Cleaned code with Lint

include/BPMDetect.h

@@ -108,9 +108,6 @@ protected:
     /// FIFO-buffer for decimated processing samples.
     soundtouch::FIFOSampleBuffer *buffer;
 
-    /// Initialize the class for processing.
-    void init(int numChannels, int sampleRate);
-
     /// Updates auto-correlation function for given number of decimated samples that 
     /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe 
     /// though).

include/FIFOSampleBuffer.h

@@ -107,7 +107,7 @@ public:
     /// When using this function to output samples, also remember to 'remove' the
     /// output samples from the buffer by calling the 
     /// 'receiveSamples(numSamples)' function
-    virtual SAMPLETYPE *ptrBegin() const;
+    virtual SAMPLETYPE *ptrBegin();
 
     /// Returns a pointer to the end of the used part of the sample buffer (i.e. 
     /// where the new samples are to be inserted). This function may be used for 

include/FIFOSamplePipe.h

@@ -66,7 +66,7 @@ public:
     /// When using this function to output samples, also remember to 'remove' the
     /// output samples from the buffer by calling the 
     /// 'receiveSamples(numSamples)' function
-    virtual SAMPLETYPE *ptrBegin() const = 0;
+    virtual SAMPLETYPE *ptrBegin() = 0;
 
     /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
     /// the sample buffer.
@@ -166,7 +166,7 @@ protected:
     /// When using this function to output samples, also remember to 'remove' the
     /// output samples from the buffer by calling the 
     /// 'receiveSamples(numSamples)' function
-    virtual SAMPLETYPE *ptrBegin() const
+    virtual SAMPLETYPE *ptrBegin()
     {
         return output->ptrBegin();
     }

source/SoundStretch/RunParameters.cpp

@@ -103,7 +103,7 @@ static int _toLowerCase(int c)
 
 
 // Constructor
-RunParameters::RunParameters(const int nParams, const char *paramStr[])
+RunParameters::RunParameters(const int nParams, const char * const paramStr[])
 {
     int i;
     int nFirstParam;
@@ -213,7 +213,7 @@ float RunParameters::parseSwitchValue(const string &str) const
 {
     int pos;
 
-    pos = str.find_first_of('=');
+    pos = (int)str.find_first_of('=');
     if (pos < 0) 
     {
         // '=' missing
@@ -266,7 +266,7 @@ void RunParameters::parseSwitchParam(const string &str)
             {
                 goalBPM = parseSwitchValue(str);
             } 
-            catch (runtime_error)
+            catch (const runtime_error)
             {
                 // illegal or missing bpm value => just calculate bpm
                 goalBPM = 0;

source/SoundStretch/RunParameters.h

@@ -65,7 +65,7 @@ public:
     float goalBPM;
     BOOL  detectBPM;
 
-    RunParameters(const int nParams, const char *paramStr[]);
+    RunParameters(const int nParams, const char * const paramStr[]);
 };
 
 #endif

source/SoundStretch/WavFile.cpp

@@ -56,10 +56,10 @@
 
 using namespace std;
 
-const static char riffStr[] = "RIFF";
+static const char riffStr[] = "RIFF";
-const static char waveStr[] = "WAVE";
+static const char waveStr[] = "WAVE";
-const static char fmtStr[]  = "fmt ";
+static const char fmtStr[]  = "fmt ";
-const static char dataStr[] = "data";
+static const char dataStr[] = "data";
 
 
 //////////////////////////////////////////////////////////////////////////////
@@ -200,7 +200,8 @@ void WavInFile::init()
 
 WavInFile::~WavInFile()
 {
-    close();
+    if (fptr) fclose(fptr);
+    fptr = NULL;
 }
 
