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add spu synch method "P" featuring very refined code from PCSX2 which tempo-shifts intelligently.

This commit is contained in:
zeromus 2009-10-23 06:14:46 +00:00
parent a3fd06019e
commit 44a36b7950
45 changed files with 9560 additions and 599 deletions
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1
desmume/src/Makefile.am
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@ -47,6 +47,7 @@ libdesmume_a_SOURCES = \
addons/compactFlash.cpp addons/gbagame.cpp addons/none.cpp addons/rumblepak.cpp addons/guitarGrip.cpp addons/expMemory.cpp fs.h \
cheatSystem.cpp cheatSystem.h \
texcache.cpp texcache.h rasterize.cpp rasterize.h \
metaspu/metaspu.cpp \
version.h
if HAVE_ALSA

414
desmume/src/SPU.cpp
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@ -43,425 +43,25 @@
#include "NDSSystem.h"
#include "matrix.h"
#include "metaspu/metaspu.h"
#define K_ADPCM_LOOPING_RECOVERY_INDEX 99999
//#undef FORCEINLINE
//#define FORCEINLINE
class ISynchronizingAudioBuffer
{
public:
virtual void enqueue_samples(s16* buf, int samples_provided) = 0;
//returns the number of samples actually supplied, which may not match the number requested
virtual int output_samples(s16* buf, int samples_requested) = 0;
};
template<typename T> inline T _abs(T val)
{
if(val<0) return -val;
else return val;
}
template<typename T> inline T moveValueTowards(T val, T target, T incr)
{
incr = _abs(incr);
T delta = _abs(target-val);
if(val<target) val += incr;
else if(val>target) val -= incr;
T newDelta = _abs(target-val);
if(newDelta >= delta)
val = target;
return val;
}
class ZeromusSynchronizer : public ISynchronizingAudioBuffer
{
public:
ZeromusSynchronizer()
: mixqueue_go(false)
,
#ifdef NDEBUG
adjustobuf(200,1000)
#else
adjustobuf(22000,44000)
#endif
{
}
bool mixqueue_go;
virtual void enqueue_samples(s16* buf, int samples_provided)
{
for(int i=0;i<samples_provided;i++) {
s16 left = *buf++;
s16 right = *buf++;
adjustobuf.enqueue(left,right);
}
}
//returns the number of samples actually supplied, which may not match the number requested
virtual int output_samples(s16* buf, int samples_requested)
{
int done = 0;
if(!mixqueue_go) {
if(adjustobuf.size > 200)
mixqueue_go = true;
}
else
{
for(int i=0;i<samples_requested;i++) {
if(adjustobuf.size==0) {
mixqueue_go = false;
break;
}
done++;
s16 left, right;
adjustobuf.dequeue(left,right);
*buf++ = left;
*buf++ = right;
}
}
return done;
}
private:
class Adjustobuf
{
public:
Adjustobuf(int _minLatency, int _maxLatency)
: size(0)
, minLatency(_minLatency)
, maxLatency(_maxLatency)
{
rollingTotalSize = 0;
targetLatency = (maxLatency + minLatency)/2;
rate = 1.0f;
cursor = 0.0f;
curr[0] = curr[1] = 0;
kAverageSize = 80000;
}
float rate, cursor;
int minLatency, targetLatency, maxLatency;
std::queue<s16> buffer;
int size;
s16 curr[2];
std::queue<int> statsHistory;
void enqueue(s16 left, s16 right)
{
buffer.push(left);
buffer.push(right);
size++;
}
s64 rollingTotalSize;
u32 kAverageSize;
void addStatistic()
{
statsHistory.push(size);
rollingTotalSize += size;
if(statsHistory.size()>kAverageSize)
{
rollingTotalSize -= statsHistory.front();
statsHistory.pop();
float averageSize = (float)(rollingTotalSize / kAverageSize);
//static int ctr=0; ctr++; if((ctr&127)==0) printf("avg size: %f curr size: %d rate: %f\n",averageSize,size,rate);
{
float targetRate;
if(averageSize < targetLatency)
{
targetRate = 1.0f - (targetLatency-averageSize)/kAverageSize;
}
else if(averageSize > targetLatency) {
targetRate = 1.0f + (averageSize-targetLatency)/kAverageSize;
} else targetRate = 1.0f;
//rate = moveValueTowards(rate,targetRate,0.001f);
rate = targetRate;
}
}
}
void dequeue(s16& left, s16& right)
{
left = right = 0;
addStatistic();
if(size==0) { return; }
cursor += rate;
while(cursor>1.0f) {
cursor -= 1.0f;
if(size>0) {
curr[0] = buffer.front(); buffer.pop();
curr[1] = buffer.front(); buffer.pop();
size--;
}
}
left = curr[0];
right = curr[1];
}
} adjustobuf;
};
class NitsujaSynchronizer : public ISynchronizingAudioBuffer
{
private:
struct ssamp
{
s16 l, r;
ssamp() {}
ssamp(s16 ll, s16 rr) : l(ll), r(rr) {}
};
std::vector<ssamp> sampleQueue;
// returns values going between 0 and y-1 in a saw wave pattern, based on x
static FORCEINLINE int pingpong(int x, int y)
{
x %= 2*y;
if(x >= y)
x = 2*y - x - 1;
return x;
// in case we want to switch to odd buffer sizes for more sharpness
//x %= 2*(y-1);
//if(x >= y)
// x = 2*(y-1) - x;
//return x;
}
static FORCEINLINE ssamp crossfade (ssamp lhs, ssamp rhs, int cur, int start, int end)
{
if(cur <= start)
return lhs;
if(cur >= end)
return rhs;
// in case we want sine wave interpolation instead of linear here
//float ang = 3.14159f * (float)(cur - start) / (float)(end - start);
//cur = start + (int)((1-cosf(ang))*0.5f * (end - start));
int inNum = cur - start;
int outNum = end - cur;
int denom = end - start;
int lrv = ((int)lhs.l * outNum + (int)rhs.l * inNum) / denom;
int rrv = ((int)lhs.r * outNum + (int)rhs.r * inNum) / denom;
return ssamp(lrv,rrv);
}
static FORCEINLINE void emit_sample(s16*& outbuf, ssamp sample)
{
*outbuf++ = sample.l;
*outbuf++ = sample.r;
}
static FORCEINLINE void emit_samples(s16*& outbuf, const ssamp* samplebuf, int samples)
{
for(int i=0;i<samples;i++)
emit_sample(outbuf,samplebuf[i]);
}
public:
NitsujaSynchronizer()
{}
virtual void enqueue_samples(s16* buf, int samples_provided)
{
for(int i=0;i<samples_provided;i++)
{
sampleQueue.push_back(ssamp(buf[0],buf[1]));
buf += 2;
}
}
virtual int output_samples(s16* buf, int samples_requested)
{
int audiosize = samples_requested;
int queued = sampleQueue.size();
// truncate input and output sizes to multiples of 8 because I am too lazy to deal with odd numbers
audiosize &= ~7;
queued &= ~7;
if(queued > 0x200 && audiosize > 0) // is there any work to do?
{
// are we going at normal speed?
// or more precisely, are the input and output queues/buffers of similar size?
if(queued > 900 || audiosize > queued * 2)
{
// not normal speed. we have to resample it somehow in this case.
if(audiosize <= queued)
{
// fast forward speed
// this is the easy case, just crossfade it and it sounds ok
for(int i = 0; i < audiosize; i++)
{
int j = i + queued - audiosize;
ssamp outsamp = crossfade(sampleQueue[i],sampleQueue[j], i,0,audiosize);
emit_sample(buf,outsamp);
}
}
else
{
// slow motion speed
// here we take a very different approach,
// instead of crossfading it, we select a single sample from the queue
// and make sure that the index we use to select a sample is constantly moving
// and that it starts at the first sample in the queue and ends on the last one.
//
// hopefully the index doesn't move discontinuously or we'll get slight crackling
// (there might still be a minor bug here that causes this occasionally)
//
// here's a diagram of how the index we sample from moves:
//
// queued (this axis represents the index we sample from. the top means the end of the queue)
// ^
// | --> audiosize (this axis represents the output index we write to, right meaning forward in output time/position)
// | A C C end
// A A B C C C
// A A A B C C C
// A A A B C C
// A A C
// start
//
// yes, this means we are spending some stretches of time playing the sound backwards,
// but the stretches are short enough that this doesn't sound weird.
// this lets us avoid most crackling problems due to the endpoints matching up.
// TODO: it might help to calculate the approximate fundamental frequency
// and reduce either buffer size such that the reflections line up with it.
int midpointX = audiosize >> 1;
int midpointY = queued >> 1;
// all we need to do here is calculate the X position of the leftmost "B" in the above diagram.
// TODO: we should calculate it with a simple equation like
// midpointXOffset = min(something,somethingElse);
// but it's a little difficult to work it out exactly
// so here's a stupid search for the value for now:
int prevA = 999999;
int midpointXOffset = queued/2;
while(true)
{
int a = abs(pingpong(midpointX - midpointXOffset, queued) - midpointY) - midpointXOffset;
if(((a > 0) != (prevA > 0) || (a < 0) != (prevA < 0)) && prevA != 999999)
{
if((a + prevA)&1) // there's some sort of off-by-one problem with this search since we're moving diagonally...
midpointXOffset++; // but this fixes it most of the time...
break; // found it
}
prevA = a;
midpointXOffset--;
if(midpointXOffset < 0)
{
midpointXOffset = 0;
break; // failed to find it. the two sides probably meet exactly in the center.
}
}
int leftMidpointX = midpointX - midpointXOffset;
int rightMidpointX = midpointX + midpointXOffset;
int leftMidpointY = pingpong(leftMidpointX, queued);
int rightMidpointY = (queued-1) - pingpong((int)audiosize-1 - rightMidpointX + queued*2, queued);
// output the left almost-half of the sound (section "A")
for(int x = 0; x < leftMidpointX; x++)
{
int i = pingpong(x, queued);
emit_sample(buf,sampleQueue[i]);
}
// output the middle stretch (section "B")
int y = leftMidpointY;
int dyMidLeft = (leftMidpointY < midpointY) ? 1 : -1;
int dyMidRight = (rightMidpointY > midpointY) ? 1 : -1;
for(int x = leftMidpointX; x < midpointX; x++, y+=dyMidLeft)
emit_sample(buf,sampleQueue[y]);
for(int x = midpointX; x < rightMidpointX; x++, y+=dyMidRight)
emit_sample(buf,sampleQueue[y]);
// output the end of the queued sound (section "C")
for(int x = rightMidpointX; x < audiosize; x++)
{
int i = (queued-1) - pingpong((int)audiosize-1 - x + queued*2, queued);
emit_sample(buf,sampleQueue[i]);
}
} //end else
sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + queued);
return audiosize;
}
else
{
// normal speed
// just output the samples straightforwardly.
//
// at almost-full speeds (like 50/60 FPS)
// what will happen is that we rapidly fluctuate between entering this branch
// and entering the "slow motion speed" branch above.
// but that's ok! because all of these branches sound similar enough that we can get away with it.
// so the two cases actually complement each other.
if(audiosize >= queued)
{
emit_samples(buf,&sampleQueue[0],queued);
sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + queued);
return queued;
}
else
{
emit_samples(buf,&sampleQueue[0],audiosize);
sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin()+audiosize);
return audiosize;
}
} //end normal speed
} //end if there is any work to do
else
{
return 0;
}
} //output_samples
private:
}; //NitsujaSynchronizer
//static ISynchronizingAudioBuffer* synchronizer = new ZeromusSynchronizer();
static ISynchronizingAudioBuffer* synchronizer = new NitsujaSynchronizer();
//static ISynchronizingAudioBuffer* synchronizer = metaspu_construct(ESynchMethod_Z);
static ISynchronizingAudioBuffer* synchronizer = metaspu_construct(ESynchMethod_N);
SPU_struct *SPU_core = 0;
SPU_struct *SPU_user = 0;
int SPU_currentCoreNum = SNDCORE_DUMMY;
static int volume = 100;
enum ESynchMode
{
ESynchMode_DualSynchAsynch,
ESynchMode_Synchronous
};
static ESynchMode synchmode = ESynchMode_DualSynchAsynch;
enum ESynchMethod
{
ESynchMethod_N, //nitsuja's
ESynchMethod_Z //zero's
};
static ESynchMode synchmode = ESynchMode_DualSynchAsynch;
static ESynchMethod synchmethod = ESynchMethod_N;
static SoundInterface_struct *SNDCore=NULL;
@ -625,9 +225,7 @@ void SPU_SetSynchMode(int mode, int method)
synchmethod = (ESynchMethod)method;
delete synchronizer;
//grr does this need to be locked? spu might need a lock method
if(synchmethod == ESynchMethod_N)
synchronizer = new NitsujaSynchronizer();
else synchronizer = new ZeromusSynchronizer();
synchronizer = metaspu_construct(synchmethod);
}
}

3
desmume/src/metaspu/README Normal file
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@ -0,0 +1,3 @@
Much of the contents of this directory was taken from the SPU2-X plugin from PCSX2 (GPL)
This in turn makes use of the SoundTouch library (LGPL)
The main metaspu files were written for DeSmuME (GPL) and ported from there to other emulators.

