' function.
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FIFOProcessor()
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{
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output = NULL;
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output = nullptr;
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}
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/// Constructor. Configures output pipe.
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@ -72,10 +72,10 @@ namespace soundtouch
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{
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/// Soundtouch library version string
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#define SOUNDTOUCH_VERSION "2.3.1"
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#define SOUNDTOUCH_VERSION "2.3.3"
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/// SoundTouch library version id
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#define SOUNDTOUCH_VERSION_ID (20301)
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#define SOUNDTOUCH_VERSION_ID (20303)
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//
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// Available setting IDs for the 'setSetting' & 'get_setting' functions:
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@ -0,0 +1,52 @@
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////////////////////////////////////////////////////////////////////////////////
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///
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/// Char type for SoundStretch
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///
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/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai 'at' iki.fi
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/// SoundTouch WWW: http://www.surina.net/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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//
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// License :
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//
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// SoundTouch audio processing library
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// Copyright (c) Olli Parviainen
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//
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// This library is free software; you can redistribute it and/or
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// modify it under the terms of the GNU Lesser General Public
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// License as published by the Free Software Foundation; either
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// version 2.1 of the License, or (at your option) any later version.
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//
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// This library is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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// Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public
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// License along with this library; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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//
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////////////////////////////////////////////////////////////////////////////////
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#ifndef SS_CHARTYPE_H
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#define SS_CHARTYPE_H
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#include <string>
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namespace soundstretch
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{
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#if _WIN32
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// wide-char types for supporting non-latin file paths in Windows
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using CHARTYPE = wchar_t;
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using STRING = std::wstring;
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#define STRING_CONST(x) (L"" x)
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#else
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// gnu platform can natively support UTF-8 paths using "char*" set
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using CHARTYPE = char;
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using STRING = std::string;
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#define STRING_CONST(x) (x)
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#endif
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}
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#endif //SS_CHARTYPE_H
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@ -42,14 +42,23 @@
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#include <string>
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#include <sstream>
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#include <cstring>
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#include <assert.h>
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#include <limits.h>
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#include <cassert>
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#include <climits>
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#include "WavFile.h"
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#include "STTypes.h"
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using namespace std;
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namespace soundstretch
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{
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#if _WIN32
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#define FOPEN(name, mode) _wfopen(name, STRING_CONST(mode))
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#else
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#define FOPEN(name, mode) fopen(name, mode)
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#endif
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static const char riffStr[] = "RIFF";
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static const char waveStr[] = "WAVE";
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static const char fmtStr[] = "fmt ";
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}
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// dummy helper-function
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static inline void _swap16Buffer(short *pData, int numBytes)
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static inline void _swap16Buffer(short*, int)
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{
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// do nothing
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}
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@ -138,7 +147,7 @@ static const char dataStr[] = "data";
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WavFileBase::WavFileBase()
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{
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convBuff = NULL;
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convBuff = nullptr;
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convBuffSize = 0;
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}
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@ -169,17 +178,13 @@ void *WavFileBase::getConvBuffer(int sizeBytes)
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// Class WavInFile
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//
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WavInFile::WavInFile(const char *fileName)
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WavInFile::WavInFile(const STRING& fileName)
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{
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// Try to open the file for reading
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fptr = fopen(fileName, "rb");
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if (fptr == NULL)
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fptr = FOPEN(fileName.c_str(), "rb");
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if (fptr == nullptr)
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{
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// didn't succeed
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string msg = "Error : Unable to open file \"";
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msg += fileName;
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msg += "\" for reading.";
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ST_THROW_RT_ERROR(msg.c_str());
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ST_THROW_RT_ERROR("Error : Unable to open file for reading.");
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}
