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Removed commented code, style cleanup

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oparviainen 2018-08-12 20:51:24 +03:00
parent cca9271e98
commit c38f0506da
17 changed files with 15 additions and 191 deletions
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23
README.html
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@ -14,7 +14,7 @@
<body class="normal"> <body class="normal">
<hr> <hr>
<h1>SoundTouch audio processing library v2.1pre</h1> <h1>SoundTouch audio processing library v2.1pre</h1>
<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2018</p> <p class="normal">SoundTouch library Copyright <EFBFBD> Olli Parviainen 2001-2018</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
@ -575,16 +575,17 @@ this corresponds to lowering the pitch by -0.318 semitones:</p>
<h3>5.1. SoundTouch library Change History </h3> <h3>5.1. SoundTouch library Change History </h3>
<p><b>2.1pre:</b></p> <p><b>2.1pre:</b></p>
<ul> <ul>
<li>Refactored C# interface example</li> <li>Refactored C# interface example</li>
<li>Disable anti-alias filter when switch <li>Disable anti-alias filter when switch
SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias
filter cause slight click if the rate change crosses zero during filter cause slight click if the rate change crosses zero during
processing</li> processing</li>
<li>Added script for building SoundTouchDll dynamic-link-library for GNU platforms</li> <li>Added script for building SoundTouchDll dynamic-link-library for GNU platforms</li>
<li>Rewrote Beats-per-Minute analysis algorithm for more reliable BPM <li>Rewrote Beats-per-Minute analysis algorithm for more reliable BPM detection</li>
detection</li> <li>Added BPM functions to SoundTouchDll API</li>
<li>Added BPM functions to SoundTouchDll API</li> <li>Migrated Visual Studio project files to MSVC 201x format</li>
<li>Migrated Visual Studio project files to MSVC 201x format</li> <li>Replaced function parameter value asserts with runtime exceptions</li>
<li>Code maintenance & style cleanup</li>
</ul> </ul>
<p><b>2.0:</b></p> <p><b>2.0:</b></p>
<ul> <ul>

76
include/BPMDetect.h
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@ -66,82 +66,8 @@ namespace soundtouch
/// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit. /// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM_VALID 190 #define MAX_BPM_VALID 190
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
/*
class BeatCollection
{
private:
int size;
// Ensure there's enough capacity in arrays
void EnsureCapacity(int newCapacity)
{
if (newCapacity > size)
{
// enlarge arrays
int oldSize = size;
float *beatPosOld = beatPos;
float *beatValuesOld = beatValues;
while (size < newCapacity) size *= 2;
printf("Alloc more %d\n", size);
beatPos = new float[size];
beatValues = new float[size];
if ((beatPos == NULL) || (beatValues == NULL))
{
ST_THROW_RT_ERROR("can't allocate memory");
}
// copy old arrays to new arrays
memcpy(beatPos, beatPosOld, sizeof(float)*oldSize);
memcpy(beatValues, beatValuesOld, sizeof(float)*oldSize);
// free old arrays
delete[] beatPosOld;
delete[] beatValuesOld;
}
}
public:
// beat position array
float *beatPos;
// beat values array
float *beatValues;
// number of beats in arrays
int numBeats;
// constructor
BeatCollection()
{
numBeats = 0;
size = 1024;
beatPos = new float[size];
beatValues = new float[size];
}
// destructor
~BeatCollection()
{
delete[] beatPos;
delete[] beatValues;
}
// add new beat position into array
void Add(float pos, float value)
{
EnsureCapacity(numBeats + 1);
beatPos[numBeats] = pos;
beatValues[numBeats] = value;
numBeats++;
}
};
*/
typedef struct typedef struct
{ {
float pos; float pos;
@ -257,7 +183,6 @@ public:
int numSamples ///< Number of samples in buffer int numSamples ///< Number of samples in buffer
); );
/// Analyzes the results and returns the BPM rate. Use this function to read result /// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of /// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function. /// 'inputSamples' function.
@ -265,7 +190,6 @@ public:
/// \return Beats-per-minute rate, or zero if detection failed. /// \return Beats-per-minute rate, or zero if detection failed.
float getBpm(); float getBpm();
/// Get beat position arrays. Note: The array includes also really low beat detection values /// Get beat position arrays. Note: The array includes also really low beat detection values
/// in absence of clear strong beats. Consumer may wish to filter low values away. /// in absence of clear strong beats. Consumer may wish to filter low values away.
/// - "pos" receive array of beat positions /// - "pos" receive array of beat positions