388
desmume/src/metaspu/SndOut.cpp Normal file
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@ -0,0 +1,388 @@
/* SPU2-X, A plugin for Emulating the Sound Processing Unit of the Playstation 2
* Developed and maintained by the Pcsx2 Development Team.
*
* Original portions from SPU2ghz are (c) 2008 by David Quintana [gigaherz]
*
* SPU2-X is free software: you can redistribute it and/or modify it under the terms
* of the GNU Lesser General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* SPU2-X is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with SPU2-X. If not, see <http://www.gnu.org/licenses/>.
*/
//#include "Global.h"
#include "types.h"
#include "SndOut.h"
#include <assert.h>
//----------------
int SndOutLatencyMS = 160;
bool timeStretchDisabled = false;
//----------------
StereoOut32 StereoOut32::Empty( 0, 0 );
StereoOut32::StereoOut32( const StereoOut16& src ) :
Left( src.Left ),
Right( src.Right )
{
}
StereoOut32::StereoOut32( const StereoOutFloat& src ) :
Left( (s32)(src.Left * 2147483647.0f) ),
Right( (s32)(src.Right * 2147483647.0f) )
{
}
StereoOut16 StereoOut32::DownSample() const
{
return StereoOut16(
Left >> SndOutVolumeShift,
Right >> SndOutVolumeShift
);
}
StereoOut32 StereoOut16::UpSample() const
{
return StereoOut32(
Left << SndOutVolumeShift,
Right << SndOutVolumeShift
);
}
//class NullOutModule: public SndOutModule
//{
//public:
// s32 Init() { return 0; }
// void Close() { }
// s32 Test() const { return 0; }
// void Configure(uptr parent) { }
// bool Is51Out() const { return false; }
// int GetEmptySampleCount() const { return 0; }
//
// const wchar_t* GetIdent() const
// {
// return L"nullout";
// }
//
// const wchar_t* GetLongName() const
// {
// return L"No Sound (Emulate SPU2 only)";
// }
//
// void ReadSettings()
// {
// }
//
// void WriteSettings() const
// {
// }
//
//} NullOut;
//
//SndOutModule* mods[]=
//{
// &NullOut,
//#ifdef _MSC_VER
// XAudio2Out,
// DSoundOut,
// WaveOut,
//#endif
// NULL // signals the end of our list
//};
//
//int FindOutputModuleById( const wchar_t* omodid )
//{
// int modcnt = 0;
// while( mods[modcnt] != NULL )
// {
// if( wcscmp( mods[modcnt]->GetIdent(), omodid ) == 0 )
// break;
// ++modcnt;
// }
// return modcnt;
//}
StereoOut32 *SndBuffer::m_buffer;
s32 SndBuffer::m_size;
s32 SndBuffer::m_rpos;
s32 SndBuffer::m_wpos;
s32 SndBuffer::m_data;
bool SndBuffer::m_underrun_freeze;
StereoOut32* SndBuffer::sndTempBuffer = NULL;
StereoOut16* SndBuffer::sndTempBuffer16 = NULL;
int SndBuffer::sndTempProgress = 0;
int GetAlignedBufferSize( int comp )
{
return (comp + SndOutPacketSize-1) & ~(SndOutPacketSize-1);
}
// Returns TRUE if there is data to be output, or false if no data
// is available to be copied.
bool SndBuffer::CheckUnderrunStatus( int& nSamples, int& quietSampleCount )
{
quietSampleCount = 0;
if( m_underrun_freeze )
{
int toFill = (int)(m_size * ( timeStretchDisabled ? 0.50f : 0.1f ) );
toFill = GetAlignedBufferSize( toFill );
// toFill is now aligned to a SndOutPacket
if( m_data < toFill )
{
quietSampleCount = nSamples;
return false;
}
m_underrun_freeze = false;
//TODO
//if( MsgOverruns() )
printf(" * SPU2 > Underrun compensation (%d packets buffered)\n", toFill / SndOutPacketSize );
lastPct = 0.0; // normalize timestretcher
}
else if( m_data < nSamples )
{
nSamples = m_data;
quietSampleCount = SndOutPacketSize - m_data;
m_underrun_freeze = true;
if( !timeStretchDisabled )
timeStretchUnderrun();
return nSamples != 0;
}
return true;
}
void SndBuffer::_InitFail()
{
// If a failure occurs, just initialize the NoSound driver. This'll allow
// the game to emulate properly (hopefully), albeit without sound.
//OutputModule = FindOutputModuleById( NullOut.GetIdent() );
//mods[OutputModule]->Init();
}
void SndBuffer::_WriteSamples(StereoOut32 *bData, int nSamples)
{
int free = m_size-m_data;
m_predictData = 0;
assert( m_data <= m_size );
// Problem:
// If the SPU2 gets out of sync with the SndOut device, the writepos of the
// circular buffer will overtake the readpos, leading to a prolonged period
// of hopscotching read/write accesses (ie, lots of staticy crap sound for
// several seconds).
//
// Compromise:
// When an overrun occurs, we adapt by discarding a portion of the buffer.
// The older portion of the buffer is discarded rather than incoming data,
// so that the overall audio synchronization is better.
if( free < nSamples )
{
// Buffer overrun!
// Dump samples from the read portion of the buffer instead of dropping
// the newly written stuff.
s32 comp;
if( !timeStretchDisabled )
{
comp = timeStretchOverrun();
}
else
{
// Toss half the buffer plus whatever's being written anew:
comp = GetAlignedBufferSize( (m_size + nSamples ) / 2 );
if( comp > (m_size-SndOutPacketSize) ) comp = m_size-SndOutPacketSize;
}
m_data -= comp;
m_rpos = (m_rpos+comp) % m_size;
//TODO
//if( MsgOverruns() )
printf(" * SPU2 > Overrun Compensation (%d packets tossed)\n", comp / SndOutPacketSize );
lastPct = 0.0; // normalize the timestretcher
}
// copy in two phases, since there's a chance the packet
// wraps around the buffer (it'd be nice to deal in packets only, but
// the timestretcher and DSP options require flexibility).
const int endPos = m_wpos + nSamples;
const int secondCopyLen = endPos - m_size;
StereoOut32* wposbuffer = &m_buffer[m_wpos];
m_data += nSamples;
if( secondCopyLen > 0 )
{
nSamples -= secondCopyLen;
memcpy( m_buffer, &bData[nSamples], secondCopyLen * sizeof( *bData ) );
m_wpos = secondCopyLen;
}
else
m_wpos += nSamples;
memcpy( wposbuffer, bData, nSamples * sizeof( *bData ) );
}
void SndBuffer::Init()
{
//if( mods[OutputModule] == NULL )
//{
// _InitFail();
// return;
//}
// initialize sound buffer
// Buffer actually attempts to run ~50%, so allocate near double what
// the requested latency is:
m_rpos = 0;
m_wpos = 0;
m_data = 0;
try
{
const float latencyMS = SndOutLatencyMS * (timeStretchDisabled ? 1.5f : 2.0f );
m_size = GetAlignedBufferSize( (int)(latencyMS * SampleRate / 1000.0f ) );
m_buffer = new StereoOut32[m_size];
m_underrun_freeze = false;
sndTempBuffer = new StereoOut32[SndOutPacketSize];
sndTempBuffer16 = new StereoOut16[SndOutPacketSize];
}
catch( std::bad_alloc& )
{
// out of memory exception (most likely)
printf( "Out of memory error occurred while initializing SPU2." );
_InitFail();
return;
}
// clear buffers!
// Fixes loopy sounds on emu resets.
memset( sndTempBuffer, 0, sizeof(StereoOut32) * SndOutPacketSize );
memset( sndTempBuffer16, 0, sizeof(StereoOut16) * SndOutPacketSize );
sndTempProgress = 0;
soundtouchInit(); // initializes the timestretching
// some crap
//spdif_set51(mods[OutputModule]->Is51Out());
// initialize module
//if( mods[OutputModule]->Init() == -1 ) _InitFail();
}
void SndBuffer::Cleanup()
{
//mods[OutputModule]->Close();
soundtouchCleanup();
//safe_delete_array( m_buffer );
//safe_delete_array( sndTempBuffer );
//safe_delete_array( sndTempBuffer16 );
delete[] m_buffer;
delete[] sndTempBuffer;
delete[] sndTempBuffer16;
}
int SndBuffer::m_dsp_progress = 0;
int SndBuffer::m_timestretch_progress = 0;
int SndBuffer::ssFreeze = 0;
void SndBuffer::ClearContents()
{
SndBuffer::soundtouchClearContents();
SndBuffer::ssFreeze = 30; //Delays sound output for about half a second.
}
void SndBuffer::Write( const StereoOut32& Sample )
{
// Log final output to wavefile.
//WaveDump::WriteCore( 1, CoreSrc_External, Sample.DownSample() );
//RecordWrite( Sample.DownSample() );
//if(mods[OutputModule] == &NullOut) // null output doesn't need buffering or stretching! :p
// return;
sndTempBuffer[sndTempProgress++] = Sample;
// If we haven't accumulated a full packet yet, do nothing more:
if(sndTempProgress < SndOutPacketSize) return;
sndTempProgress = 0;
//Don't play anything directly after loading a savestate, avoids static killing your speakers.
// if ( ssFreeze > 0 )
// {
// ssFreeze--;
// return;
// }
//#ifndef __LINUX__
// else if( dspPluginEnabled )
// {
// // Convert in, send to winamp DSP, and convert out.
//
// for( int i=0; i<SndOutPacketSize; ++i ) { sndTempBuffer16[i] = sndTempBuffer[i].DownSample(); }
// m_dsp_progress += DspProcess( (s16*)sndTempBuffer16, SndOutPacketSize );
//
// // Some ugly code to ensure full packet handling:
// int ei = 0;
// while( m_dsp_progress >= SndOutPacketSize )
// {
// for( int i=0; i<SndOutPacketSize; ++i, ++ei ) { sndTempBuffer[i] = sndTempBuffer16[ei].UpSample(); }
//
// if( !timeStretchDisabled )
// timeStretchWrite();
// else
// _WriteSamples(sndTempBuffer, sndTempProgress);
//
// m_dsp_progress -= SndOutPacketSize;
// }
//
// // copy any leftovers to the front of the dsp buffer.
// if( m_dsp_progress > 0 )
// {
// memcpy( &sndTempBuffer16[ei], sndTempBuffer16,
// sizeof(sndTempBuffer16[0]) * m_dsp_progress
// );
// }
// }
//#endif
// else
{
if( !timeStretchDisabled )
timeStretchWrite();
else
_WriteSamples(sndTempBuffer, SndOutPacketSize);
}
}
s32 SndBuffer::Test()
{
//if( mods[OutputModule] == NULL )
// return -1;
//return mods[OutputModule]->Test();
return 0;
}

