From c38f0506dabef9dc2a30d364e91463ab79752b37 Mon Sep 17 00:00:00 2001
From: oparviainen <oparviai@iki.fi>
Date: Sun, 12 Aug 2018 20:51:24 +0300
Subject: [PATCH] Removed commented code, style cleanup

---
 README.html                            | 23 ++++----
 include/BPMDetect.h                    | 76 --------------------------
 include/FIFOSamplePipe.h               |  9 ---
 source/SoundStretch/RunParameters.cpp  |  3 -
 source/SoundStretch/WavFile.cpp        | 21 -------
 source/SoundStretch/WavFile.h          |  1 -
 source/SoundStretch/main.cpp           |  4 --
 source/SoundTouch/AAFilter.cpp         |  7 ---
 source/SoundTouch/FIFOSampleBuffer.cpp |  1 -
 source/SoundTouch/FIRFilter.cpp        |  5 +-
 source/SoundTouch/PeakFinder.cpp       |  6 +-
 source/SoundTouch/RateTransposer.cpp   |  3 -
 source/SoundTouch/RateTransposer.h     | 12 ----
 source/SoundTouch/SoundTouch.cpp       | 29 ----------
 source/SoundTouch/TDStretch.h          |  3 -
 source/SoundTouch/cpu_detect_x86.cpp   |  1 -
 source/SoundTouch/mmx_optimized.cpp    |  2 -
 17 files changed, 15 insertions(+), 191 deletions(-)
diff --git a/README.html b/README.html
index 677319d..6dc3d71 100644
--- a/README.html
+++ b/README.html
@@ -14,7 +14,7 @@
 <body class="normal">
 <hr>
 <h1>SoundTouch audio processing library v2.1pre</h1>
-<p class="normal">SoundTouch library Copyright Š Olli Parviainen 2001-2018</p>
+<p class="normal">SoundTouch library Copyright ďż˝ Olli Parviainen 2001-2018</p>
 <hr>
 <h2>1. Introduction </h2>
 <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>
     <p><b>2.1pre:</b></p>
     <ul>
-        <li>Refactored C# interface example</li>
-        <li>Disable anti-alias filter when switch 
-        SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias 
-        filter cause slight click if the rate change crosses zero during 
-        processing</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 
-		detection</li>
-		<li>Added BPM functions to SoundTouchDll API</li>
-		<li>Migrated Visual Studio project files to MSVC 201x format</li>
+      <li>Refactored C# interface example</li>
+      <li>Disable anti-alias filter when switch 
+      SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias 
+      filter cause slight click if the rate change crosses zero during 
+      processing</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 detection</li>
+      <li>Added BPM functions to SoundTouchDll API</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>
     <p><b>2.0:</b></p>
     <ul>
diff --git a/include/BPMDetect.h b/include/BPMDetect.h
index 8bd776e..8ece784 100644
--- a/include/BPMDetect.h
+++ b/include/BPMDetect.h
@@ -66,82 +66,8 @@ namespace soundtouch
     /// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit.
     #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
     {
         float pos;
@@ -257,7 +183,6 @@ public:
             int numSamples                            ///< Number of samples in buffer
         );
 
-
         /// 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
         /// 'inputSamples' function.
@@ -265,7 +190,6 @@ public:
         /// \return Beats-per-minute rate, or zero if detection failed.
         float getBpm();
 
-
         /// 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.
         /// - "pos" receive array of beat positions
diff --git a/include/FIFOSamplePipe.h b/include/FIFOSamplePipe.h
index b08f836..38ef31a 100644
--- a/include/FIFOSamplePipe.h
+++ b/include/FIFOSamplePipe.h
@@ -127,7 +127,6 @@ public:
 };
 
 
-
 /// 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,
 /// so that samples that are fed into beginning of the pipe automatically go through 
@@ -150,7 +149,6 @@ protected:
         output = pOutput;
     }
 
-
     /// Constructor. Doesn't define output pipe; it has to be set be 
     /// 'setOutPipe' function.
     FIFOProcessor()
@@ -158,7 +156,6 @@ protected:
         output = NULL;
     }
 
-
     /// Constructor. Configures output pipe.
     FIFOProcessor(FIFOSamplePipe *pOutput   ///< Output pipe.
                  )
@@ -166,13 +163,11 @@ protected:
         output = pOutput;
     }
 
-
     /// Destructor.
     virtual ~FIFOProcessor()
     {
     }
 
-
     /// Returns a pointer to the beginning of the output samples. 
     /// This function is provided for accessing the output samples directly. 
     /// Please be careful for not to corrupt the book-keeping!
@@ -199,7 +194,6 @@ public:
         return output->receiveSamples(outBuffer, maxSamples);
     }
 