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init();
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@ -192,9 +197,7 @@ WavInFile::WavInFile(FILE *file)
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fptr = file;
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if (!file)
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{
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// didn't succeed
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string msg = "Error : Unable to access input stream for reading";
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ST_THROW_RT_ERROR(msg.c_str());
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ST_THROW_RT_ERROR("Error : Unable to access input stream for reading");
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}
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init();
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hdrsOk = readWavHeaders();
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if (hdrsOk != 0)
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{
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// Something didn't match in the wav file headers
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ST_THROW_RT_ERROR("Input file is corrupt or not a WAV file");
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}
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|
@ -223,7 +225,6 @@ void WavInFile::init()
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(header.format.byte_per_sample < 1) || (header.format.byte_per_sample > 320) ||
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(header.format.bits_per_sample < 8) || (header.format.bits_per_sample > 32))
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{
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// Something didn't match in the wav file headers
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ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters.");
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}
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|
@ -234,7 +235,7 @@ void WavInFile::init()
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WavInFile::~WavInFile()
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{
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if (fptr) fclose(fptr);
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fptr = NULL;
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fptr = nullptr;
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}
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@ -450,7 +451,7 @@ int WavInFile::read(float *buffer, int maxElems)
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int WavInFile::eof() const
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{
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// return true if all data has been read or file eof has reached
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return (dataRead == header.data.data_len || feof(fptr));
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return ((uint)dataRead == header.data.data_len || feof(fptr));
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}
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@ -703,17 +704,13 @@ uint WavInFile::getElapsedMS() const
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// Class WavOutFile
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//
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WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
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WavOutFile::WavOutFile(const STRING& fileName, int sampleRate, int bits, int channels)
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{
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bytesWritten = 0;
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fptr = fopen(fileName, "wb");
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if (fptr == NULL)
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fptr = FOPEN(fileName.c_str(), "wb");
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if (fptr == nullptr)
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{
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string msg = "Error : Unable to open file \"";
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msg += fileName;
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msg += "\" for writing.";
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//pmsg = msg.c_str;
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ST_THROW_RT_ERROR(msg.c_str());
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ST_THROW_RT_ERROR("Error : Unable to open file for writing.");
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}
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fillInHeader(sampleRate, bits, channels);
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{
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bytesWritten = 0;
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fptr = file;
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if (fptr == NULL)
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if (fptr == nullptr)
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{
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string msg = "Error : Unable to access output file stream.";
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ST_THROW_RT_ERROR(msg.c_str());
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ST_THROW_RT_ERROR("Error : Unable to access output file stream.");
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}
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fillInHeader(sampleRate, bits, channels);
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|
@ -740,7 +736,7 @@ WavOutFile::~WavOutFile()
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{
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finishHeader();
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if (fptr) fclose(fptr);
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fptr = NULL;
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fptr = nullptr;
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}
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|
@ -875,7 +871,7 @@ void WavOutFile::write(const short *buffer, int numElems)
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// use temp buffer to swap byte order if necessary
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short* pTemp = (short*)getConvBuffer(numElems * sizeof(short));
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memcpy(pTemp, buffer, numElems * 2);
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memcpy(pTemp, buffer, (size_t)numElems * 2L);
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_swap16Buffer(pTemp, numElems);
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res = (int)fwrite(pTemp, 2, numElems, fptr);
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|
@ -984,3 +980,5 @@ void WavOutFile::write(const float *buffer, int numElems)
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}
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bytesWritten += numBytes;
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}
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}
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|
|
|
@ -40,7 +40,12 @@
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#ifndef WAVFILE_H
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#define WAVFILE_H
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#include <stdio.h>
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#include <cstdio>
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#include <string>
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#include "SS_CharTypes.h"
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namespace soundstretch
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{
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#ifndef uint
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typedef unsigned int uint;
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|
@ -118,9 +123,6 @@ private:
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/// File pointer.
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FILE *fptr;
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/// Position within the audio stream
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long position;
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/// Counter of how many bytes of sample data have been read from the file.
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long dataRead;
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|
@ -148,7 +150,7 @@ private:
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public:
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/// Constructor: Opens the given WAV file. If the file can't be opened,
|
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/// throws 'runtime_error' exception.
