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3rdparty/soundtouch: Bump to v2.3.3

This commit is contained in:
JordanTheToaster 2024-08-06 03:23:12 +01:00 committed by lightningterror
parent 30e7de7555
commit ebc3923b35
32 changed files with 954 additions and 915 deletions
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39
3rdparty/soundtouch/README.html vendored
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@ -15,8 +15,8 @@
<body class="normal"> <body class="normal">
<hr> <hr>
<h1>SoundTouch audio processing library v2.3.1</h1> <h1>SoundTouch audio processing library v2.3.3</h1>
<p class="normal">SoundTouch library Copyright &copy; Olli Parviainen 2001-2021</p> <p class="normal">SoundTouch library Copyright &copy; Olli Parviainen 2001-2024</p>
<hr> <hr>
<h2>1. Introduction </h2> <h2>1. Introduction </h2>
<p>SoundTouch is an open-source audio processing library that allows <p>SoundTouch is an open-source audio processing library that allows
@ -35,7 +35,7 @@
<p>Author email: oparviai 'at' iki.fi </p> <p>Author email: oparviai 'at' iki.fi </p>
<p>SoundTouch WWW page: <a href="http://soundtouch.surina.net">http://soundtouch.surina.net</a></p> <p>SoundTouch WWW page: <a href="http://soundtouch.surina.net">http://soundtouch.surina.net</a></p>
<p>SoundTouch git repository: <a <p>SoundTouch git repository: <a
href="https://gitlab.com/soundtouch/soundtouch.git">https://gitlab.com/soundtouch/soundtouch.git</a></p> href="https://codeberg.org/soundtouch/soundtouch.git">https://codeberg.org/soundtouch/soundtouch.git</a></p>
<hr> <hr>
<h2>2. Compiling SoundTouch</h2> <h2>2. Compiling SoundTouch</h2>
<p>Before compiling, notice that you can choose the sample data format if it's <p>Before compiling, notice that you can choose the sample data format if it's
@ -131,10 +131,12 @@
</table> </table>
<b>Compiling portable Shared Library / DLL version</b> <b>Compiling portable Shared Library / DLL version</b>
<p> The GNU autotools compilation does not automatically create a shared-library version of <p> The GNU autotools compilation automatically builds an additional dynamic-link version
SoundTouch (.so or .dll) that features position-independent code and C-language of SoundTouch library that features position-independent code and "C"-style API that is
api that are more suitable for cross-language development than C++ libraries.</p> more suitable for calling the SoundTouch routines from other programming languages.</p>
<p> Use script "make-gnu-dll-sh" to build a portable dynamic library version if such is desired.</p> <p>This dynamic-link library is built under source/SoundTouchDLL directory, whose
subdirectories also comtain simple example apps that use the dynamic-link library.
</p>
<h4><b>2.2.2 Compiling with cmake</b></h4> <h4><b>2.2.2 Compiling with cmake</b></h4>
<p>'cmake' build scripts are provided as an alternative to the autotools toolchain.</p> <p>'cmake' build scripts are provided as an alternative to the autotools toolchain.</p>
@ -145,6 +147,9 @@
cmake . cmake .
make -j make -j
make install</pre> make install</pre>
<p>To list available build options:</p>
<pre>
cmake -LH</pre>
<p>To compile the additional portable Shared Library / DLL version with the native C-language API:</p> <p>To compile the additional portable Shared Library / DLL version with the native C-language API:</p>
<pre> <pre>
cmake . -DSOUNDTOUCH_DLL=ON cmake . -DSOUNDTOUCH_DLL=ON
@ -448,7 +453,7 @@
<h2><a name="SoundStretch"></a>4. SoundStretch audio processing utility <h2><a name="SoundStretch"></a>4. SoundStretch audio processing utility
</h2> </h2>
<p>SoundStretch audio processing utility<br> <p>SoundStretch audio processing utility<br>
Copyright (c) Olli Parviainen 2002-2015</p> Copyright (c) Olli Parviainen 2002-2024</p>
<p>SoundStretch is a simple command-line application that can change <p>SoundStretch is a simple command-line application that can change
tempo, pitch and playback rates of WAV sound files. This program is tempo, pitch and playback rates of WAV sound files. This program is
intended primarily to demonstrate how the "SoundTouch" library can be intended primarily to demonstrate how the "SoundTouch" library can be
@ -603,6 +608,18 @@
<hr> <hr>
<h2>5. Change History</h2> <h2>5. Change History</h2>
<h3>5.1. SoundTouch library Change History </h3> <h3>5.1. SoundTouch library Change History </h3>
<p><b>2.3.3:</b></p>
<ul class="current">
<li>Fixing compiler warnings, maintenance fixes to make/build files for various systems
</li>
</ul>
<p><b>2.3.2:</b></p>
<ul>
<li>Improve autotools makefiles to build the `SoundTouchDLL` dynamic-link link library with
C-style API. This library variation is easier to import and use from other programming
languages than the default C++ library.
</li>
</ul>
<p><b>2.3.1:</b></p> <p><b>2.3.1:</b></p>
<ul> <ul>
<li>Adjusted cmake build settings and header files that cmake installs</li> <li>Adjusted cmake build settings and header files that cmake installs</li>
@ -865,11 +882,14 @@
<li> Initial release</li> <li> Initial release</li>
</ul> </ul>
<h3>5.2. SoundStretch application Change History </h3> <h3>5.2. SoundStretch application Change History </h3>
<p><b>2.3.3:</b></p>
<ul class="current_soundstretch">
<li>Added support for Asian / non-latin filenames in Windows. Gnu platform has supported them already earlier.</li>
</ul>
<p><b>1.9:</b></p> <p><b>1.9:</b></p>
<ul> <ul>
<li>Added support for WAV file 'fact' information chunk.</li> <li>Added support for WAV file 'fact' information chunk.</li>
</ul> </ul>
<p><b>1.7.0:</b></p> <p><b>1.7.0:</b></p>
<ul> <ul>
<li>Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer <li>Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer
@ -966,6 +986,7 @@
<li> Michael Pruett</li> <li> Michael Pruett</li>
<li> Rajeev Puran</li> <li> Rajeev Puran</li>
<li> RJ Ryan</li> <li> RJ Ryan</li>
<li> Serge Sans Paille</li>
<li> John Sheehy</li> <li> John Sheehy</li>
<li> Tim Shuttleworth</li> <li> Tim Shuttleworth</li>
<li> Albert Sirvent</li> <li> Albert Sirvent</li>