 
@@ -216,7 +217,7 @@ void WavInFile::rewind()
 }
 
 
-int WavInFile::checkCharTags()
+int WavInFile::checkCharTags() const
 {
     // header.format.fmt should equal to 'fmt '
     if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
@@ -244,10 +245,11 @@ int WavInFile::read(char *buffer, int maxElems)
     if (afterDataRead > header.data.data_len) 
     {
         // Don't read more samples than are marked available in header
-        numBytes = header.data.data_len - dataRead;
+        numBytes = (int)header.data.data_len - (int)dataRead;
         assert(numBytes >= 0);
     }
 
+    assert(buffer);
     numBytes = fread(buffer, 1, numBytes, fptr);
     dataRead += numBytes;
 
@@ -261,6 +263,7 @@ int WavInFile::read(short *buffer, int maxElems)
     int numBytes;
     int numElems;
 
+    assert(buffer);
     if (header.format.bits_per_sample == 8)
     {
         // 8 bit format
@@ -286,7 +289,7 @@ int WavInFile::read(short *buffer, int maxElems)
         if (afterDataRead > header.data.data_len) 
         {
             // Don't read more samples than are marked available in header
-            numBytes = header.data.data_len - dataRead;
+            numBytes = (int)header.data.data_len - (int)dataRead;
             assert(numBytes >= 0);
         }
 
@@ -332,13 +335,6 @@ int WavInFile::eof() const
 }
 
 
-void WavInFile::close()
-{
-    fclose(fptr);
-    fptr = NULL;
-}
-
-
 
 // test if character code is between a white space ' ' and little 'z'
 static int isAlpha(char c)
@@ -348,9 +344,9 @@ static int isAlpha(char c)
 
 
 // test if all characters are between a white space ' ' and little 'z'
-static int isAlphaStr(char *str)
+static int isAlphaStr(const char *str)
 {
-    int c;
+    char c;
 
     c = str[0];
     while (c) 
@@ -408,7 +404,7 @@ int WavInFile::readHeaderBlock()
         header.format.format_len = nLen;
 
         // calculate how much length differs from expected
-        nDump = nLen - (sizeof(header.format) - 8);
+        nDump = nLen - ((int)sizeof(header.format) - 8);
 
         // if format_len is larger than expected, read only as much data as we've space for
         if (nDump > 0)
@@ -518,7 +514,8 @@ uint WavInFile::getDataSizeInBytes() const
 
 uint WavInFile::getNumSamples() const
 {
-    return header.data.data_len / header.format.byte_per_sample;
+    if (header.format.byte_per_sample == 0) return 0;
+    return header.data.data_len / (unsigned short)header.format.byte_per_sample;
 }
 
 
@@ -576,7 +573,9 @@ WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
 
 WavOutFile::~WavOutFile()
 {
-    close();
+    finishHeader();
+    if (fptr) fclose(fptr);
+    fptr = NULL;
 }
 
 
@@ -601,11 +600,11 @@ void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
     header.format.format_len = 0x10;
     header.format.fixed = 1;
     header.format.channel_number = (short)channels;
-    header.format.sample_rate = sampleRate;
+    header.format.sample_rate = (int)sampleRate;
     header.format.bits_per_sample = (short)bits;
     header.format.byte_per_sample = (short)(bits * channels / 8);
-    header.format.byte_rate = header.format.byte_per_sample * sampleRate;
+    header.format.byte_rate = header.format.byte_per_sample * (int)sampleRate;
-    header.format.sample_rate = sampleRate;
+    header.format.sample_rate = (int)sampleRate;
 
     // fill in the 'data' part..
 
@@ -658,14 +657,6 @@ void WavOutFile::writeHeader()
 
 
 
-void WavOutFile::close()
-{
-    finishHeader();
-    fclose(fptr);
-    fptr = NULL;
-}
-
-
 void WavOutFile::write(const char *buffer, int numElems)
 {
     int res;

source/SoundStretch/WavFile.h

@@ -114,7 +114,7 @@ private:
 
     /// Checks WAV file header tags.
     /// \return zero if all ok, nonzero if file format is invalid.
-    int checkCharTags();
+    int checkCharTags() const;
 