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WavInFile(const char *filename);
|
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WavInFile(const STRING& filename);
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WavInFile(FILE *file);
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|
@ -244,7 +246,7 @@ private:
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public:
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/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
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/// if file creation fails.
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WavOutFile(const char *fileName, ///< Filename
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WavOutFile(const STRING& fileName, ///< Filename
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int sampleRate, ///< Sample rate (e.g. 44100 etc)
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int bits, ///< Bits per sample (8 or 16 bits)
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int channels ///< Number of channels (1=mono, 2=stereo)
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|
@ -274,4 +276,6 @@ public:
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);
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};
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|
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}
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#endif
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|
|
|
@ -54,7 +54,7 @@ using namespace soundtouch;
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static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
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{
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FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
|
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if (fptr == NULL) return;
|
||||
if (fptr == nullptr) return;
|
||||
|
||||
for (int i = 0; i < len; i ++)
|
||||
{
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|
|
|
@ -376,8 +376,6 @@ void BPMDetect::updateBeatPos(int process_samples)
|
|||
// detect beats
|
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for (int i = 0; i < skipstep; i++)
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{
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LONG_SAMPLETYPE max = 0;
|
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float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
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sum -= beat_lpf.update(sum);
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|
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|
@ -556,13 +554,13 @@ float BPMDetect::getBpm()
|
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/// - "values" receive array of beat detection strengths
|
||||
/// - max_num indicates max.size of "pos" and "values" array.
|
||||
///
|
||||
/// You can query a suitable array sized by calling this with NULL in "pos" & "values".
|
||||
/// You can query a suitable array sized by calling this with nullptr in "pos" & "values".
|
||||
///
|
||||
/// \return number of beats in the arrays.
|
||||
int BPMDetect::getBeats(float *pos, float *values, int max_num)
|
||||
{
|
||||
int num = (int)beats.size();
|
||||
if ((!pos) || (!values)) return num; // pos or values NULL, return just size
|
||||
if ((!pos) || (!values)) return num; // pos or values nullptr, return just size
|
||||
|
||||
for (int i = 0; (i < num) && (i < max_num); i++)
|
||||
{
|
||||
|
|
|
@ -50,8 +50,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
|
|||
{
|
||||
assert(numChannels > 0);
|
||||
sizeInBytes = 0; // reasonable initial value
|
||||
buffer = NULL;
|
||||
bufferUnaligned = NULL;
|
||||
buffer = nullptr;
|
||||
bufferUnaligned = nullptr;
|
||||
samplesInBuffer = 0;
|
||||
bufferPos = 0;
|
||||
channels = (uint)numChannels;
|
||||
|
@ -63,8 +63,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
|
|||
FIFOSampleBuffer::~FIFOSampleBuffer()
|
||||
{
|
||||
delete[] bufferUnaligned;
|
||||
bufferUnaligned = NULL;
|
||||
buffer = NULL;
|
||||
bufferUnaligned = nullptr;
|
||||
buffer = nullptr;
|
||||
}
|
||||
|
||||
|
||||
|
@ -166,7 +166,7 @@ void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
|
|||
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
|
||||
assert(sizeInBytes % 2 == 0);