2
3rdparty/soundtouch/soundtouch/BPMDetect.h vendored
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@ -196,7 +196,7 @@ namespace soundtouch
/// - "values" receive array of beat detection strengths /// - "values" receive array of beat detection strengths
/// - max_num indicates max.size of "pos" and "values" array. /// - 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. /// \return number of beats in the arrays.
int getBeats(float *pos, float *strength, int max_num); int getBeats(float *pos, float *strength, int max_num);

10
3rdparty/soundtouch/soundtouch/FIFOSamplePipe.h vendored
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@ -88,11 +88,11 @@ public:
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data. void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
) )
{ {
int oNumSamples = other.numSamples(); const uint oNumSamples = other.numSamples();
putSamples(other.ptrBegin(), oNumSamples); putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples); other.receiveSamples(oNumSamples);
}; }
/// Output samples from beginning of the sample buffer. Copies requested samples to /// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than /// output buffer and removes them from the sample buffer. If there are less than
@ -144,8 +144,8 @@ protected:
/// Sets output pipe. /// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput) void setOutPipe(FIFOSamplePipe *pOutput)
{ {
assert(output == NULL); assert(output == nullptr);
assert(pOutput != NULL); assert(pOutput != nullptr);
output = pOutput; output = pOutput;
} }
@ -153,7 +153,7 @@ protected:
/// 'setOutPipe' function. /// 'setOutPipe' function.
FIFOProcessor() FIFOProcessor()
{ {
output = NULL; output = nullptr;
} }
/// Constructor. Configures output pipe. /// Constructor. Configures output pipe.

4
3rdparty/soundtouch/soundtouch/SoundTouch.h vendored
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@ -72,10 +72,10 @@ namespace soundtouch
{ {
/// Soundtouch library version string /// Soundtouch library version string
#define SOUNDTOUCH_VERSION "2.3.1" #define SOUNDTOUCH_VERSION "2.3.3"
/// SoundTouch library version id /// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (20301) #define SOUNDTOUCH_VERSION_ID (20303)
// //
// Available setting IDs for the 'setSetting' & 'get_setting' functions: // Available setting IDs for the 'setSetting' & 'get_setting' functions:

52
3rdparty/soundtouch/source/SoundStretch/SS_CharTypes.h vendored Normal file
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@ -0,0 +1,52 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Char type for SoundStretch
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// 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 SS_CHARTYPE_H
#define SS_CHARTYPE_H
#include <string>
namespace soundstretch
{
#if _WIN32
// wide-char types for supporting non-latin file paths in Windows
using CHARTYPE = wchar_t;
using STRING = std::wstring;
#define STRING_CONST(x) (L"" x)
#else
// gnu platform can natively support UTF-8 paths using "char*" set
using CHARTYPE = char;
using STRING = std::string;
#define STRING_CONST(x) (x)
#endif
}
#endif //SS_CHARTYPE_H