     /// Reads a single WAV file header block.
     /// \return zero if all ok, nonzero if file format is invalid.
@@ -133,10 +133,6 @@ public:
     /// Destructor: Closes the file.
     ~WavInFile();
 
-    /// Close the file. Notice that file is automatically closed also when the 
-    /// class instance is deleted.
-    void close();
-
     /// Rewind to beginning of the file
     void rewind();
 
@@ -249,12 +245,6 @@ public:
     void write(const float *buffer,     ///< Pointer to sample data buffer.
                int numElems             ///< How many array items are to be written to file.
                );
-
-    /// Finalize & close the WAV file. Automatically supplements the WAV file header
-    /// information according to written data etc.
-    ///
-    /// Notice that file is automatically closed also when the class instance is deleted.
-    void close();
 };
 
 #endif

source/SoundStretch/main.cpp

@@ -91,9 +91,9 @@ static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParamet
     }
 
     // ... open output file with same sound parameters
-    bits = (*inFile)->getNumBits();
+    bits = (int)(*inFile)->getNumBits();
-    samplerate = (*inFile)->getSampleRate();
+    samplerate = (int)(*inFile)->getSampleRate();
-    channels = (*inFile)->getNumChannels();
+    channels = (int)(*inFile)->getNumChannels();
 
     if (params->outFileName)
     {
@@ -122,8 +122,8 @@ static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunPar
     int sampleRate;
     int channels;
 
-    sampleRate = inFile->getSampleRate();
+    sampleRate = (int)inFile->getSampleRate();
-    channels = inFile->getNumChannels();
+    channels = (int)inFile->getNumChannels();
     pSoundTouch->setSampleRate(sampleRate);
     pSoundTouch->setChannels(channels);
 
@@ -132,7 +132,7 @@ static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunPar
     pSoundTouch->setRateChange(params->rateDelta);
 
     pSoundTouch->setSetting(SETTING_USE_QUICKSEEK, params->quick);
-    pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !params->noAntiAlias);
+    pSoundTouch->setSetting(SETTING_USE_AA_FILTER, !(params->noAntiAlias));
 
     // print processing information
     if (params->outFileName)
@@ -174,7 +174,8 @@ static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outF
 
     if ((inFile == NULL) || (outFile == NULL)) return;  // nothing to do.
 
-    nChannels = inFile->getNumChannels();
+    nChannels = (int)inFile->getNumChannels();
+    assert(nChannels > 0);
     buffSizeSamples = BUFF_SIZE / nChannels;
 
     // Process samples read from the input file
@@ -184,7 +185,7 @@ static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outF
 
         // Read a chunk of samples from the input file
         num = inFile->read(sampleBuffer, BUFF_SIZE);
-        nSamples = num / inFile->getNumChannels();
+        nSamples = num / (int)inFile->getNumChannels();
 
         // Feed the samples into SoundTouch processor
         pSoundTouch->putSamples(sampleBuffer, nSamples);
@@ -228,7 +229,7 @@ static void detectBPM(WavInFile *inFile, RunParameters *params)
     fprintf(stderr, "Detecting BPM rate...");
     fflush(stderr);
 
-    nChannels = inFile->getNumChannels();
+    nChannels = (int)inFile->getNumChannels();
     assert(BUFF_SIZE % nChannels == 0);
 
     // Process the 'inFile' in small blocks, repeat until whole file has 
@@ -272,7 +273,7 @@ static void detectBPM(WavInFile *inFile, RunParameters *params)
 
 
 
-int main(const int nParams, const char *paramStr[])
+int main(const int nParams, const char * const paramStr[])
 {
     WavInFile *inFile;
     WavOutFile *outFile;
@@ -309,7 +310,7 @@ int main(const int nParams, const char *paramStr[])
 
         fprintf(stderr, "Done!\n");
     } 
-    catch (runtime_error &e) 
+    catch (const runtime_error &e) 
     {
         // An exception occurred during processing, display an error message
         fprintf(stderr, "%s\n", e.what());

source/SoundTouch/3dnow_win.cpp

@@ -82,17 +82,13 @@ using namespace soundtouch;
 //////////////////////////////////////////////////////////////////////////////
 