|
||||
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
|
||||
if (tempUnaligned == NULL)
|
||||
if (tempUnaligned == nullptr)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
|
||||
}
|
||||
|
|
|
@ -59,8 +59,8 @@ FIRFilter::FIRFilter()
|
|||
resultDivider = 0;
|
||||
length = 0;
|
||||
lengthDiv8 = 0;
|
||||
filterCoeffs = NULL;
|
||||
filterCoeffsStereo = NULL;
|
||||
filterCoeffs = nullptr;
|
||||
filterCoeffsStereo = nullptr;
|
||||
}
|
||||
|
||||
|
||||
|
@ -75,15 +75,11 @@ FIRFilter::~FIRFilter()
|
|||
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
|
||||
{
|
||||
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
|
||||
// hint compiler autovectorization that loop length is divisible by 8
|
||||
int ilength = length & -8;
|
||||
uint ilength = length & -8;
|
||||
|
||||
assert((length != 0) && (length == ilength) && (src != NULL) && (dest != NULL) && (filterCoeffs != NULL));
|
||||
assert((length != 0) && (length == ilength) && (src != nullptr) && (dest != nullptr) && (filterCoeffs != nullptr));
|
||||
assert(numSamples > ilength);
|
||||
|
||||
end = 2 * (numSamples - ilength);
|
||||
|
||||
|
@ -96,7 +92,7 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
|
|||
suml = sumr = 0;
|
||||
ptr = src + j;
|
||||
|
||||
for (int i = 0; i < ilength; i ++)
|
||||
for (uint i = 0; i < ilength; i ++)
|
||||
{
|
||||
suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
|
||||
sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
|
||||
|
@ -121,11 +117,6 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
|
|||
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
|
||||
{
|
||||
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
|
||||
|
||||
// hint compiler autovectorization that loop length is divisible by 8
|
||||
int ilength = length & -8;
|
||||
|
@ -160,16 +151,10 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
|
|||
{
|
||||
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(src != nullptr);
|
||||
assert(dest != nullptr);
|
||||
assert(filterCoeffs != nullptr);
|
||||
assert(numChannels < 16);
|
||||
|
||||
// hint compiler autovectorization that loop length is divisible by 8
|
||||
|
@ -288,7 +273,7 @@ uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSample
|
|||
|
||||
// Operator 'new' is overloaded so that it automatically creates a suitable instance
|
||||
// depending on if we've a MMX-capable CPU available or not.
|
||||
void * FIRFilter::operator new(size_t s)
|
||||
void * FIRFilter::operator new(size_t)
|
||||
{
|
||||
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
|
||||
ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
|
||||
|
@ -301,6 +286,7 @@ FIRFilter * FIRFilter::newInstance()
|
|||
uint uExtensions;
|
||||
|
||||
uExtensions = detectCPUextensions();
|
||||
(void)uExtensions;
|
||||
|
||||
// Check if MMX/SSE instruction set extensions supported by CPU
|
||||
|
||||
|
|
|
@ -38,7 +38,7 @@
|
|||
namespace soundtouch
|
||||
{
|
||||
|
||||
class InterpolateCubic final : public TransposerBase
|
||||
class InterpolateCubic : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
|
@ -58,7 +58,7 @@ public:
|
|||
|
||||
virtual void resetRegisters() override;
|
||||
|
||||
int getLatency() const override
|
||||
virtual int getLatency() const override
|
||||
{
|
||||
return 1;
|
||||
}
|
||||
|
|
|
@ -39,7 +39,7 @@ namespace soundtouch
|
|||
{
|
||||
|
||||
/// Linear transposer class that uses integer arithmetic
|
||||
class InterpolateLinearInteger final : public TransposerBase
|
||||
class InterpolateLinearInteger : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
int iFract;
|
||||
|
@ -61,7 +61,7 @@ public:
|
|||
|
||||
virtual void resetRegisters() override;
|
||||
|
||||
int getLatency() const override
|
||||
virtual int getLatency() const override
|
||||
{
|
||||
return 0;
|
||||
}
|
||||
|
@ -69,25 +69,25 @@ public:
|
|||
|
||||
|
||||
/// Linear transposer class that uses floating point arithmetic
|
||||
class InterpolateLinearFloat final : public TransposerBase
|
||||
class InterpolateLinearFloat : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
double fract;
|
||||
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples) override;
|
||||
int &srcSamples);
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
int &srcSamples) override;
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples) override;
|
||||