534
3rdparty/soundtouch/source/SoundStretch/WavFile.cpp vendored
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@ -42,17 +42,26 @@
#include <string> #include <string>
#include <sstream> #include <sstream>
#include <cstring> #include <cstring>
#include <assert.h> #include <cassert>
#include <limits.h> #include <climits>
#include "WavFile.h" #include "WavFile.h"
#include "STTypes.h" #include "STTypes.h"
using namespace std; using namespace std;
namespace soundstretch
{
#if _WIN32
#define FOPEN(name, mode) _wfopen(name, STRING_CONST(mode))
#else
#define FOPEN(name, mode) fopen(name, mode)
#endif
static const char riffStr[] = "RIFF"; static const char riffStr[] = "RIFF";
static const char waveStr[] = "WAVE"; static const char waveStr[] = "WAVE";
static const char fmtStr[] = "fmt "; static const char fmtStr[] = "fmt ";
static const char factStr[] = "fact"; static const char factStr[] = "fact";
static const char dataStr[] = "data"; static const char dataStr[] = "data";
@ -66,67 +75,67 @@ static const char dataStr[] = "data";
// while PowerPC of Mac's and many other RISC cpu's are big-endian. // while PowerPC of Mac's and many other RISC cpu's are big-endian.
#ifdef BYTE_ORDER #ifdef BYTE_ORDER
// In gcc compiler detect the byte order automatically // In gcc compiler detect the byte order automatically
#if BYTE_ORDER == BIG_ENDIAN #if BYTE_ORDER == BIG_ENDIAN
// big-endian platform. // big-endian platform.
#define _BIG_ENDIAN_ #define _BIG_ENDIAN_
#endif #endif
#endif #endif
#ifdef _BIG_ENDIAN_ #ifdef _BIG_ENDIAN_
// big-endian CPU, swap bytes in 16 & 32 bit words // big-endian CPU, swap bytes in 16 & 32 bit words
// helper-function to swap byte-order of 32bit integer // helper-function to swap byte-order of 32bit integer
static inline int _swap32(int &dwData) 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++)
{ {
dwData = ((dwData >> 24) & 0x000000FF) | pData[i] = _swap16(pData[i]);
((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 #else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such // little-endian CPU, WAV file is ok as such
// dummy helper-function // dummy helper-function
static inline int _swap32(int &dwData) static inline int _swap32(int& dwData)
{ {
// do nothing // do nothing
return dwData; return dwData;
} }
// dummy helper-function // dummy helper-function
static inline short _swap16(short &wData) static inline short _swap16(short& wData)
{ {
// do nothing // do nothing
return wData; return wData;
} }
// dummy helper-function // dummy helper-function
static inline void _swap16Buffer(short *pData, int numBytes) static inline void _swap16Buffer(short*, int)
{ {
// do nothing // do nothing
} }
#endif // BIG_ENDIAN #endif // BIG_ENDIAN
@ -138,7 +147,7 @@ static const char dataStr[] = "data";
WavFileBase::WavFileBase() WavFileBase::WavFileBase()
{ {
convBuff = NULL; convBuff = nullptr;
convBuffSize = 0; convBuffSize = 0;
} }
@ -151,7 +160,7 @@ WavFileBase::~WavFileBase()
/// Get pointer to conversion buffer of at min. given size /// Get pointer to conversion buffer of at min. given size
void *WavFileBase::getConvBuffer(int sizeBytes) void* WavFileBase::getConvBuffer(int sizeBytes)
{ {
if (convBuffSize < sizeBytes) if (convBuffSize < sizeBytes)
{ {
@ -169,32 +178,26 @@ void *WavFileBase::getConvBuffer(int sizeBytes)
// Class WavInFile // Class WavInFile
// //
WavInFile::WavInFile(const char *fileName) WavInFile::WavInFile(const STRING& fileName)
{ {
// Try to open the file for reading // Try to open the file for reading
fptr = fopen(fileName, "rb"); fptr = FOPEN(fileName.c_str(), "rb");
if (fptr == NULL) if (fptr == nullptr)
{ {
// didn't succeed ST_THROW_RT_ERROR("Error : Unable to open file for reading.");
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for reading.";
ST_THROW_RT_ERROR(msg.c_str());
} }
init(); init();
} }
WavInFile::WavInFile(FILE *file) WavInFile::WavInFile(FILE* file)
{ {
// Try to open the file for reading // Try to open the file for reading
fptr = file; fptr = file;
if (!file) if (!file)
{ {
// didn't succeed ST_THROW_RT_ERROR("Error : Unable to access input stream for reading");
string msg = "Error : Unable to access input stream for reading";
ST_THROW_RT_ERROR(msg.c_str());
} }
init(); init();
@ -213,17 +216,15 @@ void WavInFile::init()
hdrsOk = readWavHeaders(); hdrsOk = readWavHeaders();
if (hdrsOk != 0) if (hdrsOk != 0)
{ {
// Something didn't match in the wav file headers
ST_THROW_RT_ERROR("Input file is corrupt or not a WAV file"); ST_THROW_RT_ERROR("Input file is corrupt or not a WAV file");
} }
// sanity check for format parameters // sanity check for format parameters
if ((header.format.channel_number < 1) || (header.format.channel_number > 9) || if ((header.format.channel_number < 1) || (header.format.channel_number > 9) ||
(header.format.sample_rate < 4000) || (header.format.sample_rate > 192000) || (header.format.sample_rate < 4000) || (header.format.sample_rate > 192000) ||
(header.format.byte_per_sample < 1) || (header.format.byte_per_sample > 320) || (header.format.byte_per_sample < 1) || (header.format.byte_per_sample > 320) ||
(header.format.bits_per_sample < 8) || (header.format.bits_per_sample > 32)) (header.format.bits_per_sample < 8) || (header.format.bits_per_sample > 32))
{ {
// Something didn't match in the wav file headers
ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters."); ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters.");
} }
@ -234,7 +235,7 @@ void WavInFile::init()
WavInFile::~WavInFile() WavInFile::~WavInFile()
{ {
if (fptr) fclose(fptr); if (fptr) fclose(fptr);
fptr = NULL; fptr = nullptr;
} }
@ -260,7 +261,7 @@ int WavInFile::checkCharTags() const
} }
int WavInFile::read(unsigned char *buffer, int maxElems) int WavInFile::read(unsigned char* buffer, int maxElems)
{ {
int numBytes; int numBytes;
uint afterDataRead; uint afterDataRead;
@ -289,7 +290,7 @@ int WavInFile::read(unsigned char *buffer, int maxElems)
} }
int WavInFile::read(short *buffer, int maxElems) int WavInFile::read(short* buffer, int maxElems)
{ {
unsigned int afterDataRead; unsigned int afterDataRead;
int numBytes; int numBytes;
@ -298,53 +299,53 @@ int WavInFile::read(short *buffer, int maxElems)
assert(buffer); assert(buffer);
switch (header.format.bits_per_sample) switch (header.format.bits_per_sample)
{ {
case 8: case 8:
{ {
// 8 bit format // 8 bit format
unsigned char *temp = (unsigned char*)getConvBuffer(maxElems); unsigned char* temp = (unsigned char*)getConvBuffer(maxElems);
int i; int i;
numElems = read(temp, maxElems); numElems = read(temp, maxElems);
// convert from 8 to 16 bit // convert from 8 to 16 bit
for (i = 0; i < numElems; i ++) for (i = 0; i < numElems; i++)
{ {
buffer[i] = (short)(((short)temp[i] - 128) * 256); buffer[i] = (short)(((short)temp[i] - 128) * 256);
} }
break; 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);
} }
case 16: numBytes = (int)fread(buffer, 1, numBytes, fptr);
{ dataRead += numBytes;
// 16 bit format numElems = numBytes / 2;
assert(sizeof(short) == 2); // 16bit samples, swap byte order if necessary
_swap16Buffer((short*)buffer, numElems);
break;
}
numBytes = maxElems * 2; default:
afterDataRead = dataRead + numBytes; {
if (afterDataRead > header.data.data_len) stringstream ss;
{ ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with ";