 #include "TDStretch.h"
-//#include <limits.h>
-
-// these are declared in 'TDStretch.cpp'
-// extern int scanOffsets[4][24];
 
 
 // Calculates cross correlation of two buffers
 double TDStretch3DNow::calcCrossCorrStereo(const float *pV1, const float *pV2) const
 {
-    uint overlapLengthLocal = overlapLength;
+    int overlapLengthLocal = overlapLength;
-    float corr;
+    float corr = 0;
 
     // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
     /*

source/SoundTouch/BPMDetect.cpp

@@ -66,8 +66,6 @@ using namespace soundtouch;
 #define INPUT_BLOCK_SAMPLES       2048
 #define DECIMATED_BLOCK_SAMPLES   256
 
-typedef unsigned short ushort;
-
 /// decay constant for calculating RMS volume sliding average approximation 
 /// (time constant is about 10 sec)
 const float avgdecay = 0.99986f;
@@ -77,18 +75,13 @@ const float avgnorm = (1 - avgdecay);
 
 
 
-BPMDetect::BPMDetect(int numChannels, int sampleRate)
+BPMDetect::BPMDetect(int numChannels, int aSampleRate)
 {
-    xcorr = NULL;
+    this->sampleRate = aSampleRate;
-
+    this->channels = numChannels;
-    buffer = new FIFOSampleBuffer();
 
     decimateSum = 0;
     decimateCount = 0;
-    decimateBy = 0;
-
-    this->sampleRate = sampleRate;
-    this->channels = numChannels;
 
     envelopeAccu = 0;
 
@@ -103,7 +96,26 @@ BPMDetect::BPMDetect(int numChannels, int sampleRate)
     RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;
 #endif
 
-    init(numChannels, sampleRate);
+    // choose decimation factor so that result is approx. 500 Hz
+    decimateBy = sampleRate / 500;
+    assert(decimateBy > 0);
+    assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
+
+    // Calculate window length & starting item according to desired min & max bpms
+    windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
+    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
+
+    assert(windowLen > windowStart);
+
+    // allocate new working objects
+    xcorr = new float[windowLen];
+    memset(xcorr, 0, windowLen * sizeof(float));
+
+    // allocate processing buffer
+    buffer = new FIFOSampleBuffer();
+    // we do processing in mono mode
+    buffer->setChannels(1);
+    buffer->clear();
 }
 
 
@@ -115,6 +127,7 @@ BPMDetect::~BPMDetect()
 }
 
 
+
 /// convert to mono, low-pass filter & decimate to about 500 Hz. 
 /// return number of outputted samples.
 ///
@@ -131,7 +144,8 @@ int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
     int count, outcount;
     LONG_SAMPLETYPE out;
 
-    assert(decimateBy != 0);
+    assert(channels > 0);
+    assert(decimateBy > 0);
     outcount = 0;
     for (count = 0; count < numsamples; count ++) 
     {
@@ -267,7 +281,7 @@ void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
         int processLength;
 
         // how many samples are processed
-        processLength = buffer->numSamples() - windowLen;
+        processLength = (int)buffer->numSamples() - windowLen;
 
         // ... calculate autocorrelations for oldest samples...
         updateXCorr(processLength);
@@ -277,31 +291,6 @@ void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
 }
 
 
-void BPMDetect::init(int numChannels, int sampleRate)
-{
-    this->sampleRate = sampleRate;
-
-    // choose decimation factor so that result is approx. 500 Hz
-    decimateBy = sampleRate / 500;
-    assert(decimateBy > 0);
-    assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
-
-    // Calculate window length & starting item according to desired min & max bpms
-    windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
-    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
-
-    assert(windowLen > windowStart);
-
-    // allocate new working objects
-    xcorr = new float[windowLen];
-    memset(xcorr, 0, windowLen * sizeof(float));
-
-    // we do processing in mono mode
-    buffer->setChannels(1);
-    buffer->clear();
-}
-
-
 
 float BPMDetect::getBpm()
 {

source/SoundTouch/FIFOSampleBuffer.cpp

@@ -63,6 +63,7 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
     samplesInBuffer = 0;
     bufferPos = 0;
     channels = (uint)numChannels;
+    ensureCapacity(32);     // allocate initial capacity 
 }
 