553
desmume/src/metaspu/SndOut.h Normal file
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@ -0,0 +1,553 @@
/* SPU2-X, A plugin for Emulating the Sound Processing Unit of the Playstation 2
* Developed and maintained by the Pcsx2 Development Team.
*
* Original portions from SPU2ghz are (c) 2008 by David Quintana [gigaherz]
*
* SPU2-X is free software: you can redistribute it and/or modify it under the terms
* of the GNU Lesser General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* SPU2-X is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with SPU2-X. If not, see <http://www.gnu.org/licenses/>.
*/
#pragma once
#include <algorithm>
struct StereoOut16;
struct StereoOut32;
struct StereoOutFloat;
struct StereoOut32
{
static StereoOut32 Empty;
s32 Left;
s32 Right;
StereoOut32() :
Left( 0 ),
Right( 0 )
{
}
StereoOut32( s32 left, s32 right ) :
Left( left ),
Right( right )
{
}
StereoOut32( const StereoOut16& src );
explicit StereoOut32( const StereoOutFloat& src );
StereoOut16 DownSample() const;
StereoOut32 operator+( const StereoOut32& right ) const
{
return StereoOut32(
Left + right.Left,
Right + right.Right
);
}
StereoOut32 operator/( int src ) const
{
return StereoOut32( Left / src, Right / src );
}
};
// Number of stereo samples per SndOut block.
// All drivers must work in units of this size when communicating with
// SndOut.
static const int SndOutPacketSize = 512;
// Overall master volume shift.
// Converts the mixer's 32 bit value into a 16 bit value.
//static const int SndOutVolumeShift = 13;
//edit - zeromus 23-oct-2009
//this is hardcoded differently for metaspu
static const int SndOutVolumeShift = 0;
// Samplerate of the SPU2. For accurate playback we need to match this
// exactly. Trying to scale samplerates and maintain SPU2's Ts timing accuracy
// is too problematic. :)
//this is hardcoded differently for metaspu
//edit - zeromus 23-oct-2009
//static const int SampleRate = 48000;
static const int SampleRate = 44100;
extern int FindOutputModuleById( const wchar_t* omodid );
struct StereoOut16
{
s16 Left;
s16 Right;
StereoOut16() :
Left( 0 ),
Right( 0 )
{
}
StereoOut16( const StereoOut32& src ) :
Left( (s16)src.Left ),
Right( (s16)src.Right )
{
}
StereoOut16( s16 left, s16 right ) :
Left( left ),
Right( right )
{
}
StereoOut32 UpSample() const;
void ResampleFrom( const StereoOut32& src )
{
// Use StereoOut32's built in conversion
*this = src.DownSample();
}
};
struct StereoOutFloat
{
float Left;
float Right;
StereoOutFloat() :
Left( 0 ),
Right( 0 )
{
}
explicit StereoOutFloat( const StereoOut32& src ) :
Left( src.Left / 2147483647.0f ),
Right( src.Right / 2147483647.0f )
{
}
explicit StereoOutFloat( s32 left, s32 right ) :
Left( left / 2147483647.0f ),
Right( right / 2147483647.0f )
{
}
StereoOutFloat( float left, float right ) :
Left( left ),
Right( right )
{
}
};
struct Stereo21Out16
{
s16 Left;
s16 Right;
s16 LFE;
void ResampleFrom( const StereoOut32& src )
{
Left = src.Left >> SndOutVolumeShift;
Right = src.Right >> SndOutVolumeShift;
LFE = (src.Left + src.Right) >> (SndOutVolumeShift + 1);
}
};
struct StereoQuadOut16
{
s16 Left;
s16 Right;
s16 LeftBack;
s16 RightBack;
void ResampleFrom( const StereoOut32& src )
{
Left = src.Left >> SndOutVolumeShift;
Right = src.Right >> SndOutVolumeShift;
LeftBack = src.Left >> SndOutVolumeShift;
RightBack = src.Right >> SndOutVolumeShift;
}
};
struct Stereo41Out16
{
s16 Left;
s16 Right;
s16 LFE;
s16 LeftBack;
s16 RightBack;
void ResampleFrom( const StereoOut32& src )
{
Left = src.Left >> SndOutVolumeShift;
Right = src.Right >> SndOutVolumeShift;
LFE = (src.Left + src.Right) >> (SndOutVolumeShift + 1);
LeftBack = src.Left >> SndOutVolumeShift;
RightBack = src.Right >> SndOutVolumeShift;
}
};
struct Stereo51Out16
{
s16 Left;
s16 Right;
s16 Center;
s16 LFE;
s16 LeftBack;
s16 RightBack;
// Implementation Note: Center and Subwoofer/LFE -->
// This method is simple and sounds nice. It relies on the speaker/soundcard
// systems do to their own low pass / crossover. Manual lowpass is wasted effort
// and can't match solid state results anyway.
void ResampleFrom( const StereoOut32& src )
{
Left = src.Left >> SndOutVolumeShift;
Right = src.Right >> SndOutVolumeShift;
Center = (src.Left + src.Right) >> (SndOutVolumeShift + 1);
LFE = Center;
LeftBack = src.Left >> SndOutVolumeShift;
RightBack = src.Right >> SndOutVolumeShift;
}
};
struct Stereo51Out16DplII
{
s16 Left;
s16 Right;
s16 Center;
s16 LFE;
s16 LeftBack;
s16 RightBack;
void ResampleFrom( const StereoOut32& src )
{
static const u8 sLogTable[256] = {
0x00,0x3C,0x60,0x78,0x8C,0x9C,0xA8,0xB4,0xBE,0xC8,0xD0,0xD8,0xDE,0xE4,0xEA,0xF0,
0xF6,0xFA,0xFE,0x04,0x08,0x0C,0x10,0x14,0x16,0x1A,0x1E,0x20,0x24,0x26,0x2A,0x2C,
0x2E,0x32,0x34,0x36,0x38,0x3A,0x3E,0x40,0x42,0x44,0x46,0x48,0x4A,0x4C,0x4E,0x50,
0x50,0x52,0x54,0x56,0x58,0x5A,0x5A,0x5C,0x5E,0x60,0x60,0x62,0x64,0x66,0x66,0x68,
0x6A,0x6A,0x6C,0x6E,0x6E,0x70,0x70,0x72,0x74,0x74,0x76,0x76,0x78,0x7A,0x7A,0x7C,
0x7C,0x7E,0x7E,0x80,0x80,0x82,0x82,0x84,0x84,0x86,0x86,0x88,0x88,0x8A,0x8A,0x8C,
0x8C,0x8C,0x8E,0x8E,0x90,0x90,0x92,0x92,0x92,0x94,0x94,0x96,0x96,0x96,0x98,0x98,
0x9A,0x9A,0x9A,0x9C,0x9C,0x9C,0x9E,0x9E,0xA0,0xA0,0xA0,0xA2,0xA2,0xA2,0xA4,0xA4,
0xA4,0xA6,0xA6,0xA6,0xA8,0xA8,0xA8,0xAA,0xAA,0xAA,0xAC,0xAC,0xAC,0xAC,0xAE,0xAE,
0xAE,0xB0,0xB0,0xB0,0xB2,0xB2,0xB2,0xB2,0xB4,0xB4,0xB4,0xB6,0xB6,0xB6,0xB6,0xB8,
0xB8,0xB8,0xB8,0xBA,0xBA,0xBA,0xBC,0xBC,0xBC,0xBC,0xBE,0xBE,0xBE,0xBE,0xC0,0xC0,
0xC0,0xC0,0xC2,0xC2,0xC2,0xC2,0xC2,0xC4,0xC4,0xC4,0xC4,0xC6,0xC6,0xC6,0xC6,0xC8,
0xC8,0xC8,0xC8,0xC8,0xCA,0xCA,0xCA,0xCA,0xCC,0xCC,0xCC,0xCC,0xCC,0xCE,0xCE,0xCE,
0xCE,0xCE,0xD0,0xD0,0xD0,0xD0,0xD0,0xD2,0xD2,0xD2,0xD2,0xD2,0xD4,0xD4,0xD4,0xD4,
0xD4,0xD6,0xD6,0xD6,0xD6,0xD6,0xD8,0xD8,0xD8,0xD8,0xD8,0xD8,0xDA,0xDA,0xDA,0xDA,
0xDA,0xDC,0xDC,0xDC,0xDC,0xDC,0xDC,0xDE,0xDE,0xDE,0xDE,0xDE,0xDE,0xE0,0xE0,0xE0,
};
static s32 Gfl=0,Gfr=0;
static s32 LMax=0,RMax=0;
static s32 LAccum;
static s32 RAccum;
static s32 ANum;
s32 ValL = src.Left >> (SndOutVolumeShift-8);
s32 ValR = src.Right >> (SndOutVolumeShift-8);
s32 XL = abs(ValL>>8);
s32 XR = abs(ValR>>8);
if(XL>LMax) LMax = XL;
if(XR>RMax) RMax = XR;
ANum++;
if(ANum>=128)
{
ANum=0;
LAccum = 1+((LAccum * 224 + LMax * 31)>>8);
RAccum = 1+((RAccum * 224 + RMax * 31)>>8);
LMax = 0;
RMax = 0;
s32 Tfl=(RAccum)*255/(LAccum);
s32 Tfr=(LAccum)*255/(RAccum);
int gMax = std::max(Tfl,Tfr);
Tfl = Tfl*255/gMax;
Tfr = Tfr*255/gMax;
if(Tfl>255) Tfl=255;
if(Tfr>255) Tfr=255;
if(Tfl<1) Tfl=1;
if(Tfr<1) Tfr=1;
Gfl = (Gfl * 200 + Tfl * 56)>>8;
Gfr = (Gfr * 200 + Tfr * 56)>>8;
}
s32 L,R,C,SUB,SL,SR;
C=(ValL+ValR)>>1; //16.8
ValL-=C;//16.8
ValR-=C;//16.8
L=ValL>>8; //16.0
R=ValR>>8; //16.0
C=C>>8; //16.0
SUB = C;
{
s32 Cfl = 1+sLogTable[Gfl];
s32 Cfr = 1+sLogTable[Gfr];
s32 VL=(ValL>>4) * Cfl; //16.12
s32 VR=(ValR>>4) * Cfr;
//s32 SC = (VL-VR)>>15;
SL = (((VR/148 - VL/209)>>4)*Cfr)>>8;
SR = (((VR/209 - VL/148)>>4)*Cfl)>>8;
}
// Random-ish values to get it to compile
int GainL = 200;
int GainR = 200;
int GainC = 180;
int GainSL = 230;
int GainSR = 230;
int GainLFE = 200;
int AddCLR = 55;
int AddCX = (C * AddCLR)>>8;
Left = (((L * GainL ))>>8) + AddCX;
Right = (((R * GainR ))>>8) + AddCX;
Center = (((C * GainC ))>>8);
LFE = (((SUB * GainLFE))>>8);
LeftBack = (((SL * GainSL ))>>8);
RightBack = (((SR * GainSR ))>>8);
}
};
struct Stereo71Out16
{
s16 Left;
s16 Right;
s16 Center;
s16 LFE;
s16 LeftBack;
s16 RightBack;
s16 LeftSide;
s16 RightSide;
void ResampleFrom( const StereoOut32& src )
{
Left = src.Left >> SndOutVolumeShift;
Right = src.Right >> SndOutVolumeShift;
Center = (src.Left + src.Right) >> (SndOutVolumeShift + 1);
LFE = Center;
LeftBack = src.Left >> SndOutVolumeShift;
RightBack = src.Right >> SndOutVolumeShift;
LeftSide = src.Left >> (SndOutVolumeShift+1);
RightSide = src.Right >> (SndOutVolumeShift+1);
}
};
struct Stereo21Out32
{
s32 Left;
s32 Right;
s32 LFE;
};
struct Stereo41Out32
{
s32 Left;
s32 Right;
s32 LFE;
s32 LeftBack;
s32 RightBack;
};
struct Stereo51Out32
{
s32 Left;
s32 Right;
s32 Center;
s32 LFE;
s32 LeftBack;
s32 RightBack;
};
// Developer Note: This is a static class only (all static members).
class SndBuffer
{
private:
static bool m_underrun_freeze;
static s32 m_predictData;
static float lastPct;
static StereoOut32* sndTempBuffer;
static StereoOut16* sndTempBuffer16;
static int sndTempProgress;
static int m_dsp_progress;
static int m_timestretch_progress;
static int m_timestretch_writepos;
static StereoOut32 *m_buffer;
static s32 m_size;
static s32 m_rpos;
static s32 m_wpos;
static s32 m_data;
static float lastEmergencyAdj;
static float cTempo;
static float eTempo;
static int freezeTempo;
static int ssFreeze;
static void _InitFail();
static void _WriteSamples(StereoOut32* bData, int nSamples);
static bool CheckUnderrunStatus( int& nSamples, int& quietSampleCount );
static void soundtouchInit();
static void soundtouchClearContents();
static void soundtouchCleanup();
static void timeStretchWrite();
static void timeStretchUnderrun();
static s32 timeStretchOverrun();
static void PredictDataWrite( int samples );
static float GetStatusPct();
static void UpdateTempoChange();
public:
static void Init();
static void Cleanup();
static void Write( const StereoOut32& Sample );
static s32 Test();
static void ClearContents();
// Note: When using with 32 bit output buffers, the user of this function is responsible
// for shifting the values to where they need to be manually. The fixed point depth of
// the sample output is determined by the SndOutVolumeShift, which is the number of bits
// to shift right to get a 16 bit result.
template< typename T >
static void ReadSamples( T* bData )
{
int nSamples = SndOutPacketSize;
// Problem:
// If the SPU2 gets even the least bit out of sync with the SndOut device,
// the readpos of the circular buffer will overtake the writepos,
// leading to a prolonged period of hopscotching read/write accesses (ie,
// lots of staticy crap sound for several seconds).
//
// Fix:
// If the read position overtakes the write position, abort the
// transfer immediately and force the SndOut driver to wait until
// the read buffer has filled up again before proceeding.
// This will cause one brief hiccup that can never exceed the user's
// set buffer length in duration.
int quietSamples;
if( CheckUnderrunStatus( nSamples, quietSamples ) )
{
assert( nSamples <= SndOutPacketSize );
// [Air] [TODO]: This loop is probably a candidate for SSE2 optimization.
const int endPos = m_rpos + nSamples;
const int secondCopyLen = endPos - m_size;
const StereoOut32* rposbuffer = &m_buffer[m_rpos];
m_data -= nSamples;
if( secondCopyLen > 0 )
{
nSamples -= secondCopyLen;
for( int i=0; i<secondCopyLen; i++ )
bData[nSamples+i].ResampleFrom( m_buffer[i] );
m_rpos = secondCopyLen;
}
else
m_rpos += nSamples;
for( int i=0; i<nSamples; i++ )
bData[i].ResampleFrom( rposbuffer[i] );
}
// If quietSamples != 0 it means we have an underrun...
// Let's just dull out some silence, because that's usually the least
// painful way of dealing with underruns:
memset( bData, 0, quietSamples * sizeof(T) );
}
};
//class SndOutModule
//{
//public:
// // Virtual destructor, because it helps fight C+++ funny-business.
// virtual ~SndOutModule() {}
//
// // Returns a unique identification string for this driver.
// // (usually just matches the driver's cpp filename)
// virtual const wchar_t* GetIdent() const=0;
//
// // Returns the long name / description for this driver.
// // (for use in configuration screen)
// virtual const wchar_t* GetLongName() const=0;
//
// virtual s32 Init()=0;
// virtual void Close()=0;
// virtual s32 Test() const=0;
//
// // Gui function: Used to open the configuration box for this driver.
// virtual void Configure(uptr parent)=0;
//
// // Loads settings from the INI file for this driver
// virtual void ReadSettings()=0;
//
// // Saves settings to the INI file for this driver
// virtual void WriteSettings() const=0;
//
// virtual bool Is51Out() const=0;
//
// // Returns the number of empty samples in the output buffer.
// // (which is effectively the amount of data played since the last update)
// virtual int GetEmptySampleCount() const=0;
//};
//
//
//#ifdef _MSC_VER
////internal
//extern SndOutModule* WaveOut;
//extern SndOutModule* DSoundOut;
//extern SndOutModule* XAudio2Out;
//#endif
//
//extern SndOutModule* mods[];
//
//// =====================================================================================================
//
//extern void RecordStart();
//extern void RecordStop();
//extern void RecordWrite( const StereoOut16& sample );
//
//extern s32 DspLoadLibrary(wchar_t *fileName, int modNum);
//extern void DspCloseLibrary();
//extern int DspProcess(s16 *buffer, int samples);
//extern void DspUpdate(); // to let the Dsp process window messages

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@ -0,0 +1,350 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Win32 version of the AMD 3DNow! optimized routines for AMD K6-2/Athlon
/// processors. All 3DNow! optimized functions have been gathered into this
/// single source code file, regardless to their class or original source code
/// file, in order to ease porting the library to other compiler and processor
/// platforms.
///
/// By the way; the performance gain depends heavily on the CPU generation: On
/// K6-2 these routines provided speed-up of even 2.4 times, while on Athlon the
/// difference to the original routines stayed at unremarkable 8%! Such a small
/// improvement on Athlon is due to 3DNow can perform only two operations in
/// parallel, and obviously also the Athlon FPU is doing a very good job with
/// the standard C floating point routines! Here these routines are anyway,
/// although it might not be worth the effort to convert these to GCC platform,
/// for Athlon CPU at least. The situation is different regarding the SSE
/// optimizations though, thanks to the four parallel operations of SSE that
/// already make a difference.
///
/// This file is to be compiled in Windows platform with Microsoft Visual C++
/// Compiler. Please see '3dnow_gcc.cpp' for the gcc compiler version for all
/// GNU platforms (if file supplied).
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support 3DNow! instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// perform a search with keywords "processor pack".
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.10 $
//
// $Id: 3dnow_win.cpp,v 1.10 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
#ifndef _WIN32
#error "wrong platform - this source code file is exclusively for Win32 platform"
#endif
using namespace soundtouch;
#ifdef ALLOW_3DNOW
// 3DNow! routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'TDStretch3DNow'
//
//////////////////////////////////////////////////////////////////////////////
#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;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
/*
c-pseudocode:
corr = 0;
for (i = 0; i < overlapLength / 4; i ++)
{
corr += pV1[0] * pV2[0];
pV1[1] * pV2[1];
pV1[2] * pV2[2];
pV1[3] * pV2[3];
pV1[4] * pV2[4];
pV1[5] * pV2[5];
pV1[6] * pV2[6];
pV1[7] * pV2[7];
pV1 += 8;
pV2 += 8;
}
*/
_asm
{
// give prefetch hints to CPU of what data are to be needed soonish.
// give more aggressive hints on pV1 as that changes more between different calls
// while pV2 stays the same.
prefetch [pV1]
prefetch [pV2]
prefetch [pV1 + 32]
mov eax, dword ptr pV2
mov ebx, dword ptr pV1
pxor mm0, mm0
mov ecx, overlapLengthLocal
shr ecx, 2 // div by four
loop1:
movq mm1, [eax]
prefetch [eax + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm1, [ebx]
prefetch [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
movq mm2, [eax + 8]
pfadd mm0, mm1
pfmul mm2, [ebx + 8]
movq mm3, [eax + 16]
pfadd mm0, mm2
pfmul mm3, [ebx + 16]
movq mm4, [eax + 24]
pfadd mm0, mm3
pfmul mm4, [ebx + 24]
add eax, 32
pfadd mm0, mm4
add ebx, 32
dec ecx
jnz loop1
// add halfs of mm0 together and return the result.
// note: mm1 is used as a dummy parameter only, we actually don't care about it's value
pfacc mm0, mm1
movd corr, mm0
femms
}
return corr;
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of 3DNow! optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilter3DNow::FIRFilter3DNow() : FIRFilter()
{
filterCoeffsUnalign = NULL;
}
FIRFilter3DNow::~FIRFilter3DNow()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for 3DNow! routine
void FIRFilter3DNow::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for 3DNow!
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((uint)filterCoeffsUnalign + 15) & -16);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// 3DNow!-optimized version of the filter routine for stereo sound
uint FIRFilter3DNow::evaluateFilterStereo(float *dest, const float *src, const uint numSamples) const
{
float *filterCoeffsLocal = filterCoeffsAlign;
uint count = (numSamples - length) & -2;
uint lengthLocal = length / 4;
assert(length != 0);
assert(count % 2 == 0);
/* original code:
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
filterCoeffsLocal = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * filterCoeffsLocal[0] +
ptr[2] * filterCoeffsLocal[2] +
ptr[4] * filterCoeffsLocal[4] +
ptr[6] * filterCoeffsLocal[6];
sumr1 += ptr[1] * filterCoeffsLocal[1] +
ptr[3] * filterCoeffsLocal[3] +
ptr[5] * filterCoeffsLocal[5] +
ptr[7] * filterCoeffsLocal[7];
suml2 += ptr[8] * filterCoeffsLocal[0] +
ptr[10] * filterCoeffsLocal[2] +
ptr[12] * filterCoeffsLocal[4] +
ptr[14] * filterCoeffsLocal[6];
sumr2 += ptr[9] * filterCoeffsLocal[1] +
ptr[11] * filterCoeffsLocal[3] +
ptr[13] * filterCoeffsLocal[5] +
ptr[15] * filterCoeffsLocal[7];
ptr += 16;
filterCoeffsLocal += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
_asm
{
mov eax, dword ptr dest
mov ebx, dword ptr src
mov edx, count
shr edx, 1
loop1:
// "outer loop" : during each round 2*2 output samples are calculated
prefetch [ebx] // give a prefetch hint to CPU what data are to be needed soonish
prefetch [filterCoeffsLocal] // give a prefetch hint to CPU what data are to be needed soonish
mov esi, ebx
mov edi, filterCoeffsLocal
pxor mm0, mm0
pxor mm1, mm1
mov ecx, lengthLocal
loop2:
// "inner loop" : during each round four FIR filter taps are evaluated for 2*2 output samples
movq mm2, [edi]
movq mm3, mm2
prefetch [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm2, [esi]
prefetch [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
pfmul mm3, [esi + 8]
movq mm4, [edi + 8]
movq mm5, mm4
pfadd mm0, mm2
pfmul mm4, [esi + 8]
pfadd mm1, mm3
pfmul mm5, [esi + 16]
movq mm2, [edi + 16]
movq mm6, mm2
pfadd mm0, mm4
pfmul mm2, [esi + 16]
pfadd mm1, mm5
pfmul mm6, [esi + 24]
movq mm3, [edi + 24]
movq mm7, mm3
pfadd mm0, mm2
pfmul mm3, [esi + 24]
pfadd mm1, mm6
pfmul mm7, [esi + 32]
add esi, 32
pfadd mm0, mm3
add edi, 32
pfadd mm1, mm7
dec ecx
jnz loop2
movq [eax], mm0
add ebx, 16
movq [eax + 8], mm1
add eax, 16
dec edx
jnz loop1
femms
}
return count;
}
#endif // ALLOW_3DNOW