int &srcSamples);
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
|
||||
|
||||
public:
|
||||
InterpolateLinearFloat();
|
||||
|
||||
void resetRegisters() override;
|
||||
virtual void resetRegisters();
|
||||
|
||||
int getLatency() const override
|
||||
int getLatency() const
|
||||
{
|
||||
return 0;
|
||||
}
|
||||
|
|
|
@ -171,9 +171,9 @@ int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
|
|||
|
||||
/// 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)
|
||||
int InterpolateShannon::transposeMulti(SAMPLETYPE *,
|
||||
const SAMPLETYPE *,
|
||||
int &)
|
||||
{
|
||||
// not implemented
|
||||
assert(false);
|
||||
|
|
|
@ -43,7 +43,7 @@
|
|||
namespace soundtouch
|
||||
{
|
||||
|
||||
class InterpolateShannon final : public TransposerBase
|
||||
class InterpolateShannon : public TransposerBase
|
||||
{
|
||||
protected:
|
||||
int transposeMono(SAMPLETYPE *dest,
|
||||
|
@ -63,7 +63,7 @@ public:
|
|||
|
||||
void resetRegisters() override;
|
||||
|
||||
int getLatency() const override
|
||||
virtual int getLatency() const override
|
||||
{
|
||||
return 3;
|
||||
}
|
||||
|
|
|
@ -131,8 +131,6 @@ void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
|
|||
// the 'set_returnBuffer_size' function.
|
||||
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint count;
|
||||
|
||||
if (nSamples == 0) return;
|
||||
|
||||
// Store samples to input buffer
|
||||
|
@ -142,7 +140,7 @@ void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
|
|||
// the filter
|
||||
if (bUseAAFilter == false)
|
||||
{
|
||||
count = pTransposer->transpose(outputBuffer, inputBuffer);
|
||||
(void)pTransposer->transpose(outputBuffer, inputBuffer);
|
||||
return;
|
||||
}
|
||||
|
||||
|
@ -309,7 +307,7 @@ TransposerBase *TransposerBase::newInstance()
|
|||
|
||||
default:
|
||||
assert(false);
|
||||
return NULL;
|
||||
return nullptr;
|
||||
}
|
||||
#endif
|
||||
}
|
||||
|
|
|
@ -413,15 +413,15 @@ int SoundTouch::getSetting(int settingId) const
|
|||
return (uint)pTDStretch->isQuickSeekEnabled();
|
||||
|
||||
case SETTING_SEQUENCE_MS:
|
||||
pTDStretch->getParameters(NULL, &temp, NULL, NULL);
|
||||
pTDStretch->getParameters(nullptr, &temp, nullptr, nullptr);
|
||||
return temp;
|
||||
|
||||
case SETTING_SEEKWINDOW_MS:
|
||||
pTDStretch->getParameters(NULL, NULL, &temp, NULL);
|
||||
pTDStretch->getParameters(nullptr, nullptr, &temp, nullptr);
|
||||
return temp;
|
||||
|
||||
case SETTING_OVERLAP_MS:
|
||||
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
|
||||
pTDStretch->getParameters(nullptr, nullptr, nullptr, &temp);
|
||||
return temp;
|
||||
|
||||
case SETTING_NOMINAL_INPUT_SEQUENCE :
|
||||
|
|
|
@ -54,25 +54,6 @@ using namespace soundtouch;
|
|||
|
||||
#define max(x, y) (((x) > (y)) ? (x) : (y))
|
||||
|
||||
/*****************************************************************************
|
||||
*
|
||||
* Constant definitions
|
||||
*
|
||||
*****************************************************************************/
|
||||
|
||||
// Table for the hierarchical mixing position seeking algorithm
|
||||
const short _scanOffsets[5][24]={
|
||||
{ 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
|
||||
868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
|
||||
{-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
|
||||
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
|
||||
{ -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0,
|
||||
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
|
||||
{ -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0,
|
||||
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
|
||||
{ 121, 114, 97, 114, 98, 105, 108, 32, 104, 99, 117, 111,
|
||||
116, 100, 110, 117, 111, 115, 0, 0, 0, 0, 0, 0}};
|
||||
|
||||
/*****************************************************************************
|
||||
*
|
||||
* Implementation of the class 'TDStretch'
|
||||
|
@ -85,8 +66,8 @@ TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
|
|||
bQuickSeek = false;
|
||||
channels = 2;
|
||||
|
||||
pMidBuffer = NULL;
|
||||
pMidBufferUnaligned = NULL;
|
||||
pMidBuffer = nullptr;
|
||||
pMidBufferUnaligned = nullptr;
|
||||
overlapLength = 0;
|
||||
|
||||
bAutoSeqSetting = true;
|
||||
|
@ -162,7 +143,7 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
|
|||
|
||||
|
||||
/// Get routine control parameters, see setParameters() function.