 
@@ -151,8 +152,9 @@ SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
 // When using this function to output samples, also remember to 'remove' the
 // outputted samples from the buffer by calling the 
 // 'receiveSamples(numSamples)' function
-SAMPLETYPE *FIFOSampleBuffer::ptrBegin() const
+SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
 {
+    assert(buffer);
     return buffer + bufferPos * channels;
 }
 
@@ -175,6 +177,7 @@ void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
         {
             throw std::runtime_error("Couldn't allocate memory!\n");
         }
+        // Align the buffer to begin at 16byte cache line boundary for optimal performance
         temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & (ulong)-16);
         memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
         delete[] bufferUnaligned;

source/SoundTouch/FIRFilter.cpp

@@ -181,7 +181,7 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
     assert(length == newLength);
 
     resultDivFactor = uResultDivFactor;
-    resultDivider = (SAMPLETYPE)::pow(2, resultDivFactor);
+    resultDivider = (SAMPLETYPE)::pow(2, (int)resultDivFactor);
 
     delete[] filterCoeffs;
     filterCoeffs = new SAMPLETYPE[length];

source/SoundTouch/FIRFilter.h

@@ -42,6 +42,7 @@
 #ifndef FIRFilter_H
 #define FIRFilter_H
 
+#include <stddef.h>
 #include "STTypes.h"
 
 namespace soundtouch
@@ -103,7 +104,7 @@ public:
 
 #ifdef ALLOW_MMX
 
-    /// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
+/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
     class FIRFilterMMX : public FIRFilter
     {
     protected:

source/SoundTouch/PeakFinder.cpp

@@ -51,6 +51,7 @@ using namespace soundtouch;
 
 PeakFinder::PeakFinder()
 {
+    minPos = maxPos = 0;
 }
 
 
@@ -140,13 +141,15 @@ double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos)
         sum += (float)i * data[i];
         wsum += data[i];
     }
+
+    if (wsum < 1e-6) return 0;
     return sum / wsum;
 }
 
 
 
 /// get exact center of peak near given position by calculating local mass of center
-double PeakFinder::getPeakCenter(const float *data, int peakpos)
+double PeakFinder::getPeakCenter(const float *data, int peakpos) const
 {
     float peakLevel;            // peak level
     int crosspos1, crosspos2;   // position where the peak 'hump' crosses cutting level
@@ -178,15 +181,15 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos)
 
 
 
-double PeakFinder::detectPeak(const float *data, int minPos, int maxPos) 
+double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos) 
 {
 
     int i;
     int peakpos;                // position of peak level
     double highPeak, peak;
 
-    this->minPos = minPos;
+    this->minPos = aminPos;
-    this->maxPos = maxPos;
+    this->maxPos = amaxPos;
 
     // find absolute peak
     peakpos = minPos;

source/SoundTouch/PeakFinder.h

@@ -71,7 +71,7 @@ protected:
                      ) const;
 
     /// get exact center of peak near given position by calculating local mass of center
-    double getPeakCenter(const float *data, int peakpos);
+    double getPeakCenter(const float *data, int peakpos) const;
 
 public:
     /// Constructor. 

source/SoundTouch/RateTransposer.cpp

@@ -158,7 +158,7 @@ BOOL RateTransposer::isAAFilterEnabled() const
 }
 
 
-AAFilter *RateTransposer::getAAFilter() const
+AAFilter *RateTransposer::getAAFilter()
 {
     return pAAFilter;
 }
@@ -548,8 +548,8 @@ uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src,
     }
     fSlopeCount -= 1.0f;
 