// Don't read more samples than are marked available in header ss << (int)header.format.bits_per_sample;
numBytes = (int)header.data.data_len - (int)dataRead; ss << " bit sample format. ";
assert(numBytes >= 0); ST_THROW_RT_ERROR(ss.str().c_str());
} }
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; return numElems;
@ -353,7 +354,7 @@ int WavInFile::read(short *buffer, int maxElems)
/// Read data in float format. Notice that when reading in float format /// Read data in float format. Notice that when reading in float format
/// 8/16/24/32 bit sample formats are supported /// 8/16/24/32 bit sample formats are supported
int WavInFile::read(float *buffer, int maxElems) int WavInFile::read(float* buffer, int maxElems)
{ {
unsigned int afterDataRead; unsigned int afterDataRead;
int numBytes; int numBytes;
@ -382,7 +383,7 @@ int WavInFile::read(float *buffer, int maxElems)
} }
// read raw data into temporary buffer // read raw data into temporary buffer
char *temp = (char*)getConvBuffer(numBytes); char* temp = (char*)getConvBuffer(numBytes);
numBytes = (int)fread(temp, 1, numBytes, fptr); numBytes = (int)fread(temp, 1, numBytes, fptr);
dataRead += numBytes; dataRead += numBytes;
@ -391,56 +392,56 @@ int WavInFile::read(float *buffer, int maxElems)
// swap byte ordert & convert to float, depending on sample format // swap byte ordert & convert to float, depending on sample format
switch (bytesPerSample) switch (bytesPerSample)
{ {
case 1: case 1:
{
unsigned char* temp2 = (unsigned char*)temp;
double conv = 1.0 / 128.0;
for (int i = 0; i < numElems; i++)
{ {
unsigned char *temp2 = (unsigned char*)temp; buffer[i] = (float)(temp2[i] * conv - 1.0);
double conv = 1.0 / 128.0;
for (int i = 0; i < numElems; i ++)
{
buffer[i] = (float)(temp2[i] * conv - 1.0);
}
break;
} }
break;
}
case 2: case 2:
{
short* temp2 = (short*)temp;
double conv = 1.0 / 32768.0;
for (int i = 0; i < numElems; i++)
{ {
short *temp2 = (short*)temp; short value = temp2[i];
double conv = 1.0 / 32768.0; buffer[i] = (float)(_swap16(value) * conv);
for (int i = 0; i < numElems; i ++)
{
short value = temp2[i];
buffer[i] = (float)(_swap16(value) * conv);
}
break;
} }
break;
}
case 3: case 3:
{
char* temp2 = (char*)temp;
double conv = 1.0 / 8388608.0;
for (int i = 0; i < numElems; i++)
{ {
char *temp2 = (char *)temp; int value = *((int*)temp2);
double conv = 1.0 / 8388608.0; value = _swap32(value) & 0x00ffffff; // take 24 bits
for (int i = 0; i < numElems; i ++) value |= (value & 0x00800000) ? 0xff000000 : 0; // extend minus sign bits
{ buffer[i] = (float)(value * conv);
int value = *((int*)temp2); temp2 += 3;
value = _swap32(value) & 0x00ffffff; // take 24 bits
value |= (value & 0x00800000) ? 0xff000000 : 0; // extend minus sign bits
buffer[i] = (float)(value * conv);
temp2 += 3;
}
break;
} }
break;
}
case 4: case 4:
{
int* temp2 = (int*)temp;
double conv = 1.0 / 2147483648.0;
assert(sizeof(int) == 4);
for (int i = 0; i < numElems; i++)
{ {
int *temp2 = (int *)temp; int value = temp2[i];
double conv = 1.0 / 2147483648.0; buffer[i] = (float)(_swap32(value) * conv);
assert(sizeof(int) == 4);
for (int i = 0; i < numElems; i ++)
{
int value = temp2[i];
buffer[i] = (float)(_swap32(value) * conv);
}
break;
} }
break;
}
} }
return numElems; return numElems;
@ -450,7 +451,7 @@ int WavInFile::read(float *buffer, int maxElems)
int WavInFile::eof() const int WavInFile::eof() const
{ {
// return true if all data has been read or file eof has reached // return true if all data has been read or file eof has reached
return (dataRead == header.data.data_len || feof(fptr)); return ((uint)dataRead == header.data.data_len || feof(fptr));
} }
@ -462,7 +463,7 @@ static int isAlpha(char c)
// test if all characters are between a white space ' ' and little 'z' // test if all characters are between a white space ' ' and little 'z'
static int isAlphaStr(const char *str) static int isAlphaStr(const char* str)
{ {
char c; char c;
@ -470,7 +471,7 @@ static int isAlphaStr(const char *str)
while (c) while (c)
{ {
if (isAlpha(c) == 0) return 0; if (isAlpha(c) == 0) return 0;
str ++; str++;
c = str[0]; c = str[0];
} }
@ -483,7 +484,7 @@ int WavInFile::readRIFFBlock()
if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1; if (fread(&(header.riff), sizeof(WavRiff), 1, fptr) != 1) return -1;
// swap 32bit data byte order if necessary // swap 32bit data byte order if necessary
_swap32((int &)header.riff.package_len); _swap32((int&)header.riff.package_len);
// header.riff.riff_char should equal to 'RIFF'); // header.riff.riff_char should equal to 'RIFF');
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1; if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
@ -500,7 +501,7 @@ int WavInFile::readHeaderBlock()
string sLabel; string sLabel;
// lead label string // lead label string
if (fread(label, 1, 4, fptr) !=4) return -1; if (fread(label, 1, 4, fptr) != 4) return -1;
label[4] = 0; label[4] = 0;
if (isAlphaStr(label) == 0) return -1; // not a valid label if (isAlphaStr(label) == 0) return -1; // not a valid label
@ -536,12 +537,12 @@ int WavInFile::readHeaderBlock()
if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1; if (fread(&(header.format.fixed), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary // swap byte order if necessary
_swap16((short &)header.format.fixed); // short int fixed; _swap16((short&)header.format.fixed); // short int fixed;
_swap16((short &)header.format.channel_number); // short int channel_number; _swap16((short&)header.format.channel_number); // short int channel_number;
_swap32((int &)header.format.sample_rate); // int sample_rate; _swap32((int&)header.format.sample_rate); // int sample_rate;
_swap32((int &)header.format.byte_rate); // int byte_rate; _swap32((int&)header.format.byte_rate); // int byte_rate;
_swap16((short &)header.format.byte_per_sample); // short int byte_per_sample; _swap16((short&)header.format.byte_per_sample); // short int byte_per_sample;
_swap16((short &)header.format.bits_per_sample); // short int bits_per_sample; _swap16((short&)header.format.bits_per_sample); // short int bits_per_sample;
// if format_len is larger than expected, skip the extra data // if format_len is larger than expected, skip the extra data
if (nDump > 0) if (nDump > 0)
@ -581,7 +582,7 @@ int WavInFile::readHeaderBlock()
if (fread(&(header.fact.fact_sample_len), nLen, 1, fptr) != 1) return -1; if (fread(&(header.fact.fact_sample_len), nLen, 1, fptr) != 1) return -1;
// swap byte order if necessary // swap byte order if necessary
_swap32((int &)header.fact.fact_sample_len); // int sample_length; _swap32((int&)header.fact.fact_sample_len); // int sample_length;
// if fact_len is larger than expected, skip the extra data // if fact_len is larger than expected, skip the extra data
if (nDump > 0) if (nDump > 0)
@ -598,7 +599,7 @@ int WavInFile::readHeaderBlock()
if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1; if (fread(&(header.data.data_len), sizeof(uint), 1, fptr) != 1) return -1;
// swap byte order if necessary // swap byte order if necessary
_swap32((int &)header.data.data_len); _swap32((int&)header.data.data_len);
return 1; return 1;
} }
@ -611,7 +612,7 @@ int WavInFile::readHeaderBlock()
// read length // read length
if (fread(&len, sizeof(len), 1, fptr) != 1) return -1; if (fread(&len, sizeof(len), 1, fptr) != 1) return -1;