9
include/FIFOSamplePipe.h
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@ -127,7 +127,6 @@ public:
}; };
/// Base-class for sound processing routines working in FIFO principle. With this base /// Base-class for sound processing routines working in FIFO principle. With this base
/// class it's easy to implement sound processing stages that can be chained together, /// class it's easy to implement sound processing stages that can be chained together,
/// so that samples that are fed into beginning of the pipe automatically go through /// so that samples that are fed into beginning of the pipe automatically go through
@ -150,7 +149,6 @@ protected:
output = pOutput; output = pOutput;
} }
/// Constructor. Doesn't define output pipe; it has to be set be /// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function. /// 'setOutPipe' function.
FIFOProcessor() FIFOProcessor()
@ -158,7 +156,6 @@ protected:
output = NULL; output = NULL;
} }
/// Constructor. Configures output pipe. /// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe. FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
) )
@ -166,13 +163,11 @@ protected:
output = pOutput; output = pOutput;
} }
/// Destructor. /// Destructor.
virtual ~FIFOProcessor() virtual ~FIFOProcessor()
{ {
} }
/// Returns a pointer to the beginning of the output samples. /// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly. /// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping! /// Please be careful for not to corrupt the book-keeping!
@ -199,7 +194,6 @@ public:
return output->receiveSamples(outBuffer, maxSamples); return output->receiveSamples(outBuffer, maxSamples);
} }
/// Adjusts book-keeping so that given number of samples are removed from beginning of the /// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere. /// sample buffer without copying them anywhere.
/// ///
@ -211,14 +205,12 @@ public:
return output->receiveSamples(maxSamples); return output->receiveSamples(maxSamples);
} }
/// Returns number of samples currently available. /// Returns number of samples currently available.
virtual uint numSamples() const virtual uint numSamples() const
{ {
return output->numSamples(); return output->numSamples();
} }
/// Returns nonzero if there aren't any samples available for outputting. /// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const virtual int isEmpty() const
{ {
@ -231,7 +223,6 @@ public:
{ {
return output->adjustAmountOfSamples(numSamples); return output->adjustAmountOfSamples(numSamples);
} }
}; };
} }

3
source/SoundStretch/RunParameters.cpp
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@ -151,7 +151,6 @@ RunParameters::RunParameters(const int nParams, const char * const paramStr[])
} }
// Checks parameter limits // Checks parameter limits
void RunParameters::checkLimits() void RunParameters::checkLimits()
{ {
@ -184,7 +183,6 @@ void RunParameters::checkLimits()
} }
// Unknown switch parameter -- throws an exception with an error message // Unknown switch parameter -- throws an exception with an error message
void RunParameters::throwIllegalParamExp(const string &str) const void RunParameters::throwIllegalParamExp(const string &str) const
{ {
@ -196,7 +194,6 @@ void RunParameters::throwIllegalParamExp(const string &str) const
} }
void RunParameters::throwLicense() const void RunParameters::throwLicense() const
{ {
ST_THROW_RT_ERROR(licenseText); ST_THROW_RT_ERROR(licenseText);

21
source/SoundStretch/WavFile.cpp
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@ -56,7 +56,6 @@ 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";
////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////
// //
// Helper functions for swapping byte order to correctly read/write WAV files // Helper functions for swapping byte order to correctly read/write WAV files
@ -228,19 +227,10 @@ void WavInFile::init()
ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters."); ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters.");
} }
/* Ignore 'fixed' field value as 32bit signed linear data can have other value than 1.
if (header.format.fixed != 1)
{
string msg = "Input file uses unsupported encoding.";
ST_THROW_RT_ERROR(msg.c_str());
}
*/
dataRead = 0; dataRead = 0;
} }
WavInFile::~WavInFile() WavInFile::~WavInFile()
{ {
if (fptr) fclose(fptr); if (fptr) fclose(fptr);
@ -248,7 +238,6 @@ WavInFile::~WavInFile()
} }
void WavInFile::rewind() void WavInFile::rewind()
{ {
int hdrsOk; int hdrsOk;
@ -465,7 +454,6 @@ int WavInFile::eof() const
} }
// test if character code is between a white space ' ' and little 'z' // test if character code is between a white space ' ' and little 'z'
static int isAlpha(char c) static int isAlpha(char c)
{ {
@ -506,8 +494,6 @@ int WavInFile::readRIFFBlock()
} }
int WavInFile::readHeaderBlock() int WavInFile::readHeaderBlock()
{ {
char label[5]; char label[5];
@ -679,7 +665,6 @@ uint WavInFile::getSampleRate() const
} }
uint WavInFile::getDataSizeInBytes() const uint WavInFile::getDataSizeInBytes() const
{ {
return header.data.data_len; return header.data.data_len;
@ -713,7 +698,6 @@ uint WavInFile::getElapsedMS() const
} }
////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////
// //
// Class WavOutFile // Class WavOutFile
@ -752,7 +736,6 @@ WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
} }
WavOutFile::~WavOutFile() WavOutFile::~WavOutFile()
{ {
finishHeader(); finishHeader();
@ -761,7 +744,6 @@ WavOutFile::~WavOutFile()
} }
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels) void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
{ {
// fill in the 'riff' part.. // fill in the 'riff' part..
@ -812,7 +794,6 @@ void WavOutFile::finishHeader()
} }
void WavOutFile::writeHeader() void WavOutFile::writeHeader()
{ {
WavHeader hdrTemp; WavHeader hdrTemp;
@ -845,7 +826,6 @@ 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;
@ -866,7 +846,6 @@ 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;