-
     /// Adjusts book-keeping so that given number of samples are removed from beginning of the 
     /// sample buffer without copying them anywhere. 
     ///
@@ -211,14 +205,12 @@ public:
         return output->receiveSamples(maxSamples);
     }
 
-
     /// Returns number of samples currently available.
     virtual uint numSamples() const
     {
         return output->numSamples();
     }
 
-
     /// Returns nonzero if there aren't any samples available for outputting.
     virtual int isEmpty() const
     {
@@ -231,7 +223,6 @@ public:
     {
         return output->adjustAmountOfSamples(numSamples);
     }
-
 };
 
 }
diff --git a/source/SoundStretch/RunParameters.cpp b/source/SoundStretch/RunParameters.cpp
index 33f726b..1960a35 100644
--- a/source/SoundStretch/RunParameters.cpp
+++ b/source/SoundStretch/RunParameters.cpp
@@ -151,7 +151,6 @@ RunParameters::RunParameters(const int nParams, const char * const paramStr[])
 }
 
 
-
 // Checks parameter limits
 void RunParameters::checkLimits()
 {
@@ -184,7 +183,6 @@ void RunParameters::checkLimits()
 }
 
 
-
 // Unknown switch parameter -- throws an exception with an error message
 void RunParameters::throwIllegalParamExp(const string &str) const
 {
@@ -196,7 +194,6 @@ void RunParameters::throwIllegalParamExp(const string &str) const
 }
 
 
-
 void RunParameters::throwLicense() const
 {
     ST_THROW_RT_ERROR(licenseText);
diff --git a/source/SoundStretch/WavFile.cpp b/source/SoundStretch/WavFile.cpp
index 68818c9..4af7a4c 100644
--- a/source/SoundStretch/WavFile.cpp
+++ b/source/SoundStretch/WavFile.cpp
@@ -56,7 +56,6 @@ static const char fmtStr[]  = "fmt ";
 static const char factStr[] = "fact";
 static const char dataStr[] = "data";
 
-
 //////////////////////////////////////////////////////////////////////////////
 //
 // Helper functions for swapping byte order to correctly read/write WAV files 
@@ -228,19 +227,10 @@ void WavInFile::init()
         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;
 }
 
 
-
 WavInFile::~WavInFile()
 {
     if (fptr) fclose(fptr);
@@ -248,7 +238,6 @@ WavInFile::~WavInFile()
 }
 
 
-
 void WavInFile::rewind()
 {
     int hdrsOk;
@@ -465,7 +454,6 @@ int WavInFile::eof() const
 }
 
 
-
 // test if character code is between a white space ' ' and little 'z'
 static int isAlpha(char c)
 {
@@ -506,8 +494,6 @@ int WavInFile::readRIFFBlock()
 }
 
 
-
-
 int WavInFile::readHeaderBlock()
 {
     char label[5];
@@ -679,7 +665,6 @@ uint WavInFile::getSampleRate() const
 }
 
 
-
 uint WavInFile::getDataSizeInBytes() const
 {
     return header.data.data_len;
@@ -713,7 +698,6 @@ uint WavInFile::getElapsedMS() const
 }
 
 
-
 //////////////////////////////////////////////////////////////////////////////
 //
 // Class WavOutFile
@@ -752,7 +736,6 @@ WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
 }
 
 
-
 WavOutFile::~WavOutFile()
 {
     finishHeader();
@@ -761,7 +744,6 @@ WavOutFile::~WavOutFile()
 }
 
 
-
 void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
 {
     // fill in the 'riff' part..
@@ -812,7 +794,6 @@ void WavOutFile::finishHeader()
 }
 
 
-
 void WavOutFile::writeHeader()
 {
     WavHeader hdrTemp;
@@ -845,7 +826,6 @@ void WavOutFile::writeHeader()
 }
 
 
-
 void WavOutFile::write(const unsigned char *buffer, int numElems)
 {
     int res;
@@ -866,7 +846,6 @@ void WavOutFile::write(const unsigned char *buffer, int numElems)
 }
 
 
-
 void WavOutFile::write(const short *buffer, int numElems)
 {
     int res;
diff --git a/source/SoundStretch/WavFile.h b/source/SoundStretch/WavFile.h
index c6a2eb3..b0f4d96 100644
--- a/source/SoundStretch/WavFile.h
+++ b/source/SoundStretch/WavFile.h
@@ -218,7 +218,6 @@ public:
 };
 
 
-
 /// Class for writing WAV audio files.
 class WavOutFile : protected WavFileBase
 {
diff --git a/source/SoundStretch/main.cpp b/source/SoundStretch/main.cpp
index 88eb003..360ae8c 100644
--- a/source/SoundStretch/main.cpp
+++ b/source/SoundStretch/main.cpp
@@ -108,7 +108,6 @@ static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParamet
 }
 