// scan through the block // scan through the block
for (i = 0; i < len; i ++) for (i = 0; i < len; i++)
{ {
if (fread(&temp, 1, 1, fptr) != 1) return -1; if (fread(&temp, 1, 1, fptr) != 1) return -1;
if (feof(fptr)) return -1; // unexpected eof if (feof(fptr)) return -1; // unexpected eof
@ -703,17 +704,13 @@ uint WavInFile::getElapsedMS() const
// Class WavOutFile // Class WavOutFile
// //
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels) WavOutFile::WavOutFile(const STRING& fileName, int sampleRate, int bits, int channels)
{ {
bytesWritten = 0; bytesWritten = 0;
fptr = fopen(fileName, "wb"); fptr = FOPEN(fileName.c_str(), "wb");
if (fptr == NULL) if (fptr == nullptr)
{ {
string msg = "Error : Unable to open file \""; ST_THROW_RT_ERROR("Error : Unable to open file for writing.");
msg += fileName;
msg += "\" for writing.";
//pmsg = msg.c_str;
ST_THROW_RT_ERROR(msg.c_str());
} }
fillInHeader(sampleRate, bits, channels); fillInHeader(sampleRate, bits, channels);
@ -721,14 +718,13 @@ WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int chann
} }
WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels) WavOutFile::WavOutFile(FILE* file, int sampleRate, int bits, int channels)
{ {
bytesWritten = 0; bytesWritten = 0;
fptr = file; fptr = file;
if (fptr == NULL) if (fptr == nullptr)
{ {
string msg = "Error : Unable to access output file stream."; ST_THROW_RT_ERROR("Error : Unable to access output file stream.");
ST_THROW_RT_ERROR(msg.c_str());
} }
fillInHeader(sampleRate, bits, channels); fillInHeader(sampleRate, bits, channels);
@ -740,7 +736,7 @@ WavOutFile::~WavOutFile()
{ {
finishHeader(); finishHeader();
if (fptr) fclose(fptr); if (fptr) fclose(fptr);
fptr = NULL; fptr = nullptr;
} }
@ -801,17 +797,17 @@ void WavOutFile::writeHeader()
// swap byte order if necessary // swap byte order if necessary
hdrTemp = header; hdrTemp = header;
_swap32((int &)hdrTemp.riff.package_len); _swap32((int&)hdrTemp.riff.package_len);
_swap32((int &)hdrTemp.format.format_len); _swap32((int&)hdrTemp.format.format_len);
_swap16((short &)hdrTemp.format.fixed); _swap16((short&)hdrTemp.format.fixed);
_swap16((short &)hdrTemp.format.channel_number); _swap16((short&)hdrTemp.format.channel_number);
_swap32((int &)hdrTemp.format.sample_rate); _swap32((int&)hdrTemp.format.sample_rate);
_swap32((int &)hdrTemp.format.byte_rate); _swap32((int&)hdrTemp.format.byte_rate);
_swap16((short &)hdrTemp.format.byte_per_sample); _swap16((short&)hdrTemp.format.byte_per_sample);
_swap16((short &)hdrTemp.format.bits_per_sample); _swap16((short&)hdrTemp.format.bits_per_sample);
_swap32((int &)hdrTemp.data.data_len); _swap32((int&)hdrTemp.data.data_len);
_swap32((int &)hdrTemp.fact.fact_len); _swap32((int&)hdrTemp.fact.fact_len);
_swap32((int &)hdrTemp.fact.fact_sample_len); _swap32((int&)hdrTemp.fact.fact_sample_len);
// write the supplemented header in the beginning of the file // write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET); fseek(fptr, 0, SEEK_SET);
@ -826,7 +822,7 @@ void WavOutFile::writeHeader()
} }
void WavOutFile::write(const unsigned char *buffer, int numElems) void WavOutFile::write(const unsigned char* buffer, int numElems)
{ {
int res; int res;
@ -846,7 +842,7 @@ void WavOutFile::write(const unsigned char *buffer, int numElems)
} }
void WavOutFile::write(const short *buffer, int numElems) void WavOutFile::write(const short* buffer, int numElems)
{ {
int res; int res;
@ -855,47 +851,47 @@ void WavOutFile::write(const short *buffer, int numElems)
switch (header.format.bits_per_sample) switch (header.format.bits_per_sample)
{ {
case 8: case 8:
{
int i;
unsigned char* temp = (unsigned char*)getConvBuffer(numElems);
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i++)
{ {
int i; temp[i] = (unsigned char)(buffer[i] / 256 + 128);
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;
} }
// write in 8bit format
write(temp, numElems);
break;
}
case 16: case 16:
{
// 16bit format
// use temp buffer to swap byte order if necessary
short* pTemp = (short*)getConvBuffer(numElems * sizeof(short));
memcpy(pTemp, buffer, (size_t)numElems * 2L);
_swap16Buffer(pTemp, numElems);
res = (int)fwrite(pTemp, 2, numElems, fptr);
if (res != numElems)
{ {
// 16bit format ST_THROW_RT_ERROR("Error while writing to a wav file.");
// 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;
} }
bytesWritten += 2 * numElems;
break;
}
default: default:
{ {
stringstream ss; stringstream ss;
ss << "\nOnly 8/16 bit sample WAV files supported in integer compilation. Can't open WAV file with "; 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 << (int)header.format.bits_per_sample;
ss << " bit sample format. "; ss << " bit sample format. ";
ST_THROW_RT_ERROR(ss.str().c_str()); ST_THROW_RT_ERROR(ss.str().c_str());
} }
} }
} }
@ -915,7 +911,7 @@ inline int saturate(float fvalue, float minval, float maxval)
} }
void WavOutFile::write(const float *buffer, int numElems) void WavOutFile::write(const float* buffer, int numElems)
{ {
int numBytes; int numBytes;
int bytesPerSample; int bytesPerSample;
@ -924,56 +920,56 @@ void WavOutFile::write(const float *buffer, int numElems)
bytesPerSample = header.format.bits_per_sample / 8; bytesPerSample = header.format.bits_per_sample / 8;
numBytes = numElems * bytesPerSample; numBytes = numElems * bytesPerSample;
void *temp = getConvBuffer(numBytes + 7); // round bit up to avoid buffer overrun with 24bit-value assignment void* temp = getConvBuffer(numBytes + 7); // round bit up to avoid buffer overrun with 24bit-value assignment
switch (bytesPerSample) switch (bytesPerSample)
{ {
case 1: case 1:
{
unsigned char* temp2 = (unsigned char*)temp;
for (int i = 0; i < numElems; i++)
{ {
unsigned char *temp2 = (unsigned char *)temp; temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
for (int i = 0; i < numElems; i ++)
{
temp2[i] = (unsigned char)saturate(buffer[i] * 128.0f + 128.0f, 0.0f, 255.0f);
}
break;
} }
break;
}
case 2: case 2:
{
short* temp2 = (short*)temp;
for (int i = 0; i < numElems; i++)
{ {
short *temp2 = (short *)temp; short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
for (int i = 0; i < numElems; i ++) temp2[i] = _swap16(value);
{
short value = (short)saturate(buffer[i] * 32768.0f, -32768.0f, 32767.0f);
temp2[i] = _swap16(value);
}
break;
} }
break;
}
case 3: case 3:
{
char* temp2 = (char*)temp;
for (int i = 0; i < numElems; i++)
{ {
char *temp2 = (char *)temp; int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
for (int i = 0; i < numElems; i ++) *((int*)temp2) = _swap32(value);
{ temp2 += 3;
int value = saturate(buffer[i] * 8388608.0f, -8388608.0f, 8388607.0f);
*((int*)temp2) = _swap32(value);
temp2 += 3;
}
break;
} }
break;
}
case 4: case 4:
{
int* temp2 = (int*)temp;
for (int i = 0; i < numElems; i++)
{ {
int *temp2 = (int *)temp; int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
for (int i = 0; i < numElems; i ++) temp2[i] = _swap32(value);
{
int value = saturate(buffer[i] * 2147483648.0f, -2147483648.0f, 2147483647.0f);
temp2[i] = _swap32(value);
}
break;
} }
break;
}
default: default:
assert(false); assert(false);
} }
int res = (int)fwrite(temp, 1, numBytes, fptr); int res = (int)fwrite(temp, 1, numBytes, fptr);
@ -984,3 +980,5 @@ void WavOutFile::write(const float *buffer, int numElems)
} }
bytesWritten += numBytes; bytesWritten += numBytes;
} }
}

