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

Eliminated alloca() call that caused compatibility woes

Browse Source
This commit is contained in:
oparviai 2014-10-08 15:26:57 +00:00
parent bfc89b45a9
commit 126d1ac41d
3 changed files with 24 additions and 7 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

17
source/SoundTouch/FIRFilter.cpp
View File

@ -61,12 +61,15 @@ FIRFilter::FIRFilter()
length = 0; length = 0;
lengthDiv8 = 0; lengthDiv8 = 0;
filterCoeffs = NULL; filterCoeffs = NULL;
sum = NULL;
sumsize = 0;
} }
FIRFilter::~FIRFilter() FIRFilter::~FIRFilter()
{ {
delete[] filterCoeffs; delete[] filterCoeffs;
delete[] sum;
} }
// Usual C-version of the filter routine for stereo sound // Usual C-version of the filter routine for stereo sound
@ -167,10 +170,18 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
} }
uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
{ {
uint i, j, end, c; uint i, j, end, c;
LONG_SAMPLETYPE *sum=(LONG_SAMPLETYPE*)alloca(numChannels*sizeof(*sum));
if (sumsize < numChannels)
{
// allocate large enough array for keeping sums
sumsize = numChannels;
delete[] sum;
sum = new LONG_SAMPLETYPE[numChannels];
}
#ifdef SOUNDTOUCH_FLOAT_SAMPLES #ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider // when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying. // because division is much slower operation than multiplying.
@ -253,7 +264,7 @@ uint FIRFilter::getLength() const
// //
// Note : The amount of outputted samples is by value of 'filter_length' // Note : The amount of outputted samples is by value of 'filter_length'
// smaller than the amount of input samples. // smaller than the amount of input samples.
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
{ {
assert(length > 0); assert(length > 0);
assert(lengthDiv8 * 8 == length); assert(lengthDiv8 * 8 == length);

8
source/SoundTouch/FIRFilter.h
View File

@ -65,13 +65,17 @@ protected:
// Memory for filter coefficients // Memory for filter coefficients
SAMPLETYPE *filterCoeffs; SAMPLETYPE *filterCoeffs;
// Memory for keeping temporary sums in multichannel processing
LONG_SAMPLETYPE *sum;
uint sumsize;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest, virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
uint numSamples) const; uint numSamples) const;
virtual uint evaluateFilterMono(SAMPLETYPE *dest, virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
uint numSamples) const; uint numSamples) const;
virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const; virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
public: public:
FIRFilter(); FIRFilter();
@ -91,7 +95,7 @@ public:
uint evaluate(SAMPLETYPE *dest, uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
uint numSamples, uint numSamples,
uint numChannels) const; uint numChannels);
uint getLength() const; uint getLength() const;

6
source/SoundTouch/SoundTouch.cpp
View File

@ -348,8 +348,8 @@ void SoundTouch::flush()
int i; int i;
int nUnprocessed; int nUnprocessed;
int nOut; int nOut;
SAMPLETYPE *buff=(SAMPLETYPE*)alloca(64*channels*sizeof(SAMPLETYPE)); SAMPLETYPE *buff = new SAMPLETYPE[64 * channels];
// check how many samples still await processing, and scale // check how many samples still await processing, and scale
// that by tempo & rate to get expected output sample count // that by tempo & rate to get expected output sample count
nUnprocessed = numUnprocessedSamples(); nUnprocessed = numUnprocessedSamples();
@ -378,6 +378,8 @@ void SoundTouch::flush()
} }
} }
delete[] buff;
// Clear working buffers // Clear working buffers
pRateTransposer->clear(); pRateTransposer->clear();
pTDStretch->clearInput(); pTDStretch->clearInput();