-    if (nSamples == 1) goto end;
+    if (nSamples > 1)
-
+    {
         while (1)
         {
             while (fSlopeCount > 1.0f) 
@@ -562,6 +562,7 @@ uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src,
             i++;
             fSlopeCount += fRate;
         }
+    }
 end:
     // Store the last sample for the next round
     sPrevSampleL = src[nSamples - 1];
@@ -593,8 +594,8 @@ uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *sr
     // now always (iSlopeCount > 1.0f)
     fSlopeCount -= 1.0f;
 
-    if (nSamples == 1) goto end;
+    if (nSamples > 1)
-
+    {
         while (1)
         {
             while (fSlopeCount > 1.0f) 
@@ -613,6 +614,7 @@ uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *sr
             i++;
             fSlopeCount += fRate;
         }
+    }
 end:
     // Store the last sample for the next round
     sPrevSampleL = src[2 * nSamples - 2];

source/SoundTouch/RateTransposer.h

@@ -45,6 +45,7 @@
 #ifndef RateTransposer_H
 #define RateTransposer_H
 
+#include <stddef.h>
 #include "AAFilter.h"
 #include "FIFOSamplePipe.h"
 #include "FIFOSampleBuffer.h"
@@ -90,7 +91,7 @@ protected:
     virtual uint transposeMono(SAMPLETYPE *dest, 
                        const SAMPLETYPE *src, 
                        uint numSamples) = 0;
-    uint transpose(SAMPLETYPE *dest, 
+    inline uint transpose(SAMPLETYPE *dest, 
                    const SAMPLETYPE *src, 
                    uint numSamples);
 
@@ -127,7 +128,7 @@ public:
     FIFOSamplePipe *getStore() { return &storeBuffer; };
 
     /// Return anti-alias filter object
-    AAFilter *getAAFilter() const;
+    AAFilter *getAAFilter();
 
     /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
     void enableAAFilter(BOOL newMode);

source/SoundTouch/SoundTouch.cpp

@@ -87,7 +87,7 @@ using namespace soundtouch;
 #define TEST_FLOAT_EQUAL(a, b)  (fabs(a - b) < 1e-10)
 
 
-/// Print library version string
+/// Print library version string for autoconf
 extern "C" void soundtouch_ac_test()
 {
     printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
@@ -149,8 +149,8 @@ void SoundTouch::setChannels(uint numChannels)
         throw std::runtime_error("Illegal number of channels");
     }
     channels = numChannels;
-    pRateTransposer->setChannels(numChannels);
+    pRateTransposer->setChannels((int)numChannels);
-    pTDStretch->setChannels(numChannels);
+    pTDStretch->setChannels((int)numChannels);
 }
 
 
@@ -284,7 +284,7 @@ void SoundTouch::setSampleRate(uint srate)
 {
     bSrateSet = TRUE;
     // set sample rate, leave other tempo changer parameters as they are.
-    pTDStretch->setParameters(srate);
+    pTDStretch->setParameters((int)srate);
 }
 
 

source/SoundTouch/SoundTouch.vcproj

@@ -331,6 +331,9 @@
 			<File
 				RelativePath="FIRFilter.h">
 			</File>
+			<File
+				RelativePath=".\PeakFinder.h">
+			</File>
 			<File
 				RelativePath="RateTransposer.h">
 			</File>

source/SoundTouch/TDStretch.cpp

@@ -53,11 +53,7 @@
 
 using namespace soundtouch;
 
-#ifndef min
+#define max(x, y) (((x) > (y)) ? (x) : (y))
-//#define min(a,b) (((a) > (b)) ? (b) : (a))
-#define max(a,b) (((a) < (b)) ? (b) : (a))
-#endif
-
 
 
 /*****************************************************************************
@@ -513,7 +509,7 @@ void TDStretch::calcSeqParameters()
     #define AUTOSEEK_K          ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
     #define AUTOSEEK_C          (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))
 