16
3rdparty/soundtouch/source/SoundStretch/WavFile.h vendored
View File

@ -40,7 +40,12 @@
#ifndef WAVFILE_H #ifndef WAVFILE_H
#define WAVFILE_H #define WAVFILE_H
#include <stdio.h> #include <cstdio>
#include <string>
#include "SS_CharTypes.h"
namespace soundstretch
{
#ifndef uint #ifndef uint
typedef unsigned int uint; typedef unsigned int uint;
@ -118,9 +123,6 @@ private:
/// File pointer. /// File pointer.
FILE *fptr; FILE *fptr;
/// Position within the audio stream
long position;
/// Counter of how many bytes of sample data have been read from the file. /// Counter of how many bytes of sample data have been read from the file.
long dataRead; long dataRead;
@ -148,7 +150,7 @@ private:
public: public:
/// Constructor: Opens the given WAV file. If the file can't be opened, /// Constructor: Opens the given WAV file. If the file can't be opened,
/// throws 'runtime_error' exception. /// throws 'runtime_error' exception.
WavInFile(const char *filename); WavInFile(const STRING& filename);
WavInFile(FILE *file); WavInFile(FILE *file);
@ -244,7 +246,7 @@ private:
public: public:
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception /// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
/// if file creation fails. /// if file creation fails.
WavOutFile(const char *fileName, ///< Filename WavOutFile(const STRING& fileName, ///< Filename
int sampleRate, ///< Sample rate (e.g. 44100 etc) int sampleRate, ///< Sample rate (e.g. 44100 etc)
int bits, ///< Bits per sample (8 or 16 bits) int bits, ///< Bits per sample (8 or 16 bits)
int channels ///< Number of channels (1=mono, 2=stereo) int channels ///< Number of channels (1=mono, 2=stereo)
@ -274,4 +276,6 @@ public:
); );
}; };
}
#endif #endif