 
-
 // Sets the 'SoundTouch' object up according to input file sound format & 
 // command line parameters
 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
 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
 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[])
 {
     WavInFile *inFile;
diff --git a/source/SoundTouch/AAFilter.cpp b/source/SoundTouch/AAFilter.cpp
index c7af576..76a3da6 100644
--- a/source/SoundTouch/AAFilter.cpp
+++ b/source/SoundTouch/AAFilter.cpp
@@ -68,7 +68,6 @@ using namespace soundtouch;
     #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
 #endif
 
-
 /*****************************************************************************
  *
  * Implementation of the class 'AAFilter'
@@ -83,14 +82,12 @@ AAFilter::AAFilter(uint len)
 }
 
 
-
 AAFilter::~AAFilter()
 {
     delete pFIR;
 }
 
 
-
 // Sets new anti-alias filter cut-off edge frequency, scaled to
 // sampling frequency (nyquist frequency = 0.5).
 // 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
 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
 void AAFilter::calculateCoeffs()
 {
@@ -170,12 +165,10 @@ void AAFilter::calculateCoeffs()
     for (i = 0; i < length; i ++) 
     {
         temp = work[i] * scaleCoeff;
-//#if SOUNDTOUCH_INTEGER_SAMPLES
         // scale & round to nearest integer
         temp += (temp >= 0) ? 0.5 : -0.5;
         // ensure no overfloods
         assert(temp >= -32768 && temp <= 32767);
-//#endif
         coeffs[i] = (SAMPLETYPE)temp;
     }
 
diff --git a/source/SoundTouch/FIFOSampleBuffer.cpp b/source/SoundTouch/FIFOSampleBuffer.cpp
index 706e869..8341163 100644
--- a/source/SoundTouch/FIFOSampleBuffer.cpp
+++ b/source/SoundTouch/FIFOSampleBuffer.cpp
@@ -265,4 +265,3 @@ uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
     }
     return samplesInBuffer;
 }
-
diff --git a/source/SoundTouch/FIRFilter.cpp b/source/SoundTouch/FIRFilter.cpp
index e41cce9..218e50e 100644
--- a/source/SoundTouch/FIRFilter.cpp
+++ b/source/SoundTouch/FIRFilter.cpp
@@ -68,6 +68,7 @@ FIRFilter::~FIRFilter()
     delete[] filterCoeffs;
 }
 
+
 // Usual C-version of the filter routine for stereo sound
 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
 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. 
 //
 // 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 
 // depending on if we've a MMX-capable CPU available or not.
 void * FIRFilter::operator new(size_t s)
diff --git a/source/SoundTouch/PeakFinder.cpp b/source/SoundTouch/PeakFinder.cpp
index 8896b14..258571a 100644
--- a/source/SoundTouch/PeakFinder.cpp
+++ b/source/SoundTouch/PeakFinder.cpp
@@ -57,7 +57,7 @@ int PeakFinder::findTop(const float *data, int peakpos) const
 
     refvalue = data[peakpos];
 
-    // seek within ą10 points
+    // seek within ďż˝10 points
     start = peakpos - 10;
     if (start < minPos) start = minPos;
     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
 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) 
 {
 
@@ -258,7 +256,7 @@ double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
 
         // accept harmonic peak if 
         // (a) it is found
-        // (b) is within ą4% of the expected harmonic interval
+        // (b) is within ďż˝4% of the expected harmonic interval
         // (c) has at least half x-corr value of the max. peak
 
         double diff = harmonic * peaktmp / highPeak;
diff --git a/source/SoundTouch/RateTransposer.cpp b/source/SoundTouch/RateTransposer.cpp
index 3cf5e1d..2efaf04 100644
--- a/source/SoundTouch/RateTransposer.cpp
+++ b/source/SoundTouch/RateTransposer.cpp
@@ -64,7 +64,6 @@ RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
 }
 
 
-
 RateTransposer::~RateTransposer()
 {
     delete pAAFilter;
@@ -72,7 +71,6 @@ RateTransposer::~RateTransposer()
 }
 
 
-
 /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
 void RateTransposer::enableAAFilter(bool newMode)
 {
@@ -96,7 +94,6 @@ AAFilter *RateTransposer::getAAFilter()
 }
 
 
-
 // Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
 // iRate, larger faster iRates.
 void RateTransposer::setRate(double newRate)
diff --git a/source/SoundTouch/RateTransposer.h b/source/SoundTouch/RateTransposer.h
index cee4a58..52b7441 100644
--- a/source/SoundTouch/RateTransposer.h
+++ b/source/SoundTouch/RateTransposer.h
@@ -125,21 +125,9 @@ public:
     RateTransposer();
     virtual ~RateTransposer();
 