-    #define CHECK_LIMITS(x, mi, ma) ((x) < (mi)) ? (mi) : (((x) > (ma)) ? (ma) : (x))
+    #define CHECK_LIMITS(x, mi, ma) (((x) < (mi)) ? (mi) : (((x) > (ma)) ? (ma) : (x)))
 
     double seq, seek;
     
@@ -816,18 +812,18 @@ void TDStretch::precalcCorrReferenceMono()
 
 // Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo' 
 // version of the routine.
-void TDStretch::overlapStereo(short *output, const short *input) const
+void TDStretch::overlapStereo(short *poutput, const short *input) const
 {
     int i;
     short temp;
-    uint cnt2;
+    int cnt2;
 
-    for (i = 0; i < (int)overlapLength ; i ++) 
+    for (i = 0; i < overlapLength ; i ++) 
     {
         temp = (short)(overlapLength - i);
         cnt2 = 2 * i;
-        output[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp )  / overlapLength;
+        poutput[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp )  / overlapLength;
-        output[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
+        poutput[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
     }
 }
 
@@ -841,15 +837,15 @@ static int _getClosest2Power(double value)
 /// Calculates overlap period length in samples.
 /// Integer version rounds overlap length to closest power of 2
 /// for a divide scaling operation.
-void TDStretch::calculateOverlapLength(int overlapMs)
+void TDStretch::calculateOverlapLength(int aoverlapMs)
 {
     int newOvl;
 
-    assert(overlapMs >= 0);
+    assert(aoverlapMs >= 0);
-    overlapDividerBits = _getClosest2Power((sampleRate * overlapMs) / 1000.0);
+    overlapDividerBits = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0);
     if (overlapDividerBits > 9) overlapDividerBits = 9;
     if (overlapDividerBits < 4) overlapDividerBits = 4;
-    newOvl = (int)pow(2, (double) overlapDividerBits);
+    newOvl = (int)pow(2, overlapDividerBits);
 
     acceptNewOverlapLength(newOvl);
 

source/SoundTouch/TDStretch.h

@@ -44,6 +44,7 @@
 #ifndef TDStretch_H
 #define TDStretch_H
 
+#include <stddef.h>
 #include "STTypes.h"
 #include "RateTransposer.h"
 #include "FIFOSamplePipe.h"
@@ -113,7 +114,6 @@ protected:
     int overlapLength;
     int seekLength;
     int seekWindowLength;
-    int maxOffset;
     int overlapDividerBits;
     int slopingDivider;
     float nominalSkip;

source/SoundTouch/sse_optimized.cpp

@@ -73,8 +73,9 @@ using namespace soundtouch;
 double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
 {
     int i;
-    float *pVec1;
+    const float *pVec1;
-    __m128 vSum, *pVec2;
+    const __m128 *pVec2;
+    __m128 vSum;
 
     // Note. It means a major slow-down if the routine needs to tolerate 
     // unaligned __m128 memory accesses. It's way faster if we can skip 
@@ -104,8 +105,8 @@ double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) con
 
     // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
     // Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
-    pVec1 = (float*)pV1;
+    pVec1 = (const float*)pV1;
-    pVec2 = (__m128*)pV2;
+    pVec2 = (const __m128*)pV2;
     vSum = _mm_setzero_ps();
 
     // Unroll the loop by factor of 4 * 4 operations
@@ -300,13 +301,13 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
     // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
     for (j = 0; j < count; j += 2)
     {
-        float *pSrc;
+        const float *pSrc;
         const __m128 *pFil;
         __m128 sum1, sum2;
         uint i;
 
-        pSrc = (float*)source;              // source audio data
+        pSrc = (const float*)source;              // source audio data
-        pFil = (__m128*)filterCoeffsAlign;  // filter coefficients. NOTE: Assumes coefficients 
+        pFil = (const __m128*)filterCoeffsAlign;  // filter coefficients. NOTE: Assumes coefficients 
                                                   // are aligned to 16-byte boundary
         sum1 = sum2 = _mm_setzero_ps();