1
source/SoundStretch/WavFile.h
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@ -218,7 +218,6 @@ public:
}; };
/// Class for writing WAV audio files. /// Class for writing WAV audio files.
class WavOutFile : protected WavFileBase class WavOutFile : protected WavFileBase
{ {

4
source/SoundStretch/main.cpp
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@ -108,7 +108,6 @@ static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParamet
} }
// Sets the 'SoundTouch' object up according to input file sound format & // Sets the 'SoundTouch' object up according to input file sound format &
// command line parameters // command line parameters
static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params) static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params)
@ -165,7 +164,6 @@ static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunPar
} }
// Processes the sound // Processes the sound
static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile) static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
{ {
@ -218,7 +216,6 @@ static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outF
} }
// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary // Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
static void detectBPM(WavInFile *inFile, RunParameters *params) static void detectBPM(WavInFile *inFile, RunParameters *params)
{ {
@ -274,7 +271,6 @@ static void detectBPM(WavInFile *inFile, RunParameters *params)
} }
int main(const int nParams, const char * const paramStr[]) int main(const int nParams, const char * const paramStr[])
{ {
WavInFile *inFile; WavInFile *inFile;

7
source/SoundTouch/AAFilter.cpp
View File

@ -68,7 +68,6 @@ using namespace soundtouch;
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y) #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif #endif
/***************************************************************************** /*****************************************************************************
* *
* Implementation of the class 'AAFilter' * Implementation of the class 'AAFilter'
@ -83,14 +82,12 @@ AAFilter::AAFilter(uint len)
} }
AAFilter::~AAFilter() AAFilter::~AAFilter()
{ {
delete pFIR; delete pFIR;
} }
// Sets new anti-alias filter cut-off edge frequency, scaled to // Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5). // sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency. // The filter will cut frequencies higher than the given frequency.
@ -101,7 +98,6 @@ void AAFilter::setCutoffFreq(double newCutoffFreq)
} }
// Sets number of FIR filter taps // Sets number of FIR filter taps
void AAFilter::setLength(uint newLength) void AAFilter::setLength(uint newLength)
{ {
@ -110,7 +106,6 @@ void AAFilter::setLength(uint newLength)
} }
// Calculates coefficients for a low-pass FIR filter using Hamming window // Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs() void AAFilter::calculateCoeffs()
{ {
@ -170,12 +165,10 @@ void AAFilter::calculateCoeffs()
for (i = 0; i < length; i ++) for (i = 0; i < length; i ++)
{ {
temp = work[i] * scaleCoeff; temp = work[i] * scaleCoeff;
//#if SOUNDTOUCH_INTEGER_SAMPLES
// scale & round to nearest integer // scale & round to nearest integer
temp += (temp >= 0) ? 0.5 : -0.5; temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods // ensure no overfloods
assert(temp >= -32768 && temp <= 32767); assert(temp >= -32768 && temp <= 32767);
//#endif
coeffs[i] = (SAMPLETYPE)temp; coeffs[i] = (SAMPLETYPE)temp;
} }

1
source/SoundTouch/FIFOSampleBuffer.cpp
View File

@ -265,4 +265,3 @@ uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
} }
return samplesInBuffer; return samplesInBuffer;
} }