-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
-    /// depending on if we're to use integer or floating point 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
     FIFOSamplePipe *getOutput() { return &outputBuffer; };
 
-    /// Returns the store buffer object
-//    FIFOSamplePipe *getStore() { return &storeBuffer; };
-
     /// Return anti-alias filter object
     AAFilter *getAAFilter();
 
diff --git a/source/SoundTouch/SoundTouch.cpp b/source/SoundTouch/SoundTouch.cpp
index 06bdd56..1618884 100644
--- a/source/SoundTouch/SoundTouch.cpp
+++ b/source/SoundTouch/SoundTouch.cpp
@@ -111,7 +111,6 @@ SoundTouch::SoundTouch()
 }
 
 
-
 SoundTouch::~SoundTouch()
 {
     delete pRateTransposer;
@@ -119,7 +118,6 @@ SoundTouch::~SoundTouch()
 }
 
 
-
 /// Get SoundTouch library version string
 const char *SoundTouch::getVersionString()
 {
@@ -156,7 +154,6 @@ void SoundTouch::setRate(double newRate)
 }
 
 
-
 // Sets new rate control value as a difference in percents compared
 // to the original rate (-50 .. +100 %)
 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
 // represent slower tempo, larger faster tempo.
 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
 // to the original tempo (-50 .. +100 %)
 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
 // represent lower pitches, larger values higher pitch.
 void SoundTouch::setPitch(double newPitch)
@@ -196,7 +190,6 @@ void SoundTouch::setPitch(double newPitch)
 }
 
 
-
 // Sets pitch change in octaves compared to the original pitch
 // (-1.00 .. +1.00)
 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
 // (-12 .. +12)
 void SoundTouch::setPitchSemiTones(int newPitch)
@@ -215,7 +207,6 @@ void SoundTouch::setPitchSemiTones(int newPitch)
 }
 
 
-
 void SoundTouch::setPitchSemiTones(double newPitch)
 {
     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");
     }
 
-    // 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 
     // processing setting
     samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
@@ -365,7 +340,6 @@ void SoundTouch::flush()
     delete[] buff;
 
     // Clear input buffers
- //   pRateTransposer->clearInput();
     pTDStretch->clearInput();
     // yet leave the output intouched as that's where the
     // 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
 // buffers.
 void SoundTouch::clear()
@@ -514,7 +487,6 @@ void SoundTouch::clear()
 }
 
 
-
 /// Returns number of samples currently unprocessed.
 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 buffer and removes them from the sample buffer. If there are less than 
 /// 'numsample' samples in the buffer, returns all that available.
diff --git a/source/SoundTouch/TDStretch.h b/source/SoundTouch/TDStretch.h
index 44fe8bc..4118f9f 100644
--- a/source/SoundTouch/TDStretch.h
+++ b/source/SoundTouch/TDStretch.h
@@ -157,7 +157,6 @@ protected:
     void calcSeqParameters();
     void adaptNormalizer();
 
-
     /// Changes the tempo of the given sound samples.
     /// Returns amount of samples returned in the "output" buffer.
     /// The maximum amount of samples that can be returned at a time is set by
@@ -242,7 +241,6 @@ public:
         return seekWindowLength - overlapLength;
     }
 
-
 	/// return approximate initial input-output latency
 	int getLatency() const
 	{
@@ -251,7 +249,6 @@ public:
 };
 
 
-
 // Implementation-specific class declarations:
 
 #ifdef SOUNDTOUCH_ALLOW_MMX
diff --git a/source/SoundTouch/cpu_detect_x86.cpp b/source/SoundTouch/cpu_detect_x86.cpp
index 84df0cf..b128610 100644
--- a/source/SoundTouch/cpu_detect_x86.cpp
+++ b/source/SoundTouch/cpu_detect_x86.cpp
@@ -68,7 +68,6 @@ void disableExtensions(uint dwDisableMask)
 }
 
 
-
 /// Checks which instruction set extensions are supported by the CPU.
 uint detectCPUextensions(void)
 {
diff --git a/source/SoundTouch/mmx_optimized.cpp b/source/SoundTouch/mmx_optimized.cpp
index 2947b3d..741ba4f 100644
--- a/source/SoundTouch/mmx_optimized.cpp
+++ b/source/SoundTouch/mmx_optimized.cpp
@@ -217,7 +217,6 @@ void TDStretchMMX::clearCrossCorrState()
 }
 
 
-
 // MMX-optimized version of the function overlapStereo
 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
 uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
 {