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

@ -54,7 +54,7 @@ using namespace soundtouch;
static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len) static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
{ {
FILE *fptr = fopen("aa_filter_coeffs.txt", "wt"); FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
if (fptr == NULL) return; if (fptr == nullptr) return;
for (int i = 0; i < len; i ++) for (int i = 0; i < len; i ++)
{ {

6
3rdparty/soundtouch/source/SoundTouch/BPMDetect.cpp vendored
View File

@ -376,8 +376,6 @@ void BPMDetect::updateBeatPos(int process_samples)
// detect beats // detect beats
for (int i = 0; i < skipstep; i++) for (int i = 0; i < skipstep; i++)
{ {
LONG_SAMPLETYPE max = 0;
float sum = beatcorr_ringbuff[beatcorr_ringbuffpos]; float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
sum -= beat_lpf.update(sum); sum -= beat_lpf.update(sum);
@ -556,13 +554,13 @@ float BPMDetect::getBpm()
/// - "values" receive array of beat detection strengths /// - "values" receive array of beat detection strengths
/// - max_num indicates max.size of "pos" and "values" array. /// - 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. /// \return number of beats in the arrays.
int BPMDetect::getBeats(float *pos, float *values, int max_num) int BPMDetect::getBeats(float *pos, float *values, int max_num)
{ {
int num = (int)beats.size(); 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++) for (int i = 0; (i < num) && (i < max_num); i++)
{ {

10
3rdparty/soundtouch/source/SoundTouch/FIFOSampleBuffer.cpp vendored
View File

@ -50,8 +50,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
{ {
assert(numChannels > 0); assert(numChannels > 0);
sizeInBytes = 0; // reasonable initial value sizeInBytes = 0; // reasonable initial value
buffer = NULL; buffer = nullptr;
bufferUnaligned = NULL; bufferUnaligned = nullptr;
samplesInBuffer = 0; samplesInBuffer = 0;
bufferPos = 0; bufferPos = 0;
channels = (uint)numChannels; channels = (uint)numChannels;
@ -63,8 +63,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
FIFOSampleBuffer::~FIFOSampleBuffer() FIFOSampleBuffer::~FIFOSampleBuffer()
{ {
delete[] bufferUnaligned; delete[] bufferUnaligned;
bufferUnaligned = NULL; bufferUnaligned = nullptr;
buffer = NULL; buffer = nullptr;
} }
@ -166,7 +166,7 @@ void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096; sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
assert(sizeInBytes % 2 == 0); assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)]; tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL) if (tempUnaligned == nullptr)
{ {
ST_THROW_RT_ERROR("Couldn't allocate memory!\n"); ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
} }

36
3rdparty/soundtouch/source/SoundTouch/FIRFilter.cpp vendored
View File

@ -59,8 +59,8 @@ FIRFilter::FIRFilter()
resultDivider = 0; resultDivider = 0;
length = 0; length = 0;
lengthDiv8 = 0; lengthDiv8 = 0;
filterCoeffs = NULL; filterCoeffs = nullptr;
filterCoeffsStereo = NULL; filterCoeffsStereo = nullptr;
} }
@ -75,15 +75,11 @@ FIRFilter::~FIRFilter()
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{ {
int j, end; 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 // 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); end = 2 * (numSamples - ilength);
@ -96,7 +92,7 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
suml = sumr = 0; suml = sumr = 0;
ptr = src + j; ptr = src + j;
for (int i = 0; i < ilength; i ++) for (uint i = 0; i < ilength; i ++)
{ {
suml += ptr[2 * i] * filterCoeffsStereo[2 * i]; suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1]; 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 uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{ {
int j, end; 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 // hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8; int ilength = length & -8;
@ -160,16 +151,10 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
{ {
int j, end; 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(length != 0);
assert(src != NULL); assert(src != nullptr);
assert(dest != NULL); assert(dest != nullptr);
assert(filterCoeffs != NULL); assert(filterCoeffs != nullptr);
assert(numChannels < 16); assert(numChannels < 16);
// hint compiler autovectorization that loop length is divisible by 8 // 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 // Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not. // 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! // 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!"); 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; uint uExtensions;
uExtensions = detectCPUextensions(); uExtensions = detectCPUextensions();
(void)uExtensions;
// Check if MMX/SSE instruction set extensions supported by CPU // Check if MMX/SSE instruction set extensions supported by CPU

4
3rdparty/soundtouch/source/SoundTouch/InterpolateCubic.h vendored
View File

@ -38,7 +38,7 @@
namespace soundtouch namespace soundtouch
{ {
class InterpolateCubic final : public TransposerBase class InterpolateCubic : public TransposerBase
{ {
protected: protected:
virtual int transposeMono(SAMPLETYPE *dest, virtual int transposeMono(SAMPLETYPE *dest,
@ -58,7 +58,7 @@ public:
virtual void resetRegisters() override; virtual void resetRegisters() override;
int getLatency() const override virtual int getLatency() const override
{ {
return 1; return 1;
} }

16
3rdparty/soundtouch/source/SoundTouch/InterpolateLinear.h vendored
View File

@ -39,7 +39,7 @@ namespace soundtouch
{ {
/// Linear transposer class that uses integer arithmetic /// Linear transposer class that uses integer arithmetic
class InterpolateLinearInteger final : public TransposerBase class InterpolateLinearInteger : public TransposerBase
{ {
protected: protected:
int iFract; int iFract;
@ -61,7 +61,7 @@ public:
virtual void resetRegisters() override; virtual void resetRegisters() override;
int getLatency() const override virtual int getLatency() const override
{ {
return 0; return 0;
} }
@ -69,25 +69,25 @@ public:
/// Linear transposer class that uses floating point arithmetic /// Linear transposer class that uses floating point arithmetic
class InterpolateLinearFloat final : public TransposerBase class InterpolateLinearFloat : public TransposerBase
{ {
protected: protected:
double fract; double fract;
virtual int transposeMono(SAMPLETYPE *dest, virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples) override; int &srcSamples);
virtual int transposeStereo(SAMPLETYPE *dest, virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples) override; int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples) override; virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
public: public:
InterpolateLinearFloat(); InterpolateLinearFloat();
void resetRegisters() override; virtual void resetRegisters();
int getLatency() const override int getLatency() const
{ {
return 0; return 0;
} }