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////////////////////////////////////////////////////////////////////////////////
///
/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
/// MMX optimization.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.9 $
//
// $Id: AAFilter.cpp,v 1.9 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "AAFilter.h"
#include "FIRFilter.h"
using namespace soundtouch;
#define PI 3.141592655357989
#define TWOPI (2 * PI)
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
*
*****************************************************************************/
AAFilter::AAFilter(const uint length)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(length);
}
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.
void AAFilter::setCutoffFreq(const double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
}
// Sets number of FIR filter taps
void AAFilter::setLength(const uint newLength)
{
length = newLength;
calculateCoeffs();
}
// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
uint i;
double cntTemp, temp, tempCoeff,h, w;
double fc2, wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;
assert(length > 0);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);
work = new double[length];
coeffs = new SAMPLETYPE[length];
fc2 = 2.0 * cutoffFreq;
wc = PI * fc2;
tempCoeff = TWOPI / (double)length;
sum = 0;
for (i = 0; i < length; i ++)
{
cntTemp = (double)i - (double)(length / 2);
temp = cntTemp * wc;
if (temp != 0)
{
h = fc2 * sin(temp) / temp; // sinc function
}
else
{
h = 1.0;
}
w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
temp = w * h;
work[i] = temp;
// calc net sum of coefficients
sum += temp;
}
// ensure the sum of coefficients is larger than zero
assert(sum > 0);
// ensure we've really designed a lowpass filter...
assert(work[length/2] > 0);
assert(work[length/2 + 1] > -1e-6);
assert(work[length/2 - 1] > -1e-6);
// Calculate a scaling coefficient in such a way that the result can be
// divided by 16384
scaleCoeff = 16384.0f / sum;
for (i = 0; i < length; i ++)
{
// scale & round to nearest integer
temp = work[i] * scaleCoeff;
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
coeffs[i] = (SAMPLETYPE)temp;
}
// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
pFIR->setCoefficients(coeffs, length, 14);
delete[] work;
delete[] coeffs;
}
// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter length'
// smaller than the amount of input samples.
uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
return pFIR->evaluate(dest, src, numSamples, numChannels);
}
uint AAFilter::getLength() const
{
return pFIR->getLength();
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.10 $
//
// $Id: AAFilter.h,v 1.10 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef AAFilter_H
#define AAFilter_H
#include "STTypes.h"
namespace soundtouch
{
class AAFilter
{
protected:
class FIRFilter *pFIR;
/// Low-pass filter cut-off frequency, negative = invalid
double cutoffFreq;
/// num of filter taps
uint length;
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(uint length);
~AAFilter();
/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
/// frequency (nyquist frequency = 0.5). The filter will cut off the
/// frequencies than that.
void setCutoffFreq(double newCutoffFreq);
/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);
uint getLength() const;
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.5 $
//
// $Id: BPMDetect.h,v 1.5 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _BPMDetect_H_
#define _BPMDetect_H_
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 230
/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
float envelopeAccu;
/// RMS volume sliding average approximation level accumulator
float RMSVolumeAccu;
/// Sample average counter.
int decimateCount;
/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;
/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;
/// Auto-correlation window length
int windowLen;
/// Number of channels (1 = mono, 2 = stereo)
int channels;
/// sample rate
int sampleRate;
/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
int windowStart;
/// 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).
void updateXCorr(int process_samples /// How many samples are processed.
);
/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);
/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);
/// Destructor.
virtual ~BPMDetect();
/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
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.
///
/// \return Beats-per-minute rate, or zero if detection failed.
float getBpm();
};
#endif // _BPMDetect_H_

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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// outputted samples from the buffer, as well as grows the buffer size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.11 $
//
// $Id: FIFOSampleBuffer.cpp,v 1.11 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdlib.h>
#include <memory.h>
#include <string.h>
#include <assert.h>
#include <stdexcept>
#include "FIFOSampleBuffer.h"
using namespace soundtouch;
// Constructor
FIFOSampleBuffer::FIFOSampleBuffer(uint numChannels)
{
sizeInBytes = 0; // reasonable initial value
buffer = NULL; //new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE)];
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = numChannels;
}
// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(const uint numChannels)
{
uint usedBytes;
usedBytes = channels * samplesInBuffer;
channels = numChannels;
samplesInBuffer = usedBytes / channels;
}
// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
}
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
memcpy(ptrEnd(numSamples), samples, sizeof(SAMPLETYPE) * numSamples * channels);
samplesInBuffer += numSamples;
}
// Increases the number of samples in the buffer without copying any actual
// samples.
//
// This function is used to update the number of samples in the sample buffer
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint numSamples)
{
uint req;
req = samplesInBuffer + numSamples;
ensureCapacity(req);
samplesInBuffer += numSamples;
}
// 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
// inserting new samples into the sample buffer directly. Please be careful!
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// succesfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
// to increase the sample count afterwards, by calling the
// 'putSamples(numSamples)' function.
SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
{
ensureCapacity(samplesInBuffer + slackCapacity);
return buffer + samplesInBuffer * channels;
}
// Returns a pointer to the beginning of the currently non-outputted samples.
// This function is provided for accessing the output samples directly.
// Please be careful!
//
// 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
{
return buffer + bufferPos * channels;
}
// Ensures that the buffer has enought capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size.
void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;
if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & -4096;
assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL)
{
throw std::runtime_error("Couldn't allocate memory!\n");
}
temp = (SAMPLETYPE *)(((ulongptr)tempUnaligned + 15) & -16);
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
{
// simply rewind the buffer (if necessary)
rewind();
}
}
// Returns the current buffer capacity in terms of samples
uint FIFOSampleBuffer::getCapacity() const
{
return sizeInBytes / (channels * sizeof(SAMPLETYPE));
}
// Returns the number of samples currently in the buffer
uint FIFOSampleBuffer::numSamples() const
{
return samplesInBuffer;
}
// Output samples from beginning of the sample buffer. Copies demanded number
// of samples to output and removes them from the sample buffer. If there
// are less than 'numsample' samples in the buffer, returns all available.
//
// Returns number of samples copied.
uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint num;
num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
return receiveSamples(num);
}
// Removes samples from the beginning of the sample buffer without copying them
// anywhere. Used to reduce the number of samples in the buffer, when accessing
// the sample buffer with the 'ptrBegin' function.
uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
{
if (maxSamples >= samplesInBuffer)
{
uint temp;
temp = samplesInBuffer;
samplesInBuffer = 0;
return temp;
}
samplesInBuffer -= maxSamples;
bufferPos += maxSamples;
return maxSamples;
}
// Returns nonzero if the sample buffer is empty
int FIFOSampleBuffer::isEmpty() const
{
return (samplesInBuffer == 0) ? 1 : 0;
}
// Clears the sample buffer
void FIFOSampleBuffer::clear()
{
samplesInBuffer = 0;
bufferPos = 0;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// output samples from the buffer as well as grows the storage size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.9 $
//
// $Id: FIFOSampleBuffer.h,v 1.9 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSampleBuffer_H
#define FIFOSampleBuffer_H
#include "FIFOSamplePipe.h"
namespace soundtouch
{
/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
/// care of storage size adjustment and data moving during input/output operations.
///
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
private:
/// Sample buffer.
SAMPLETYPE *buffer;
// Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
// 16-byte aligned location of this buffer
SAMPLETYPE *bufferUnaligned;
/// Sample buffer size in bytes
uint sizeInBytes;
/// How many samples are currently in buffer.
uint samplesInBuffer;
/// Channels, 1=mono, 2=stereo.
uint channels;
/// Current position pointer to the buffer. This pointer is increased when samples are
/// removed from the pipe so that it's necessary to actually rewind buffer (move data)
/// only new data when is put to the pipe.
uint bufferPos;
/// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
/// beginning of the buffer.
void rewind();
/// Ensures that the buffer has capacity for at least this many samples.
void ensureCapacity(const uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
public:
/// Constructor
FIFOSampleBuffer(uint numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
///< Default is stereo.
);
/// destructor
~FIFOSampleBuffer();
/// 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!
///
/// 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;
/// 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
/// inserting new samples into the sample buffer directly. Please be careful
/// not corrupt the book-keeping!
///
/// When using this function as means for inserting new samples, also remember
/// to increase the sample count afterwards, by calling the
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
);
/// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples.
///
/// This function is used to update the number of samples in the sample buffer
/// when accessing the buffer directly with 'ptrEnd' function. Please be
/// careful though!
virtual void putSamples(uint numSamples ///< Number of samples been inserted.
);
/// 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.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
);
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
);
/// Returns number of samples currently available.
virtual uint numSamples() const;
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(uint numChannels);
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;
/// Clears all the samples.
virtual void clear();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
/// samples by operating like a first-in-first-out pipe: New samples are fed
/// into one end of the pipe with the 'putSamples' function, and the processed
/// samples are received from the other end with the 'receiveSamples' function.
///
/// 'FIFOProcessor' : A base class for classes the do signal processing with
/// the samples while operating like a first-in-first-out pipe. When samples
/// are input with the 'putSamples' function, the class processes them
/// and moves the processed samples to the given 'output' pipe object, which
/// may be either another processing stage, or a fifo sample buffer object.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.8 $
//
// $Id: FIFOSamplePipe.h,v 1.8 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSamplePipe_H
#define FIFOSamplePipe_H
#include <assert.h>
#include <stdlib.h>
#include "STTypes.h"
namespace soundtouch
{
/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe
{
public:
/// 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!
///
/// 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;
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) = 0;
// Moves samples from the 'other' pipe instance to this instance.
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
)
{
int oNumSamples = other.numSamples();
putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples);
};
/// 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.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
) = 0;
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) = 0;
/// Returns number of samples currently available.
virtual uint numSamples() const = 0;
// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const = 0;
/// Clears all the samples.
virtual void clear() = 0;
};
/// 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
/// all the processing stages.
///
/// When samples are input to this class, they're first processed and then put to
/// the FIFO pipe that's defined as output of this class. This output pipe can be
/// either other processing stage or a FIFO sample buffer.
class FIFOProcessor :public FIFOSamplePipe
{
protected:
/// Internal pipe where processed samples are put.
FIFOSamplePipe *output;
/// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput)
{
assert(output == NULL);
assert(pOutput != NULL);
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = NULL;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
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!
///
/// 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
{
return output->ptrBegin();
}
public:
/// 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.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
)
{
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.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
)
{
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
{
return output->isEmpty();
}
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.16 $
//
// $Id: FIRFilter.cpp,v 1.16 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include <stdexcept>
#include "FIRFilter.h"
#include "cpu_detect.h"
using namespace soundtouch;
/*****************************************************************************
*
* Implementation of the class 'FIRFilter'
*
*****************************************************************************/
FIRFilter::FIRFilter()
{
resultDivFactor = 0;
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
}
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
{
uint i, j, end;
LONG_SAMPLETYPE suml, sumr;
#ifdef FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = 2 * (numSamples - length);
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
suml >>= resultDivFactor;
sumr >>= resultDivFactor;
// saturate to 16 bit integer limits
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
}
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE sum;
#ifdef FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = numSamples - length;
for (j = 0; j < end; j ++)
{
sum = 0;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
sum += src[i + 0] * filterCoeffs[i + 0] +
src[i + 1] * filterCoeffs[i + 1] +
src[i + 2] * filterCoeffs[i + 2] +
src[i + 3] * filterCoeffs[i + 3];
}
#ifdef INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
src ++;
}
return end;
}
// Set filter coeffiecients and length.
//
// Throws an exception if filter length isn't divisible by 8
void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
{
assert(newLength > 0);
if (newLength % 8) throw std::runtime_error("FIR filter length not divisible by 8");
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
#ifdef INTEGER_SAMPLES
resultDivider = (SAMPLETYPE)(1<<resultDivFactor);
#else
resultDivider = (SAMPLETYPE)powf(2, (SAMPLETYPE)resultDivFactor);
#endif
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
}
uint FIRFilter::getLength() const
{
return length;
}
// Applies the filter to the given sequence of samples.
//
// Note : The amount of outputted samples is by value of 'filter_length'
// smaller than the amount of input samples.
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
assert(numChannels == 1 || numChannels == 2);
assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
assert(resultDivFactor >= 0);
if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
} else {
return evaluateFilterMono(dest, src, numSamples);
}
}
// 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)
{
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
throw std::runtime_error("Don't use 'new FIRFilter', use 'newInstance' member instead!");
return NULL;
}
FIRFilter * FIRFilter::newInstance()
{
uint uExtensions = 0;
#if !defined(_MSC_VER) || !defined(__x86_64__)
uExtensions = detectCPUextensions();
#endif
// Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
#ifdef ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // ALLOW_MMX
#ifdef ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new FIRFilterSSE;
}
else
#endif // ALLOW_SSE
#ifdef ALLOW_3DNOW
if (uExtensions & SUPPORT_3DNOW)
{
// 3DNow! support
return ::new FIRFilter3DNow;
}
else
#endif // ALLOW_3DNOW
{
// ISA optimizations not supported, use plain C version
return ::new FIRFilter;
}
}