5
source/SoundTouch/FIRFilter.cpp
View File

@ -68,6 +68,7 @@ FIRFilter::~FIRFilter()
delete[] filterCoeffs; delete[] filterCoeffs;
} }
// Usual C-version of the filter routine for stereo sound // Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{ {
@ -126,8 +127,6 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
} }
// Usual C-version of the filter routine for mono sound // Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{ {
@ -253,7 +252,6 @@ uint FIRFilter::getLength() const
} }
// Applies the filter to the given sequence of samples. // Applies the filter to the given sequence of samples.
// //
// Note : The amount of outputted samples is by value of 'filter_length' // Note : The amount of outputted samples is by value of 'filter_length'
@ -283,7 +281,6 @@ 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 s)

6
source/SoundTouch/PeakFinder.cpp
View File

@ -57,7 +57,7 @@ int PeakFinder::findTop(const float *data, int peakpos) const
refvalue = data[peakpos]; refvalue = data[peakpos];
// seek within ±10 points // seek within <EFBFBD>10 points
start = peakpos - 10; start = peakpos - 10;
if (start < minPos) start = minPos; if (start < minPos) start = minPos;
end = peakpos + 10; end = peakpos + 10;
@ -171,7 +171,6 @@ double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos)
} }
/// get exact center of peak near given position by calculating local mass of center /// get exact center of peak near given position by calculating local mass of center
double PeakFinder::getPeakCenter(const float *data, int peakpos) const double PeakFinder::getPeakCenter(const float *data, int peakpos) const
{ {
@ -211,7 +210,6 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
} }
double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos) double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
{ {
@ -258,7 +256,7 @@ double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
// accept harmonic peak if // accept harmonic peak if
// (a) it is found // (a) it is found
// (b) is within ±4% of the expected harmonic interval // (b) is within <EFBFBD>4% of the expected harmonic interval
// (c) has at least half x-corr value of the max. peak // (c) has at least half x-corr value of the max. peak
double diff = harmonic * peaktmp / highPeak; double diff = harmonic * peaktmp / highPeak;

3
source/SoundTouch/RateTransposer.cpp
View File

@ -64,7 +64,6 @@ RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
} }
RateTransposer::~RateTransposer() RateTransposer::~RateTransposer()
{ {
delete pAAFilter; delete pAAFilter;
@ -72,7 +71,6 @@ RateTransposer::~RateTransposer()
} }
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(bool newMode) void RateTransposer::enableAAFilter(bool newMode)
{ {
@ -96,7 +94,6 @@ AAFilter *RateTransposer::getAAFilter()
} }
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower // Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates. // iRate, larger faster iRates.
void RateTransposer::setRate(double newRate) void RateTransposer::setRate(double newRate)

12
source/SoundTouch/RateTransposer.h
View File

@ -125,21 +125,9 @@ public:
RateTransposer(); RateTransposer();
virtual ~RateTransposer(); virtual ~RateTransposer();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetic.
// static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// integer or floating point arithmetic are to be used.
// static RateTransposer *newInstance();
/// Returns the output buffer object /// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; }; FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
// FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object /// Return anti-alias filter object
AAFilter *getAAFilter(); AAFilter *getAAFilter();

