Azahar-Mirror/soundtouch
1
0
Fork 0
mirror of https://github.com/azahar-emu/soundtouch synced 2025-11-07 07:30:02 +01:00
Code Issues Packages Projects Releases Wiki Activity

Cleaned code with Lint

Browse Source
This commit is contained in:
oparviai 2009-02-21 16:00:14 +00:00
parent 8f880269e0
commit c17eb6821a
22 changed files with 146 additions and 172 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
include/BPMDetect.h
View File

@ -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).

2
include/FIFOSampleBuffer.h
View File

@ -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

4
include/FIFOSamplePipe.h
View File

@ -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();
}

6
source/SoundStretch/RunParameters.cpp
View File

@ -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;

2
source/SoundStretch/RunParameters.h
View File

@ -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

53
source/SoundStretch/WavFile.cpp
View File

@ -56,10 +56,10 @@
using namespace std;
const static char riffStr[] = "RIFF";
const static char waveStr[] = "WAVE";
const static char fmtStr[] = "fmt ";
const static char dataStr[] = "data";
static const char riffStr[] = "RIFF";
static const char waveStr[] = "WAVE";
static const char fmtStr[] = "fmt ";
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.sample_rate = sampleRate;
header.format.byte_rate = header.format.byte_per_sample * (int)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;

12
source/SoundStretch/WavFile.h
View File

@ -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

23
source/SoundStretch/main.cpp
View File

@ -91,9 +91,9 @@ static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParamet
}
// ... open output file with same sound parameters
bits = (*inFile)->getNumBits();
samplerate = (*inFile)->getSampleRate();
channels = (*inFile)->getNumChannels();
bits = (int)(*inFile)->getNumBits();
samplerate = (int)(*inFile)->getSampleRate();
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();
channels = inFile->getNumChannels();
sampleRate = (int)inFile->getSampleRate();
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());

8
source/SoundTouch/3dnow_win.cpp
View File

@ -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;
float corr;
int overlapLengthLocal = overlapLength;
float corr = 0;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
/*

65
source/SoundTouch/BPMDetect.cpp
View File

@ -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;
buffer = new FIFOSampleBuffer();
this->sampleRate = aSampleRate;
this->channels = numChannels;
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()
{

5
source/SoundTouch/FIFOSampleBuffer.cpp
View File

@ -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;

2
source/SoundTouch/FIRFilter.cpp
View File

@ -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];

3
source/SoundTouch/FIRFilter.h
View File

@ -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:

11
source/SoundTouch/PeakFinder.cpp
View File

@ -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->maxPos = maxPos;
this->minPos = aminPos;
this->maxPos = amaxPos;
// find absolute peak
peakpos = minPos;

2
source/SoundTouch/PeakFinder.h
View File

@ -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.

56
source/SoundTouch/RateTransposer.cpp
View File

@ -158,7 +158,7 @@ BOOL RateTransposer::isAAFilterEnabled() const
}
AAFilter *RateTransposer::getAAFilter() const
AAFilter *RateTransposer::getAAFilter()
{
return pAAFilter;
}
@ -548,19 +548,20 @@ uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src,
}
fSlopeCount -= 1.0f;
if (nSamples == 1) goto end;
while (1)
if (nSamples > 1)
{
while (fSlopeCount > 1.0f)
while (1)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
}
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
i++;
fSlopeCount += fRate;
}
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
i++;
fSlopeCount += fRate;
}
end:
// Store the last sample for the next round
@ -593,25 +594,26 @@ uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *sr
// now always (iSlopeCount > 1.0f)
fSlopeCount -= 1.0f;
if (nSamples == 1) goto end;
while (1)
if (nSamples > 1)
{
while (fSlopeCount > 1.0f)
while (1)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= nSamples - 1) goto end;
}
srcPos = 2 * used;
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos]
+ fSlopeCount * src[srcPos + 2]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1]
+ fSlopeCount * src[srcPos + 3]);
i++;
fSlopeCount += fRate;
}
srcPos = 2 * used;
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos]
+ fSlopeCount * src[srcPos + 2]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1]
+ fSlopeCount * src[srcPos + 3]);
i++;
fSlopeCount += fRate;
}
end:
// Store the last sample for the next round

5
source/SoundTouch/RateTransposer.h
View File

@ -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);

8
source/SoundTouch/SoundTouch.cpp
View File

@ -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);
pTDStretch->setChannels(numChannels);
pRateTransposer->setChannels((int)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);
}

3
source/SoundTouch/SoundTouch.vcproj
View File

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

26
source/SoundTouch/TDStretch.cpp
View File

@ -53,11 +53,7 @@
using namespace soundtouch;
#ifndef min
//#define min(a,b) (((a) > (b)) ? (b) : (a))
#define max(a,b) (((a) < (b)) ? (b) : (a))
#endif
#define max(x, y) (((x) > (y)) ? (x) : (y))
/*****************************************************************************
@ -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;
output[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
poutput[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * 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);
overlapDividerBits = _getClosest2Power((sampleRate * overlapMs) / 1000.0);
assert(aoverlapMs >= 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);

2
source/SoundTouch/TDStretch.h
View File

@ -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;

17
source/SoundTouch/sse_optimized.cpp
View File

@ -73,8 +73,9 @@ using namespace soundtouch;
double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
{
int i;
float *pVec1;
__m128 vSum, *pVec2;
const float *pVec1;
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;
pVec2 = (__m128*)pV2;
pVec1 = (const float*)pV1;
pVec2 = (const __m128*)pV2;
vSum = _mm_setzero_ps();
// Unroll the loop by factor of 4 * 4 operations
@ -300,14 +301,14 @@ 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
pFil = (__m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
pSrc = (const float*)source; // source audio data
pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
sum1 = sum2 = _mm_setzero_ps();
for (i = 0; i < length / 8; i ++)