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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.17 $
//
// $Id: FIRFilter.h,v 1.17 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIRFilter_H
#define FIRFilter_H
#include "STTypes.h"
namespace soundtouch
{
class FIRFilter
{
protected:
// Number of FIR filter taps
uint length;
// Number of FIR filter taps divided by 8
uint lengthDiv8;
// Result divider factor in 2^k format
uint resultDivFactor;
// Result divider value.
SAMPLETYPE resultDivider;
// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
public:
FIRFilter();
virtual ~FIRFilter();
/// 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 * operator new(size_t s);
static FIRFilter *newInstance();
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter_length'
/// smaller than the amount of input samples.
///
/// \return Number of samples copied to 'dest'.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
uint getLength() const;
virtual void setCoefficients(const SAMPLETYPE *coeffs,
uint newLength,
uint uResultDivFactor);
};
// Optional subclasses that implement CPU-specific optimizations:
#ifdef ALLOW_MMX
/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
class FIRFilterMMX : public FIRFilter
{
protected:
short *filterCoeffsUnalign;
short *filterCoeffsAlign;
virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
public:
FIRFilterMMX();
~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_MMX
#ifdef ALLOW_3DNOW
/// Class that implements 3DNow! optimized functions exclusive for floating point samples type.
class FIRFilter3DNow : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilter3DNow();
~FIRFilter3DNow();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_3DNOW
#ifdef ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilterSSE();
~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // ALLOW_SSE
}
#endif // FIRFilter_H

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## Process this file with automake to create Makefile.in
##
## $Id: Makefile.am,v 1.3 2006/02/05 18:33:34 Olli Exp $
##
## Copyright (C) 2003 - David W. Durham
##
## This file is part of SoundTouch, an audio processing library for pitch/time adjustments
##
## SoundTouch is free software; you can redistribute it and/or modify it under the
## terms of the GNU General Public License as published by the Free Software
## Foundation; either version 2 of the License, or (at your option) any later
## version.
##
## SoundTouch is distributed in the hope that it will be useful, but WITHOUT ANY
## WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
## A PARTICULAR PURPOSE. See the GNU General Public License for more details.
##
## You should have received a copy of the GNU General Public License along with
## this program; if not, write to the Free Software Foundation, Inc., 59 Temple
## Place - Suite 330, Boston, MA 02111-1307, USA
AUTOMAKE_OPTIONS = foreign
noinst_HEADERS=AAFilter.h cpu_detect.h FIRFilter.h RateTransposer.h TDStretch.h cpu_detect_x86_gcc.cpp
noinst_LIBRARIES = libSoundTouch.a
libSoundTouch_a_CXXFLAGS = -msse -mmmx
libSoundTouch_a_CFLAGS = -msse -mmmx
#lib_LTLIBRARIES=libSoundTouch.la
# the mmx_gcc.cpp and cpu_detect_x86_gcc.cpp may need to be conditionally included here from things discovered in configure.ac
libSoundTouch_a_SOURCES=AAFilter.cpp FIRFilter.cpp FIFOSampleBuffer.cpp mmx_optimized.cpp sse_optimized.cpp \
RateTransposer.cpp SoundTouch.cpp TDStretch.cpp WavFile.cpp cpu_detect_x86_gcc.cpp
# ??? test for -fcheck-new in configure.ac
# other compiler flags to add
AM_CXXFLAGS=-O3 -msse -fcheck-new
#-I../../include
# other linking flags to add
#libSoundTouch_la_LIBADD=

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application)
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/03/19 10:05:49 $
// File revision : $Revision: 1.13 $
//
// $Id: RateTransposer.cpp,v 1.13 2006/03/19 10:05:49 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include <limits.h>
#include "RateTransposer.h"
#include "AAFilter.h"
using namespace soundtouch;
/// A linear samplerate transposer class that uses integer arithmetics.
/// for the transposing.
class RateTransposerInteger : public RateTransposer
{
protected:
int iSlopeCount;
uint uRate;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposerInteger();
virtual ~RateTransposerInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
};
/// A linear samplerate transposer class that uses floating point arithmetics
/// for the transposing.
class RateTransposerFloat : public RateTransposer
{
protected:
float fSlopeCount;
float fRateStep;
SAMPLETYPE sPrevSampleL, sPrevSampleR;
virtual void resetRegisters();
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposerFloat();
virtual ~RateTransposerFloat();
};
#ifndef min
#define min(a,b) ((a > b) ? b : a)
#define max(a,b) ((a < b) ? b : a)
#endif
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX/SSE/etc-capable CPU available or not.
void * RateTransposer::operator new(size_t s)
{
// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
assert(FALSE);
return NULL;
}
RateTransposer *RateTransposer::newInstance()
{
#ifdef INTEGER_SAMPLES
return ::new RateTransposerInteger;
#else
return ::new RateTransposerFloat;
#endif
}
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
uChannels = 2;
bUseAAFilter = TRUE;
// Instantiates the anti-alias filter with default tap length
// of 32
pAAFilter = new AAFilter(32);
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(const BOOL newMode)
{
bUseAAFilter = newMode;
}
/// Returns nonzero if anti-alias filter is enabled.
BOOL RateTransposer::isAAFilterEnabled() const
{
return bUseAAFilter;
}
AAFilter *RateTransposer::getAAFilter() const
{
return pAAFilter;
}
// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower
// uRate, larger faster uRates.
void RateTransposer::setRate(float newRate)
{
float fCutoff;
fRate = newRate;
// design a new anti-alias filter
if (newRate > 1.0f)
{
fCutoff = 0.5f / newRate;
}
else
{
fCutoff = 0.5f * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
// Outputs as many samples of the 'outputBuffer' as possible, and if there's
// any room left, outputs also as many of the incoming samples as possible.
// The goal is to drive the outputBuffer empty.
//
// It's allowed for 'output' and 'input' parameters to point to the same
// memory position.
void RateTransposer::flushStoreBuffer()
{
if (storeBuffer.isEmpty()) return;
outputBuffer.moveSamples(storeBuffer);
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
processSamples(samples, numSamples);
}
// Transposes up the sample rate, causing the observed playback 'rate' of the
// sound to decrease
void RateTransposer::upsample(const SAMPLETYPE *src, uint numSamples)
{
int count, sizeTemp, num;
// If the parameter 'uRate' value is smaller than 'SCALE', first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// First check that there's enough room in 'storeBuffer'
// (+16 is to reserve some slack in the destination buffer)
sizeTemp = (int)((float)numSamples / fRate + 16.0f);
// Transpose the samples, store the result into the end of "storeBuffer"
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, numSamples);
storeBuffer.putSamples(count);
// Apply the anti-alias filter to samples in "store output", output the
// result to "dest"
num = storeBuffer.numSamples();
count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
storeBuffer.ptrBegin(), num, uChannels);
outputBuffer.putSamples(count);
// Remove the processed samples from "storeBuffer"
storeBuffer.receiveSamples(count);
}
// Transposes down the sample rate, causing the observed playback 'rate' of the
// sound to increase
void RateTransposer::downsample(const SAMPLETYPE *src, uint numSamples)
{
int count, sizeTemp;
// If the parameter 'uRate' value is larger than 'SCALE', first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose. */
// Add the new samples to the end of the storeBuffer */
storeBuffer.putSamples(src, numSamples);
// Anti-alias filter the samples to prevent folding and output the filtered
// data to tempBuffer. Note : because of the FIR filter length, the
// filtering routine takes in 'filter_length' more samples than it outputs.
assert(tempBuffer.isEmpty());
sizeTemp = storeBuffer.numSamples();
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
storeBuffer.ptrBegin(), sizeTemp, uChannels);
// Remove the filtered samples from 'storeBuffer'
storeBuffer.receiveSamples(count);
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
sizeTemp = (int)((float)numSamples / fRate + 16.0f);
count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
outputBuffer.putSamples(count);
}
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint numSamples)
{
uint count;
uint sizeReq;
if (numSamples == 0) return;
assert(pAAFilter);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == FALSE)
{
sizeReq = (int)((float)numSamples / fRate + 1.0f);
count = transpose(outputBuffer.ptrEnd(sizeReq), src, numSamples);
outputBuffer.putSamples(count);
return;
}
// Transpose with anti-alias filter
if (fRate < 1.0f)
{
upsample(src, numSamples);
}
else
{
downsample(src, numSamples);
}
}
// Transposes the sample rate of the given samples using linear interpolation.
// Returns the number of samples returned in the "dest" buffer
inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
if (uChannels == 2)
{
return transposeStereo(dest, src, numSamples);
}
else
{
return transposeMono(dest, src, numSamples);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(const uint numchannels)
{
if (uChannels == numchannels) return;
assert(numchannels == 1 || numchannels == 2);
uChannels = numchannels;
storeBuffer.setChannels(uChannels);
tempBuffer.setChannels(uChannels);
outputBuffer.setChannels(uChannels);
// Inits the linear interpolation registers
resetRegisters();
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
storeBuffer.clear();
}
// Returns nonzero if there aren't any samples available for outputting.
uint RateTransposer::isEmpty()
{
int res;
res = FIFOProcessor::isEmpty();
if (res == 0) return 0;
return storeBuffer.isEmpty();
}
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerInteger - integer arithmetic implementation
//
/// fixed-point interpolation routine precision
#define SCALE 65536
// Constructor
RateTransposerInteger::RateTransposerInteger() : RateTransposer()
{
// call these here as these are virtual functions; calling these
// from the base class constructor wouldn't execute the overloaded
// versions (<master yoda>peculiar C++ can be</my>).
resetRegisters();
setRate(1.0f);
}
RateTransposerInteger::~RateTransposerInteger()
{
}
void RateTransposerInteger::resetRegisters()
{
iSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
unsigned int i, used;
LONG_SAMPLETYPE temp, vol1;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= numSamples - 1) goto end;
}
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[used] * vol1 + iSlopeCount * src[used + 1];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[numSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Stereo' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
unsigned int srcPos, i, used;
LONG_SAMPLETYPE temp, vol1;
if (numSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (iSlopeCount <= SCALE)
{
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = vol1 * sPrevSampleL + iSlopeCount * src[0];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = vol1 * sPrevSampleR + iSlopeCount * src[1];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
// now always (iSlopeCount > SCALE)
iSlopeCount -= SCALE;
while (1)
{
while (iSlopeCount > SCALE)
{
iSlopeCount -= SCALE;
used ++;
if (used >= numSamples - 1) goto end;
}
srcPos = 2 * used;
vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);
temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];
dest[2 * i] = (SAMPLETYPE)(temp / SCALE);
temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];
dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);
i++;
iSlopeCount += uRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * numSamples - 2];
sPrevSampleR = src[2 * numSamples - 1];
return i;
}
// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower
// uRate, larger faster uRates.
void RateTransposerInteger::setRate(float newRate)
{
uRate = (int)(newRate * SCALE + 0.5f);
RateTransposer::setRate(newRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// RateTransposerFloat - floating point arithmetic implementation
//
//////////////////////////////////////////////////////////////////////////////
// Constructor
RateTransposerFloat::RateTransposerFloat() : RateTransposer()
{
// call these here as these are virtual functions; calling these
// from the base class constructor wouldn't execute the overloaded
// versions (<master yoda>peculiar C++ can be</my>).
resetRegisters();
setRate(1.0f);
}
RateTransposerFloat::~RateTransposerFloat()
{
}
void RateTransposerFloat::resetRegisters()
{
fSlopeCount = 0;
sPrevSampleL =
sPrevSampleR = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
unsigned int i, used;
used = 0;
i = 0;
// Process the last sample saved from the previous call first...
while (fSlopeCount <= 1.0f)
{
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
i++;
fSlopeCount += fRate;
}
fSlopeCount -= 1.0f;
if (numSamples == 1) goto end;
while (1)
{
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= numSamples - 1) goto end;
}
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);
i++;
fSlopeCount += fRate;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[numSamples - 1];
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples)
{
unsigned int srcPos, i, used;
if (numSamples == 0) return 0; // no samples, no work
used = 0;
i = 0;
// Process the last sample saved from the sPrevSampleLious call first...
while (fSlopeCount <= 1.0f)
{
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);
i++;
fSlopeCount += fRate;
}
// now always (iSlopeCount > 1.0f)
fSlopeCount -= 1.0f;
if (numSamples == 1) goto end;
while (1)
{
while (fSlopeCount > 1.0f)
{
fSlopeCount -= 1.0f;
used ++;
if (used >= numSamples - 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;
}
end:
// Store the last sample for the next round
sPrevSampleL = src[2 * numSamples - 2];
sPrevSampleR = src[2 * numSamples - 1];
return i;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application).
///
/// Use either of the derived classes of 'RateTransposerInteger' or
/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
/// algorithm implementation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.10 $
//
// $Id: RateTransposer.h,v 1.10 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef RateTransposer_H
#define RateTransposer_H
#include "AAFilter.h"
#include "FIFOSamplePipe.h"
#include "FIFOSampleBuffer.h"
#include "STTypes.h"
namespace soundtouch
{
/// A common linear samplerate transposer class.
///
/// Note: Use function "RateTransposer::newInstance()" to create a new class
/// instance instead of the "new" operator; that function automatically
/// chooses a correct implementation depending on if integer or floating
/// arithmetics are to be used.
class RateTransposer : public FIFOProcessor
{
protected:
/// Anti-alias filter object
AAFilter *pAAFilter;
float fRate;
uint uChannels;
/// Buffer for collecting samples to feed the anti-alias filter between
/// two batches
FIFOSampleBuffer storeBuffer;
/// Buffer for keeping samples between transposing & anti-alias filter
FIFOSampleBuffer tempBuffer;
/// Output sample buffer
FIFOSampleBuffer outputBuffer;
BOOL bUseAAFilter;
void init();
virtual void resetRegisters() = 0;
virtual uint transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
virtual uint transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) = 0;
uint transpose(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples);
void flushStoreBuffer();
void downsample(const SAMPLETYPE *src,
uint numSamples);
void upsample(const SAMPLETYPE *src,
uint numSamples);
/// Transposes sample rate by applying anti-alias filter to prevent folding.
/// Returns amount of samples returned in the "dest" buffer.
/// The maximum amount of samples that can be returned at a time is set by
/// the 'set_returnBuffer_size' function.
void processSamples(const SAMPLETYPE *src,
uint numSamples);
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 arithmetics.
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 ot floating point arithmetics 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() const;
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(BOOL newMode);
/// Returns nonzero if anti-alias filter is enabled.
BOOL isAAFilterEnabled() const;
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(float newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
void putSamples(const SAMPLETYPE *samples, uint numSamples);
/// Clears all the samples in the object
void clear();
/// Returns nonzero if there aren't any samples available for outputting.
uint isEmpty();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Common type definitions for SoundTouch audio processing library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.16 $
//
// $Id: STTypes.h,v 1.16 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef STTypes_H
#define STTypes_H
//#define INTEGER_SAMPLES 1
typedef unsigned int uint;
typedef unsigned long ulong;
#ifdef __x86_64__
typedef unsigned long long ulongptr;
#else
typedef unsigned long ulongptr;
#endif
#ifdef __GNUC__
// In GCC, include soundtouch_config.h made by config scritps
/* Define to 1 if you have the <inttypes.h> header file. */
#define HAVE_INTTYPES_H 1
/* Define to 1 if you have the `m' library (-lm). */
#define HAVE_LIBM 1
/* Define to 1 if your system has a GNU libc compatible `malloc' function, and
to 0 otherwise. */
#define HAVE_MALLOC 1
/* Define to 1 if you have the <memory.h> header file. */
#define HAVE_MEMORY_H 1
/* Define to 1 if you have the <stdint.h> header file. */
#define HAVE_STDINT_H 1
/* Define to 1 if you have the <stdlib.h> header file. */
#define HAVE_STDLIB_H 1
/* Define to 1 if you have the <strings.h> header file. */
#define HAVE_STRINGS_H 1
/* Define to 1 if you have the <string.h> header file. */
#define HAVE_STRING_H 1
/* Define to 1 if you have the <sys/stat.h> header file. */
#define HAVE_SYS_STAT_H 1
/* Define to 1 if you have the <sys/types.h> header file. */
#define HAVE_SYS_TYPES_H 1
/* Define to 1 if you have the <unistd.h> header file. */
#define HAVE_UNISTD_H 1
/* Use Integer as Sample type */
//#define INTEGER_SAMPLES 1
/* Define as the return type of signal handlers (`int' or `void'). */
#define RETSIGTYPE void
/* Define to 1 if you have the ANSI C header files. */
#define STDC_HEADERS 1
#endif
#ifndef _WINDEF_
// if these aren't defined already by Windows headers, define now
typedef int BOOL;
#ifndef FALSE
#define FALSE 0
#endif
#ifndef TRUE
#define TRUE 1
#endif
#endif // _WINDEF_
namespace soundtouch
{
/// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file:
//#undef INTEGER_SAMPLES
//#undef FLOAT_SAMPLES
#if !(INTEGER_SAMPLES || FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
////
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
//#define INTEGER_SAMPLES 1 //< 16bit integer samples
#define FLOAT_SAMPLES 1 //< 32bit float samples
#endif
/// Define this to allow CPU-specific assembler optimizations. Notice that
/// having this enabled on non-x86 platforms doesn't matter; the compiler can
/// drop unsupported extensions on different platforms automatically.
/// However, if you're having difficulties getting the optimized routines
/// compiled with your compler (e.g. some gcc compiler versions may be picky),
/// you may wish to disable the optimizations to make the library compile.
#if !defined(_MSC_VER) || !defined(__x86_64__)
#define ALLOW_OPTIMIZATIONS 1
#define ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#endif
// If defined, allows the SIMD-optimized routines to take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
// calculations as in normal C implementation.
#ifdef INTEGER_SAMPLES
// 16bit integer sample type
typedef short SAMPLETYPE;
// data type for sample accumulation: Use 32bit integer to prevent overflows
typedef long LONG_SAMPLETYPE;
#ifdef FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // FLOAT_SAMPLES
#ifdef ALLOW_OPTIMIZATIONS
#if (_WIN32 || __i386__ || __x86_64__)
// Allow MMX optimizations
#define ALLOW_MMX 1
#endif
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision.
typedef double LONG_SAMPLETYPE;
#ifdef ALLOW_OPTIMIZATIONS
// Allow 3DNow! and SSE optimizations
#if _WIN32
// #define ALLOW_3DNOW 1
#endif
#if (_WIN32 || __i386__ || __x86_64__)
#define ALLOW_SSE 1
#endif
#endif
#endif // INTEGER_SAMPLES
};
#endif