29
source/SoundTouch/SoundTouch.cpp
View File

@ -111,7 +111,6 @@ SoundTouch::SoundTouch()
} }
SoundTouch::~SoundTouch() SoundTouch::~SoundTouch()
{ {
delete pRateTransposer; delete pRateTransposer;
@ -119,7 +118,6 @@ SoundTouch::~SoundTouch()
} }
/// Get SoundTouch library version string /// Get SoundTouch library version string
const char *SoundTouch::getVersionString() const char *SoundTouch::getVersionString()
{ {
@ -156,7 +154,6 @@ void SoundTouch::setRate(double newRate)
} }
// Sets new rate control value as a difference in percents compared // Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %) // to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(double newRate) void SoundTouch::setRateChange(double newRate)
@ -166,7 +163,6 @@ void SoundTouch::setRateChange(double newRate)
} }
// Sets new tempo control value. Normal tempo = 1.0, smaller values // Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo. // represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(double newTempo) void SoundTouch::setTempo(double newTempo)
@ -176,7 +172,6 @@ void SoundTouch::setTempo(double newTempo)
} }
// Sets new tempo control value as a difference in percents compared // Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %) // to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(double newTempo) void SoundTouch::setTempoChange(double newTempo)
@ -186,7 +181,6 @@ void SoundTouch::setTempoChange(double newTempo)
} }
// Sets new pitch control value. Original pitch = 1.0, smaller values // Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch. // represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(double newPitch) void SoundTouch::setPitch(double newPitch)
@ -196,7 +190,6 @@ void SoundTouch::setPitch(double newPitch)
} }
// Sets pitch change in octaves compared to the original pitch // Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00) // (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(double newPitch) void SoundTouch::setPitchOctaves(double newPitch)
@ -206,7 +199,6 @@ void SoundTouch::setPitchOctaves(double newPitch)
} }
// Sets pitch change in semi-tones compared to the original pitch // Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12) // (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch) void SoundTouch::setPitchSemiTones(int newPitch)
@ -215,7 +207,6 @@ void SoundTouch::setPitchSemiTones(int newPitch)
} }
void SoundTouch::setPitchSemiTones(double newPitch) void SoundTouch::setPitchSemiTones(double newPitch)
{ {
setPitchOctaves(newPitch / 12.0); setPitchOctaves(newPitch / 12.0);
@ -294,22 +285,6 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined"); ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
} }
// Transpose the rate of the new samples if necessary
/* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
{
// yet flush the last samples in the pitch transposer buffer
// (may happen if 'rate' changes from a non-zero value to zero)
pTDStretch->moveSamples(*pRateTransposer);
}
pTDStretch->putSamples(samples, nSamples);
}
*/
// accumulate how many samples are expected out from processing, given the current // accumulate how many samples are expected out from processing, given the current
// processing setting // processing setting
samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo); samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
@ -365,7 +340,6 @@ void SoundTouch::flush()
delete[] buff; delete[] buff;
// Clear input buffers // Clear input buffers
// pRateTransposer->clearInput();
pTDStretch->clearInput(); pTDStretch->clearInput();
// yet leave the output intouched as that's where the // yet leave the output intouched as that's where the
// flushed samples are! // flushed samples are!
@ -502,7 +476,6 @@ int SoundTouch::getSetting(int settingId) const
} }
// Clears all the samples in the object's output and internal processing // Clears all the samples in the object's output and internal processing
// buffers. // buffers.
void SoundTouch::clear() void SoundTouch::clear()
@ -514,7 +487,6 @@ void SoundTouch::clear()
} }
/// Returns number of samples currently unprocessed. /// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const uint SoundTouch::numUnprocessedSamples() const
{ {
@ -531,7 +503,6 @@ uint SoundTouch::numUnprocessedSamples() const
} }
/// 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
/// 'numsample' samples in the buffer, returns all that available. /// 'numsample' samples in the buffer, returns all that available.

3
source/SoundTouch/TDStretch.h
View File

@ -157,7 +157,6 @@ protected:
void calcSeqParameters(); void calcSeqParameters();
void adaptNormalizer(); void adaptNormalizer();
/// Changes the tempo of the given sound samples. /// Changes the tempo of the given sound samples.
/// Returns amount of samples returned in the "output" buffer. /// Returns amount of samples returned in the "output" buffer.
/// The maximum amount of samples that can be returned at a time is set by /// The maximum amount of samples that can be returned at a time is set by
@ -242,7 +241,6 @@ public:
return seekWindowLength - overlapLength; return seekWindowLength - overlapLength;
} }
/// return approximate initial input-output latency /// return approximate initial input-output latency
int getLatency() const int getLatency() const
{ {
@ -251,7 +249,6 @@ public:
}; };
// Implementation-specific class declarations: // Implementation-specific class declarations:
#ifdef SOUNDTOUCH_ALLOW_MMX #ifdef SOUNDTOUCH_ALLOW_MMX

1
source/SoundTouch/cpu_detect_x86.cpp
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@ -68,7 +68,6 @@ void disableExtensions(uint dwDisableMask)
} }
/// Checks which instruction set extensions are supported by the CPU. /// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void) uint detectCPUextensions(void)
{ {

2
source/SoundTouch/mmx_optimized.cpp
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@ -217,7 +217,6 @@ void TDStretchMMX::clearCrossCorrState()
} }
// MMX-optimized version of the function overlapStereo // MMX-optimized version of the function overlapStereo
void TDStretchMMX::overlapStereo(short *output, const short *input) const void TDStretchMMX::overlapStereo(short *output, const short *input) const
{ {
@ -333,7 +332,6 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
} }
// mmx-optimized version of the filter routine for stereo sound // mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
{ {