6
3rdparty/soundtouch/source/SoundTouch/InterpolateShannon.cpp vendored
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@ -171,9 +171,9 @@ int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
/// Transpose stereo audio. Returns number of produced output samples, and /// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples /// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeMulti(SAMPLETYPE *pdest, int InterpolateShannon::transposeMulti(SAMPLETYPE *,
const SAMPLETYPE *psrc, const SAMPLETYPE *,
int &srcSamples) int &)
{ {
// not implemented // not implemented
assert(false); assert(false);

4
3rdparty/soundtouch/source/SoundTouch/InterpolateShannon.h vendored
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@ -43,7 +43,7 @@
namespace soundtouch namespace soundtouch
{ {
class InterpolateShannon final : public TransposerBase class InterpolateShannon : public TransposerBase
{ {
protected: protected:
int transposeMono(SAMPLETYPE *dest, int transposeMono(SAMPLETYPE *dest,
@ -63,7 +63,7 @@ public:
void resetRegisters() override; void resetRegisters() override;
int getLatency() const override virtual int getLatency() const override
{ {
return 3; return 3;
} }

6
3rdparty/soundtouch/source/SoundTouch/RateTransposer.cpp vendored
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@ -131,8 +131,6 @@ void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
// the 'set_returnBuffer_size' function. // the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples) void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{ {
uint count;
if (nSamples == 0) return; if (nSamples == 0) return;
// Store samples to input buffer // Store samples to input buffer
@ -142,7 +140,7 @@ void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
// the filter // the filter
if (bUseAAFilter == false) if (bUseAAFilter == false)
{ {
count = pTransposer->transpose(outputBuffer, inputBuffer); (void)pTransposer->transpose(outputBuffer, inputBuffer);
return; return;
} }
@ -309,7 +307,7 @@ TransposerBase *TransposerBase::newInstance()
default: default:
assert(false); assert(false);
return NULL; return nullptr;
} }
#endif #endif
} }

6
3rdparty/soundtouch/source/SoundTouch/SoundTouch.cpp vendored
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@ -413,15 +413,15 @@ int SoundTouch::getSetting(int settingId) const
return (uint)pTDStretch->isQuickSeekEnabled(); return (uint)pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS: case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL); pTDStretch->getParameters(nullptr, &temp, nullptr, nullptr);
return temp; return temp;
case SETTING_SEEKWINDOW_MS: case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(NULL, NULL, &temp, NULL); pTDStretch->getParameters(nullptr, nullptr, &temp, nullptr);
return temp; return temp;
case SETTING_OVERLAP_MS: case SETTING_OVERLAP_MS:
pTDStretch->getParameters(NULL, NULL, NULL, &temp); pTDStretch->getParameters(nullptr, nullptr, nullptr, &temp);
return temp; return temp;
case SETTING_NOMINAL_INPUT_SEQUENCE : case SETTING_NOMINAL_INPUT_SEQUENCE :

28
3rdparty/soundtouch/source/SoundTouch/TDStretch.cpp vendored
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@ -54,25 +54,6 @@ using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y)) #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' * Implementation of the class 'TDStretch'
@ -85,8 +66,8 @@ TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
bQuickSeek = false; bQuickSeek = false;
channels = 2; channels = 2;
pMidBuffer = NULL; pMidBuffer = nullptr;
pMidBufferUnaligned = NULL; pMidBufferUnaligned = nullptr;
overlapLength = 0; overlapLength = 0;
bAutoSeqSetting = true; bAutoSeqSetting = true;
@ -162,7 +143,7 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
/// Get routine control parameters, see setParameters() function. /// 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. /// value isn't returned.
void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const 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 // 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. // 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! // 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!"); 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; uint uExtensions;
uExtensions = detectCPUextensions(); uExtensions = detectCPUextensions();
(void)uExtensions;
// Check if MMX/SSE instruction set extensions supported by CPU // Check if MMX/SSE instruction set extensions supported by CPU

2
3rdparty/soundtouch/source/SoundTouch/TDStretch.h vendored
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@ -217,7 +217,7 @@ public:
); );
/// Get routine control parameters, see setParameters() function. /// 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. /// value isn't returned.
void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const; void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;

4
3rdparty/soundtouch/source/SoundTouch/mmx_optimized.cpp vendored
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@ -294,8 +294,8 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
FIRFilterMMX::FIRFilterMMX() : FIRFilter() FIRFilterMMX::FIRFilterMMX() : FIRFilter()
{ {
filterCoeffsAlign = NULL; filterCoeffsAlign = nullptr;
filterCoeffsUnalign = NULL; filterCoeffsUnalign = nullptr;
} }

14
3rdparty/soundtouch/source/SoundTouch/sse_optimized.cpp vendored
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@ -195,16 +195,16 @@ double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2,
FIRFilterSSE::FIRFilterSSE() : FIRFilter() FIRFilterSSE::FIRFilterSSE() : FIRFilter()
{ {
filterCoeffsAlign = NULL; filterCoeffsAlign = nullptr;
filterCoeffsUnalign = NULL; filterCoeffsUnalign = nullptr;
} }
FIRFilterSSE::~FIRFilterSSE() FIRFilterSSE::~FIRFilterSSE()
{ {
delete[] filterCoeffsUnalign; delete[] filterCoeffsUnalign;
filterCoeffsAlign = NULL; filterCoeffsAlign = nullptr;
filterCoeffsUnalign = NULL; filterCoeffsUnalign = nullptr;
} }
@ -245,10 +245,10 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
if (count < 2) return 0; if (count < 2) return 0;
assert(source != NULL); assert(source != nullptr);
assert(dest != NULL); assert(dest != nullptr);
assert((length % 8) == 0); assert((length % 8) == 0);
assert(filterCoeffsAlign != NULL); assert(filterCoeffsAlign != nullptr);
assert(((ulongptr)filterCoeffsAlign) % 16 == 0); assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2' // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'