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desmume/src/metaspu/SoundTouch/SoundTouch.cbp Normal file
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<?xml version="1.0" encoding="UTF-8" standalone="yes" ?>
<CodeBlocks_project_file>
<FileVersion major="1" minor="6" />
<Project>
<Option title="SoundTouch" />
<Option pch_mode="2" />
<Option compiler="gcc" />
<Build>
<Target title="Debug">
<Option output="../../deps/debug/libsoundtouch-dbg" prefix_auto="1" extension_auto="1" />
<Option working_dir="" />
<Option object_output="./.objs/debug" />
<Option type="2" />
<Option compiler="gcc" />
<Option createDefFile="1" />
<Compiler>
<Add option="-g" />
<Add option="-O0" />
</Compiler>
</Target>
<Target title="Release">
<Option output="../../deps/libsoundtouch" prefix_auto="1" extension_auto="1" />
<Option working_dir="" />
<Option object_output="./.objs/release" />
<Option type="2" />
<Option compiler="gcc" />
<Option createDefFile="1" />
<Compiler>
<Add option="-fexpensive-optimizations" />
<Add option="-O3" />
<Add option="-W" />
</Compiler>
<Linker>
<Add option="-s" />
</Linker>
</Target>
<Target title="Devel">
<Option output="../../deps/devel/libsoundtouch-dev" prefix_auto="1" extension_auto="1" />
<Option working_dir="" />
<Option object_output="./.objs/devel" />
<Option type="2" />
<Option compiler="gcc" />
<Option createDefFile="1" />
<Compiler>
<Add option="-O1" />
<Add option="-W" />
<Add option="-g" />
</Compiler>
</Target>
</Build>
<Compiler>
<Add option="-march=athlon-xp" />
<Add option="-march=prescott" />
</Compiler>
<Unit filename="AAFilter.cpp" />
<Unit filename="AAFilter.h" />
<Unit filename="BPMDetect.h" />
<Unit filename="FIFOSampleBuffer.cpp" />
<Unit filename="FIFOSampleBuffer.h" />
<Unit filename="FIFOSamplePipe.h" />
<Unit filename="FIRFilter.cpp" />
<Unit filename="FIRFilter.h" />
<Unit filename="RateTransposer.cpp" />
<Unit filename="RateTransposer.h" />
<Unit filename="STTypes.h" />
<Unit filename="SoundTouch.cpp" />
<Unit filename="SoundTouch.h" />
<Unit filename="TDStretch.cpp" />
<Unit filename="TDStretch.h" />
<Unit filename="WavFile.cpp" />
<Unit filename="WavFile.h" />
<Unit filename="cpu_detect.h" />
<Unit filename="cpu_detect_x86_gcc.cpp" />
<Unit filename="mmx_optimized.cpp" />
<Unit filename="sse_optimized.cpp" />
<Extensions>
<envvars />
<code_completion />
<lib_finder disable_auto="1" />
<debugger />
</Extensions>
</Project>
</CodeBlocks_project_file>

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desmume/src/metaspu/SoundTouch/SoundTouch.cpp Normal file
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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.13 $
//
// $Id: SoundTouch.cpp,v 1.13 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <assert.h>
#include <stdlib.h>
#include <memory.h>
#include <math.h>
#include <stdexcept>
#include <stdio.h>
#include "SoundTouch.h"
#include "TDStretch.h"
#include "RateTransposer.h"
#include "cpu_detect.h"
using namespace soundtouch;
/// Print library version string
extern "C" void soundtouch_ac_test()
{
printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
}
SoundTouch::SoundTouch()
{
// Initialize rate transposer and tempo changer instances
pRateTransposer = RateTransposer::newInstance();
pTDStretch = TDStretch::newInstance();
setOutPipe(pTDStretch);
rate = tempo = 0;
virtualPitch =
virtualRate =
virtualTempo = 1.0;
calcEffectiveRateAndTempo();
channels = 0;
bSrateSet = FALSE;
}
SoundTouch::~SoundTouch()
{
delete pRateTransposer;
delete pTDStretch;
}
/// Get SoundTouch library version string
const char *SoundTouch::getVersionString()
{
static const char *_version = SOUNDTOUCH_VERSION;
return _version;
}
/// Get SoundTouch library version Id
uint SoundTouch::getVersionId()
{
return SOUNDTOUCH_VERSION_ID;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (numChannels != 1 && numChannels != 2)
{
throw std::runtime_error("Illegal number of channels");
}
channels = numChannels;
pRateTransposer->setChannels(numChannels);
pTDStretch->setChannels(numChannels);
}
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
void SoundTouch::setRate(float newRate)
{
virtualRate = newRate;
calcEffectiveRateAndTempo();
}
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(float newRate)
{
virtualRate = 1.0f + 0.01f * newRate;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(float newTempo)
{
virtualTempo = newTempo;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(float newTempo)
{
virtualTempo = 1.0f + 0.01f * newTempo;
calcEffectiveRateAndTempo();
}
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(float newPitch)
{
virtualPitch = newPitch;
calcEffectiveRateAndTempo();
}
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(float newPitch)
{
virtualPitch = (float)exp(0.69314718056f * newPitch);
calcEffectiveRateAndTempo();
}
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch)
{
setPitchOctaves((float)newPitch / 12.0f);
}
void SoundTouch::setPitchSemiTones(float newPitch)
{
setPitchOctaves(newPitch / 12.0f);
}
// Calculates 'effective' rate and tempo values from the
// nominal control values.
void SoundTouch::calcEffectiveRateAndTempo()
{
float oldTempo = tempo;
float oldRate = rate;
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
if (rate != oldRate) pRateTransposer->setRate(rate);
if (tempo != oldTempo) pTDStretch->setTempo(tempo);
if (rate > 1.0f)
{
if (output != pRateTransposer)
{
FIFOSamplePipe *transOut;
assert(output == pTDStretch);
// move samples in the current output buffer to the output of pRateTransposer
transOut = pRateTransposer->getOutput();
transOut->moveSamples(*output);
// move samples in tempo changer's input to pitch transposer's input
pRateTransposer->moveSamples(*pTDStretch->getInput());
output = pRateTransposer;
}
}
else
{
if (output != pTDStretch)
{
FIFOSamplePipe *tempoOut;
assert(output == pRateTransposer);
// move samples in the current output buffer to the output of pTDStretch
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
}
}
// Sets sample rate.
void SoundTouch::setSampleRate(uint srate)
{
bSrateSet = TRUE;
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters(srate);
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
if (bSrateSet == FALSE)
{
throw std::runtime_error("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
{
throw std::runtime_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, numSamples);
}
*/
else if (rate <= 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
pRateTransposer->putSamples(samples, numSamples);
pTDStretch->moveSamples(*pRateTransposer);
}
else
{
assert(rate > 1.0f);
// evaluate the tempo changer, then transpose the rate up,
assert(output == pRateTransposer);
pTDStretch->putSamples(samples, numSamples);
pRateTransposer->moveSamples(*pTDStretch);
}
}
// Flushes the last samples from the processing pipeline to the output.
// Clears also the internal processing buffers.
//
// Note: This function is meant for extracting the last samples of a sound
// stream. This function may introduce additional blank samples in the end
// of the sound stream, and thus it's not recommended to call this function
// in the middle of a sound stream.
void SoundTouch::flush()
{
int i;
uint nOut;
SAMPLETYPE buff[128];
nOut = numSamples();
memset(buff, 0, 128 * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by
// feeding blank samples into the processing pipeline until new,
// processed samples appear in the output (not however, more than
// 8ksamples in any case)
for (i = 0; i < 128; i ++)
{
putSamples(buff, 64);
if (numSamples() != nOut) break; // new samples have appeared in the output!
}
// Clear working buffers
pRateTransposer->clear();
pTDStretch->clearInput();
// yet leave the 'tempoChanger' output intouched as that's where the
// flushed samples are!
}
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
BOOL SoundTouch::setSetting(uint settingId, uint value)
{
uint sampleRate, sequenceMs, seekWindowMs, overlapMs;
// read current tdstretch routine parameters
pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
switch (settingId)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);
return TRUE;
case SETTING_AA_FILTER_LENGTH :
// sets anti-alias filter length
pRateTransposer->getAAFilter()->setLength(value);
return TRUE;
case SETTING_USE_QUICKSEEK :
// enables / disables tempo routine quick seeking algorithm
pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);
return TRUE;
case SETTING_SEQUENCE_MS:
// change time-stretch sequence duration parameter
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
return TRUE;
case SETTING_SEEKWINDOW_MS:
// change time-stretch seek window length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
return TRUE;
case SETTING_OVERLAP_MS:
// change time-stretch overlap length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
return TRUE;
default :
return FALSE;
}
}
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// Returns the setting value.
uint SoundTouch::getSetting(uint settingId) const
{
uint temp;
switch (settingId)
{
case SETTING_USE_AA_FILTER :
return pRateTransposer->isAAFilterEnabled();
case SETTING_AA_FILTER_LENGTH :
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL);
return temp;
case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(NULL, NULL, &temp, NULL);
return temp;
case SETTING_OVERLAP_MS:
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
return temp;
default :
return 0;
}
}
// Clears all the samples in the object's output and internal processing
// buffers.
void SoundTouch::clear()
{
pRateTransposer->clear();
pTDStretch->clear();
}
/// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const
{
FIFOSamplePipe * psp;
if (pTDStretch)
{
psp = pTDStretch->getInput();
if (psp)
{
return psp->numSamples();
}
}
return 0;
}