|
||||
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
|
||||
/// Any of the parameters to this function can be nullptr, in such case corresponding parameter
|
||||
/// value isn't returned.
|
||||
void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const
|
||||
{
|
||||
|
@ -759,7 +740,7 @@ void TDStretch::acceptNewOverlapLength(int newOverlapLength)
|
|||
|
||||
// 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 * TDStretch::operator new(size_t s)
|
||||
void * TDStretch::operator new(size_t)
|
||||
{
|
||||
// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
|
||||
ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
|
||||
|
@ -772,6 +753,7 @@ TDStretch * TDStretch::newInstance()
|
|||
uint uExtensions;
|
||||
|
||||
uExtensions = detectCPUextensions();
|
||||
(void)uExtensions;
|
||||
|
||||
// Check if MMX/SSE instruction set extensions supported by CPU
|
||||
|
||||
|
|
|
@ -217,7 +217,7 @@ public:
|
|||
);
|
||||
|
||||
/// Get routine control parameters, see setParameters() function.
|
||||
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
|
||||
/// Any of the parameters to this function can be nullptr, in such case corresponding parameter
|
||||
/// value isn't returned.
|
||||
void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
|
||||
|
||||
|
|
|
@ -294,8 +294,8 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
|
|||
|
||||
FIRFilterMMX::FIRFilterMMX() : FIRFilter()
|
||||
{
|
||||
filterCoeffsAlign = NULL;
|
||||
filterCoeffsUnalign = NULL;
|
||||
filterCoeffsAlign = nullptr;
|
||||
filterCoeffsUnalign = nullptr;
|
||||
}
|
||||
|
||||
|
||||
|
|
|
@ -195,16 +195,16 @@ double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2,
|
|||
|
||||
FIRFilterSSE::FIRFilterSSE() : FIRFilter()
|
||||
{
|
||||
filterCoeffsAlign = NULL;
|
||||
filterCoeffsUnalign = NULL;
|
||||
filterCoeffsAlign = nullptr;
|
||||
filterCoeffsUnalign = nullptr;
|
||||
}
|
||||
|
||||
|
||||
FIRFilterSSE::~FIRFilterSSE()
|
||||
{
|
||||
delete[] filterCoeffsUnalign;
|
||||
filterCoeffsAlign = NULL;
|
||||
filterCoeffsUnalign = NULL;
|
||||
filterCoeffsAlign = nullptr;
|
||||
filterCoeffsUnalign = nullptr;
|
||||
}
|
||||
|
||||
|
||||
|
@ -245,10 +245,10 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
|
|||
|
||||
if (count < 2) return 0;
|
||||
|
||||
assert(source != NULL);
|
||||
assert(dest != NULL);
|
||||
assert(source != nullptr);
|
||||
assert(dest != nullptr);
|
||||
assert((length % 8) == 0);
|
||||
assert(filterCoeffsAlign != NULL);
|
||||
assert(filterCoeffsAlign != nullptr);
|
||||
assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
|
||||
|
||||
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
|
||||
|
|
Loading…
Reference in New Issue