252
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//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.14 $
//
// $Id: SoundTouch.h,v 1.14 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef SoundTouch_H
#define SoundTouch_H
#include "FIFOSamplePipe.h"
#include "STTypes.h"
namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.3.1"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID 010301
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
#define SETTING_USE_AA_FILTER 0
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
#define SETTING_AA_FILTER_LENGTH 1
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
#define SETTING_USE_QUICKSEEK 2
/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
/// to how long sequences the original sound is chopped in the time-stretch algorithm.
/// See "STTypes.h" or README for more information.
#define SETTING_SEQUENCE_MS 3
/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
/// best possible overlapping location. This determines from how wide window the algorithm
/// may look for an optimal joining location when mixing the sound sequences back together.
/// See "STTypes.h" or README for more information.
#define SETTING_SEEKWINDOW_MS 4
/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
/// are mixed back together, to form a continuous sound stream, this parameter defines over
/// how long period the two consecutive sequences are let to overlap each other.
/// See "STTypes.h" or README for more information.
#define SETTING_OVERLAP_MS 5
class SoundTouch : public FIFOProcessor
{
private:
/// Rate transposer class instance
class RateTransposer *pRateTransposer;
/// Time-stretch class instance
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
float virtualPitch;
/// Flag: Has sample rate been set?
BOOL bSrateSet;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
protected :
/// Number of channels
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
float tempo;
public:
SoundTouch();
virtual ~SoundTouch();
/// Get SoundTouch library version string
static const char *getVersionString();
/// Get SoundTouch library version Id
static uint getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(float newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(float newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(float newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(float newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(float newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(float newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(float newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Sets sample rate.
void setSampleRate(uint srate);
/// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
void flush();
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Pointer to sample buffer.
uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'TRUE' if the setting was succesfully changed
BOOL setSetting(uint settingId, ///< Setting ID number. see SETTING_... defines.
uint value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
uint getSetting(uint settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
///
/// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
/// - numSamples() : Get number of 'ready' samples that can be received with
/// function 'receiveSamples()'
/// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
/// - clear() : Clears all samples from ready/processing buffers.
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like
/// method with several performance-increasing tweaks.
///
/// Note : MMX optimized functions reside in a separate, platform-specific
/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.24 $
//
// $Id: TDStretch.cpp,v 1.24 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <string.h>
#include <stdlib.h>
#include <memory.h>
#include <limits.h>
#include <math.h>
#include <assert.h>
#include "STTypes.h"
#include "cpu_detect.h"
#include "TDStretch.h"
using namespace soundtouch;
#ifndef min
#define min(a,b) ((a > b) ? b : a)
#define max(a,b) ((a < b) ? b : a)
#endif
/*****************************************************************************
*
* Constant definitions
*
*****************************************************************************/
// Table for the hierarchical mixing position seeking algorithm
int scanOffsets[4][24]={
{ 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
{-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
/*****************************************************************************
*
* Implementation of the class 'TDStretch'
*
*****************************************************************************/
TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
{
bQuickseek = FALSE;
channels = 2;
bMidBufferDirty = FALSE;
pMidBuffer = NULL;
pRefMidBufferUnaligned = NULL;
overlapLength = 0;
setParameters(48000, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
setTempo(1.0f);
}
TDStretch::~TDStretch()
{
delete[] pMidBuffer;
delete[] pRefMidBufferUnaligned;
}
// Calculates the x having the closest 2^x value for the given value
static int _getClosest2Power(double value)
{
return (int)(log(value) / log(2.0) + 0.5);
}
// Sets routine control parameters. These control are certain time constants
// defining how the sound is stretched to the desired duration.
//
// 'sampleRate' = sample rate of the sound
// 'sequenceMS' = one processing sequence length in milliseconds (default = 82 ms)
// 'seekwindowMS' = seeking window length for scanning the best overlapping
// position (default = 28 ms)
// 'overlapMS' = overlapping length (default = 12 ms)
void TDStretch::setParameters(uint aSampleRate, uint aSequenceMS,
uint aSeekWindowMS, uint aOverlapMS)
{
this->sampleRate = aSampleRate;
this->sequenceMs = aSequenceMS;
this->seekWindowMs = aSeekWindowMS;
this->overlapMs = aOverlapMS;
seekLength = (sampleRate * seekWindowMs) / 1000;
seekWindowLength = (sampleRate * sequenceMs) / 1000;
maxOffset = seekLength;
calculateOverlapLength(overlapMs);
// set tempo to recalculate 'sampleReq'
setTempo(tempo);
}
/// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
/// value isn't returned.
void TDStretch::getParameters(uint *pSampleRate, uint *pSequenceMs, uint *pSeekWindowMs, uint *pOverlapMs)
{
if (pSampleRate)
{
*pSampleRate = sampleRate;
}
if (pSequenceMs)
{
*pSequenceMs = sequenceMs;
}
if (pSeekWindowMs)
{
*pSeekWindowMs = seekWindowMs;
}
if (pOverlapMs)
{
*pOverlapMs = overlapMs;
}
}
// Overlaps samples in 'midBuffer' with the samples in 'input'
void TDStretch::overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const
{
int i, itemp;
for (i = 0; i < (int)overlapLength ; i ++)
{
itemp = overlapLength - i;
output[i] = (input[i] * i + pMidBuffer[i] * itemp ) / overlapLength; // >> overlapDividerBits;
}
}
void TDStretch::clearMidBuffer()
{
if (bMidBufferDirty)
{
memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);
bMidBufferDirty = FALSE;
}
}
void TDStretch::clearInput()
{
inputBuffer.clear();
clearMidBuffer();
}
// Clears the sample buffers
void TDStretch::clear()
{
outputBuffer.clear();
inputBuffer.clear();
clearMidBuffer();
}
// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
// to enable
void TDStretch::enableQuickSeek(BOOL enable)
{
bQuickseek = enable;
}
// Returns nonzero if the quick seeking algorithm is enabled.
BOOL TDStretch::isQuickSeekEnabled() const
{
return bQuickseek;
}
// Seeks for the optimal overlap-mixing position.
uint TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
{
if (channels == 2)
{
// stereo sound
if (bQuickseek)
{
return seekBestOverlapPositionStereoQuick(refPos);
}
else
{
return seekBestOverlapPositionStereo(refPos);
}
}
else
{
// mono sound
if (bQuickseek)
{
return seekBestOverlapPositionMonoQuick(refPos);
}
else
{
return seekBestOverlapPositionMono(refPos);
}
}
}
// Overlaps samples in 'midBuffer' with the samples in 'inputBuffer' at position
// of 'ovlPos'.
inline void TDStretch::overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const
{
if (channels == 2)
{
// stereo sound
overlapStereo(output, input + 2 * ovlPos);
} else {
// mono sound.
overlapMono(output, input + ovlPos);
}
}
// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
uint TDStretch::seekBestOverlapPositionStereo(const SAMPLETYPE *refPos)
{
uint bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
uint i;
// Slopes the amplitudes of the 'midBuffer' samples
precalcCorrReferenceStereo();
bestCorr = INT_MIN;
bestOffs = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
for (i = 0; i < seekLength; i ++)
{
// Calculates correlation value for the mixing position corresponding
// to 'i'
corr = calcCrossCorrStereo(refPos + 2 * i, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = i;
}
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
uint TDStretch::seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos)
{
uint j;
uint bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
uint scanCount, corrOffset, tempOffset;
// Slopes the amplitude of the 'midBuffer' samples
precalcCorrReferenceStereo();
bestCorr = INT_MIN;
bestOffs = 0;
corrOffset = 0;
tempOffset = 0;
// Scans for the best correlation value using four-pass hierarchical search.
//
// The look-up table 'scans' has hierarchical position adjusting steps.
// In first pass the routine searhes for the highest correlation with
// relatively coarse steps, then rescans the neighbourhood of the highest
// correlation with better resolution and so on.
for (scanCount = 0;scanCount < 4; scanCount ++)
{
j = 0;
while (scanOffsets[scanCount][j])
{
tempOffset = corrOffset + scanOffsets[scanCount][j];
if (tempOffset >= seekLength) break;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = calcCrossCorrStereo(refPos + 2 * tempOffset, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
j ++;
}
corrOffset = bestOffs;
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
// Seeks for the optimal overlap-mixing position. The 'mono' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
uint TDStretch::seekBestOverlapPositionMono(const SAMPLETYPE *refPos)
{
uint bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
uint tempOffset;
const SAMPLETYPE *compare;
// Slopes the amplitude of the 'midBuffer' samples
precalcCorrReferenceMono();
bestCorr = INT_MIN;
bestOffs = 0;
// Scans for the best correlation value by testing each possible position
// over the permitted range.
for (tempOffset = 0; tempOffset < seekLength; tempOffset ++)
{
compare = refPos + tempOffset;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = calcCrossCorrMono(pRefMidBuffer, compare);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
// Seeks for the optimal overlap-mixing position. The 'mono' version of the
// routine
//
// The best position is determined as the position where the two overlapped
// sample sequences are 'most alike', in terms of the highest cross-correlation
// value over the overlapping period
uint TDStretch::seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos)
{
uint j;
uint bestOffs;
LONG_SAMPLETYPE bestCorr, corr;
uint scanCount, corrOffset, tempOffset;
// Slopes the amplitude of the 'midBuffer' samples
precalcCorrReferenceMono();
bestCorr = INT_MIN;
bestOffs = 0;
corrOffset = 0;
tempOffset = 0;
// Scans for the best correlation value using four-pass hierarchical search.
//
// The look-up table 'scans' has hierarchical position adjusting steps.
// In first pass the routine searhes for the highest correlation with
// relatively coarse steps, then rescans the neighbourhood of the highest
// correlation with better resolution and so on.
for (scanCount = 0;scanCount < 4; scanCount ++)
{
j = 0;
while (scanOffsets[scanCount][j])
{
tempOffset = corrOffset + scanOffsets[scanCount][j];
if (tempOffset >= seekLength) break;
// Calculates correlation value for the mixing position corresponding
// to 'tempOffset'
corr = calcCrossCorrMono(refPos + tempOffset, pRefMidBuffer);
// Checks for the highest correlation value
if (corr > bestCorr)
{
bestCorr = corr;
bestOffs = tempOffset;
}
j ++;
}
corrOffset = bestOffs;
}
// clear cross correlation routine state if necessary (is so e.g. in MMX routines).
clearCrossCorrState();
return bestOffs;
}
/// clear cross correlation routine state if necessary
void TDStretch::clearCrossCorrState()
{
// default implementation is empty.
}
// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
// tempo, larger faster tempo.
void TDStretch::setTempo(float newTempo)
{
uint intskip;
tempo = newTempo;
// Calculate ideal skip length (according to tempo value)
nominalSkip = tempo * (seekWindowLength - overlapLength);
skipFract = 0;
intskip = (int)(nominalSkip + 0.5f);
// Calculate how many samples are needed in the 'inputBuffer' to
// process another batch of samples
sampleReq = max(intskip + overlapLength, seekWindowLength) + maxOffset;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void TDStretch::setChannels(uint numChannels)
{
if (channels == numChannels) return;
assert(numChannels == 1 || numChannels == 2);
channels = numChannels;
inputBuffer.setChannels(channels);
outputBuffer.setChannels(channels);
}
// nominal tempo, no need for processing, just pass the samples through
// to outputBuffer
void TDStretch::processNominalTempo()
{
assert(tempo == 1.0f);
if (bMidBufferDirty)
{
// If there are samples in pMidBuffer waiting for overlapping,
// do a single sliding overlapping with them in order to prevent a
// clicking distortion in the output sound
if (inputBuffer.numSamples() < overlapLength)
{
// wait until we've got overlapLength input samples
return;
}
// Mix the samples in the beginning of 'inputBuffer' with the
// samples in 'midBuffer' using sliding overlapping
overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), 0);
outputBuffer.putSamples(overlapLength);
inputBuffer.receiveSamples(overlapLength);
clearMidBuffer();
// now we've caught the nominal sample flow and may switch to
// bypass mode
}
// Simply bypass samples from input to output
outputBuffer.moveSamples(inputBuffer);
}
// Processes as many processing frames of the samples 'inputBuffer', store
// the result into 'outputBuffer'
void TDStretch::processSamples()
{
uint ovlSkip, offset;
int temp;
/* Removed this small optimization - can introduce a click to sound when tempo setting
crosses the nominal value
if (tempo == 1.0f)
{
// tempo not changed from the original, so bypass the processing
processNominalTempo();
return;
}
*/
if (bMidBufferDirty == FALSE)
{
// if midBuffer is empty, move the first samples of the input stream
// into it
if (inputBuffer.numSamples() < overlapLength)
{
// wait until we've got overlapLength samples
return;
}
memcpy(pMidBuffer, inputBuffer.ptrBegin(), channels * overlapLength * sizeof(SAMPLETYPE));
inputBuffer.receiveSamples(overlapLength);
bMidBufferDirty = TRUE;
}
// Process samples as long as there are enough samples in 'inputBuffer'
// to form a processing frame.
while (inputBuffer.numSamples() >= sampleReq)
{
// If tempo differs from the normal ('SCALE'), scan for the best overlapping
// position
offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
// Mix the samples in the 'inputBuffer' at position of 'offset' with the
// samples in 'midBuffer' using sliding overlapping
// ... first partially overlap with the end of the previous sequence
// (that's in 'midBuffer')
overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), offset);
outputBuffer.putSamples(overlapLength);
// ... then copy sequence samples from 'inputBuffer' to output
temp = (seekWindowLength - 2 * overlapLength);// & 0xfffffffe;
if (temp > 0)
{
outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), temp);
}
// Copies the end of the current sequence from 'inputBuffer' to
// 'midBuffer' for being mixed with the beginning of the next
// processing sequence and so on
assert(offset + seekWindowLength <= inputBuffer.numSamples());
memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + seekWindowLength - overlapLength),
channels * sizeof(SAMPLETYPE) * overlapLength);
bMidBufferDirty = TRUE;
// Remove the processed samples from the input buffer. Update
// the difference between integer & nominal skip step to 'skipFract'
// in order to prevent the error from accumulating over time.
skipFract += nominalSkip; // real skip size
ovlSkip = (int)skipFract; // rounded to integer skip
skipFract -= ovlSkip; // maintain the fraction part, i.e. real vs. integer skip
inputBuffer.receiveSamples(ovlSkip);
}
}
// Adds 'numsamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void TDStretch::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
// Add the samples into the input buffer
inputBuffer.putSamples(samples, numSamples);
// Process the samples in input buffer
processSamples();
}
/// Set new overlap length parameter & reallocate RefMidBuffer if necessary.
void TDStretch::acceptNewOverlapLength(uint newOverlapLength)
{
uint prevOvl;
prevOvl = overlapLength;
overlapLength = newOverlapLength;
if (overlapLength > prevOvl)
{
delete[] pMidBuffer;
delete[] pRefMidBufferUnaligned;
pMidBuffer = new SAMPLETYPE[overlapLength * 2];
bMidBufferDirty = TRUE;
clearMidBuffer();
pRefMidBufferUnaligned = new SAMPLETYPE[2 * overlapLength + 16 / sizeof(SAMPLETYPE)];
// ensure that 'pRefMidBuffer' is aligned to 16 byte boundary for efficiency
pRefMidBuffer = (SAMPLETYPE *)((((ulongptr)pRefMidBufferUnaligned) + 15) & -16);
}
}
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX/SSE/etc-capable CPU available or not.
void * TDStretch::operator new(size_t s)
{
// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
assert(FALSE);
return NULL;
}
TDStretch * TDStretch::newInstance()
{
uint uExtensions = 0;
#if !defined(_MSC_VER) || !defined(__x86_64__)
uExtensions = detectCPUextensions();
#endif
// Check if MMX/SSE/3DNow! instruction set extensions supported by CPU
#ifdef ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new TDStretchMMX;
}
else
#endif // ALLOW_MMX
#ifdef ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new TDStretchSSE;
}
else
#endif // ALLOW_SSE
#ifdef ALLOW_3DNOW
if (uExtensions & SUPPORT_3DNOW)
{
// 3DNow! support
return ::new TDStretch3DNow;
}
else
#endif // ALLOW_3DNOW
{
// ISA optimizations not supported, use plain C version
return ::new TDStretch;
}
}
//////////////////////////////////////////////////////////////////////////////
//
// Integer arithmetics specific algorithm implementations.
//
//////////////////////////////////////////////////////////////////////////////
#ifdef INTEGER_SAMPLES
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceStereo()
{
int i, cnt2;
int temp, temp2;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = i * (overlapLength - i);
cnt2 = i * 2;
temp2 = (pMidBuffer[cnt2] * temp) / slopingDivider;
pRefMidBuffer[cnt2] = (short)(temp2);
temp2 = (pMidBuffer[cnt2 + 1] * temp) / slopingDivider;
pRefMidBuffer[cnt2 + 1] = (short)(temp2);
}
}
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceMono()
{
int i;
long temp;
long temp2;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = i * (overlapLength - i);
temp2 = (pMidBuffer[i] * temp) / slopingDivider;
pRefMidBuffer[i] = (short)temp2;
}
}
// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo'
// version of the routine.
void TDStretch::overlapStereo(short *output, const short *input) const
{
int i;
short temp;
uint cnt2;
for (i = 0; i < (int)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;
}
}
/// Calculates overlap period length in samples.
/// Integer version rounds overlap length to closest power of 2
/// for a divide scaling operation.
void TDStretch::calculateOverlapLength(uint overlapMs)
{
uint newOvl;
overlapDividerBits = _getClosest2Power((sampleRate * overlapMs) / 1000.0);
if (overlapDividerBits > 9) overlapDividerBits = 9;
if (overlapDividerBits < 4) overlapDividerBits = 4;
newOvl = 1<<overlapDividerBits;
acceptNewOverlapLength(newOvl);
// calculate sloping divider so that crosscorrelation operation won't
// overflow 32-bit register. Max. sum of the crosscorrelation sum without
// divider would be 2^30*(N^3-N)/3, where N = overlap length
slopingDivider = (newOvl * newOvl - 1) / 3;
}
long TDStretch::calcCrossCorrMono(const short *mixingPos, const short *compare) const
{
long corr;
uint i;
corr = 0;
for (i = 1; i < overlapLength; i ++)
{
corr += (mixingPos[i] * compare[i]) >> overlapDividerBits;
}
return corr;
}
long TDStretch::calcCrossCorrStereo(const short *mixingPos, const short *compare) const
{
long corr;
uint i;
corr = 0;
for (i = 2; i < 2 * overlapLength; i += 2)
{
corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;
}
return corr;
}
#endif // INTEGER_SAMPLES
//////////////////////////////////////////////////////////////////////////////
//
// Floating point arithmetics specific algorithm implementations.
//
#ifdef FLOAT_SAMPLES
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceStereo()
{
int i, cnt2;
float temp;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = (float)i * (float)(overlapLength - i);
cnt2 = i * 2;
pRefMidBuffer[cnt2] = (float)(pMidBuffer[cnt2] * temp);
pRefMidBuffer[cnt2 + 1] = (float)(pMidBuffer[cnt2 + 1] * temp);
}
}
// Slopes the amplitude of the 'midBuffer' samples so that cross correlation
// is faster to calculate
void TDStretch::precalcCorrReferenceMono()
{
int i;
float temp;
for (i=0 ; i < (int)overlapLength ;i ++)
{
temp = (float)i * (float)(overlapLength - i);
pRefMidBuffer[i] = (float)(pMidBuffer[i] * temp);
}
}
// SSE-optimized version of the function overlapStereo
void TDStretch::overlapStereo(float *output, const float *input) const
{
int i;
uint cnt2;
float fTemp;
float fScale;
float fi;
fScale = 1.0f / (float)overlapLength;
for (i = 0; i < (int)overlapLength ; i ++)
{
fTemp = (float)(overlapLength - i) * fScale;
fi = (float)i * fScale;
cnt2 = 2 * i;
output[cnt2 + 0] = input[cnt2 + 0] * fi + pMidBuffer[cnt2 + 0] * fTemp;
output[cnt2 + 1] = input[cnt2 + 1] * fi + pMidBuffer[cnt2 + 1] * fTemp;
}
}
/// Calculates overlap period length in samples.
void TDStretch::calculateOverlapLength(uint overlapMs)
{
uint newOvl;
newOvl = (sampleRate * overlapMs) / 1000;
if (newOvl < 16) newOvl = 16;
// must be divisible by 8
newOvl -= newOvl % 8;
acceptNewOverlapLength(newOvl);
}
double TDStretch::calcCrossCorrMono(const float *mixingPos, const float *compare) const
{
double corr;
uint i;
corr = 0;
for (i = 1; i < overlapLength; i ++)
{
corr += mixingPos[i] * compare[i];
}
return corr;
}
double TDStretch::calcCrossCorrStereo(const float *mixingPos, const float *compare) const
{
double corr;
uint i;
corr = 0;
for (i = 2; i < 2 * overlapLength; i += 2)
{
corr += mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1];
}
return corr;
}
#endif // FLOAT_SAMPLES

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desmume/src/metaspu/SoundTouch/TDStretch.h Normal file
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@ -0,0 +1,236 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.16 $
//
// $Id: TDStretch.h,v 1.16 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef TDStretch_H
#define TDStretch_H
#include "STTypes.h"
#include "RateTransposer.h"
#include "FIFOSamplePipe.h"
namespace soundtouch
{
// Default values for sound processing parameters:
/// Default length of a single processing sequence, in milliseconds. This determines to how
/// long sequences the original sound is chopped in the time-stretch algorithm.
///
/// The larger this value is, the lesser sequences are used in processing. In principle
/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
/// and vice versa.
///
/// Increasing this value reduces computational burden & vice versa.
#define DEFAULT_SEQUENCE_MS 63
#define DEFAULT_SEEKWINDOW_MS 17
#define DEFAULT_OVERLAP_MS 7
/// Class that does the time-stretch (tempo change) effect for the processed
/// sound.
class TDStretch : public FIFOProcessor
{
protected:
uint channels;
uint sampleReq;
float tempo;
SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pRefMidBuffer;
SAMPLETYPE *pRefMidBufferUnaligned;
uint overlapLength;
uint overlapDividerBits;
uint slopingDivider;
uint seekLength;
uint seekWindowLength;
uint maxOffset;
float nominalSkip;
float skipFract;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
BOOL bQuickseek;
BOOL bMidBufferDirty;
uint sampleRate;
uint sequenceMs;
uint seekWindowMs;
uint overlapMs;
void acceptNewOverlapLength(uint newOverlapLength);
virtual void clearCrossCorrState();
void calculateOverlapLength(uint overlapMs);
virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
virtual uint seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);
virtual uint seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);
virtual uint seekBestOverlapPositionMono(const SAMPLETYPE *refPos);
virtual uint seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);
uint seekBestOverlapPosition(const SAMPLETYPE *refPos);
virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
void clearMidBuffer();
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
void precalcCorrReferenceMono();
void precalcCorrReferenceStereo();
void processNominalTempo();
/// 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
/// the 'set_returnBuffer_size' function.
void processSamples();
public:
TDStretch();
virtual ~TDStretch();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX/SSE/etc-capable CPU available or not.
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 feature set depending on if the CPU
/// supports MMX/SSE/etc extensions.
static TDStretch *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the input buffer object
FIFOSamplePipe *getInput() { return &inputBuffer; };
/// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
/// tempo, larger faster tempo.
void setTempo(float newTempo);
/// Returns nonzero if there aren't any samples available for outputting.
virtual void clear();
/// Clears the input buffer
void clearInput();
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Enables/disables the quick position seeking algorithm. Zero to disable,
/// nonzero to enable
void enableQuickSeek(BOOL enable);
/// Returns nonzero if the quick seeking algorithm is enabled.
BOOL isQuickSeekEnabled() const;
/// Sets routine control parameters. These control are certain time constants
/// defining how the sound is stretched to the desired duration.
//
/// 'sampleRate' = sample rate of the sound
/// 'sequenceMS' = one processing sequence length in milliseconds
/// 'seekwindowMS' = seeking window length for scanning the best overlapping
/// position
/// 'overlapMS' = overlapping length
void setParameters(uint sampleRate, ///< Samplerate of sound being processed (Hz)
uint sequenceMS = DEFAULT_SEQUENCE_MS, ///< Single processing sequence length (ms)
uint seekwindowMS = DEFAULT_SEEKWINDOW_MS, ///< Offset seeking window length (ms)
uint overlapMS = DEFAULT_OVERLAP_MS ///< Sequence overlapping length (ms)
);
/// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter
/// value isn't returned.
void getParameters(uint *pSampleRate, uint *pSequenceMs, uint *pSeekWindowMs, uint *pOverlapMs);
/// Adds 'numsamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Input sample data
uint numSamples ///< Number of samples in 'samples' so that one sample
///< contains both channels if stereo
);
};
// Implementation-specific class declarations:
//#ifdef ALLOW_MMX
// /// Class that implements MMX optimized routines for 16bit integer samples type.
// class TDStretchMMX : public TDStretch
// {
// protected:
// long calcCrossCorrStereo(const short *mixingPos, const short *compare) const;
// virtual void overlapStereo(short *output, const short *input) const;
// virtual void clearCrossCorrState();
// };
//#endif /// ALLOW_MMX
//
//
//#ifdef ALLOW_3DNOW
// /// Class that implements 3DNow! optimized routines for floating point samples type.
// class TDStretch3DNow : public TDStretch
// {
// protected:
// double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
// };
//#endif /// ALLOW_3DNOW
#ifdef ALLOW_SSE
/// Class that implements SSE optimized routines for floating point samples type.
class TDStretchSSE : public TDStretch
{
protected:
double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;
};
#endif /// ALLOW_SSE
}
#endif /// TDStretch_H

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////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define _BIG_ENDIAN_ during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain,
/// but the reason for 'yet another' one is that those generic WAV reader
/// libraries are exhaustingly large and cumbersome! Wanted to have something
/// simpler here, i.e. something that's not already larger than rest of the
/// SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.15 $
//
// $Id: WavFile.cpp,v 1.15 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdio.h>
#include <stdexcept>
#include <string>
#include <assert.h>
#include <limits.h>
#include <cstdlib>
#include <cstring>
#include "WavFile.h"
using namespace std;
const static char riffStr[] = "RIFF";
const static char waveStr[] = "WAVE";
const static char fmtStr[] = "fmt ";
const static char dataStr[] = "data";
//////////////////////////////////////////////////////////////////////////////
//
// Helper functions for swapping byte order to correctly read/write WAV files
// with big-endian CPU's: Define compile-time definition _BIG_ENDIAN_ to
// turn-on the conversion if it appears necessary.
//
// For example, Intel x86 is little-endian and doesn't require conversion,
// while PowerPC of Mac's and many other RISC cpu's are big-endian.
#ifdef BYTE_ORDER
// In gcc compiler detect the byte order automatically
#if BYTE_ORDER == BIG_ENDIAN
// big-endian platform.
#define _BIG_ENDIAN_
#endif
#endif
#ifdef _BIG_ENDIAN_
// big-endian CPU, swap bytes in 16 & 32 bit words
// helper-function to swap byte-order of 32bit integer
static inline void _swap32(unsigned int &dwData)
{
dwData = ((dwData >> 24) & 0x000000FF) |
((dwData >> 8) & 0x0000FF00) |
((dwData << 8) & 0x00FF0000) |
((dwData << 24) & 0xFF000000);
}
// helper-function to swap byte-order of 16bit integer
static inline void _swap16(unsigned short &wData)
{
wData = ((wData >> 8) & 0x00FF) |
((wData << 8) & 0xFF00);
}
// helper-function to swap byte-order of buffer of 16bit integers
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumWords)
{
unsigned long i;
for (i = 0; i < dwNumWords; i ++)
{
_swap16(pData[i]);
}
}
#else // BIG_ENDIAN
// little-endian CPU, WAV file is ok as such
// dummy helper-function
static inline void _swap32(unsigned int &dwData)
{
// do nothing
}
// dummy helper-function
static inline void _swap16(unsigned short &wData)
{
// do nothing
}
// dummy helper-function
static inline void _swap16Buffer(unsigned short *pData, unsigned int dwNumBytes)
{
// do nothing
}
#endif // BIG_ENDIAN
//////////////////////////////////////////////////////////////////////////////
//
// Class WavInFile
//
WavInFile::WavInFile(const char *fileName)
{
int hdrsOk;
// Try to open the file for reading
fptr = fopen(fileName, "rb");
if (fptr == NULL)
{
// didn't succeed
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for reading.";
throw runtime_error(msg);
}
// Read the file headers
hdrsOk = readWavHeaders();
if (hdrsOk != 0)
{
// Something didn't match in the wav file headers
string msg = "File \"";
msg += fileName;
msg += "\" is corrupt or not a WAV file";
throw runtime_error(msg);
}
if (header.format.fixed != 1)
{
string msg = "File \"";
msg += fileName;
msg += "\" uses unsupported encoding.";
throw runtime_error(msg);
}
dataRead = 0;
}
WavInFile::~WavInFile()
{
close();
}
void WavInFile::rewind()
{
int hdrsOk;
fseek(fptr, 0, SEEK_SET);
hdrsOk = readWavHeaders();
assert(hdrsOk == 0);
dataRead = 0;
}
int WavInFile::checkCharTags()
{
// header.format.fmt should equal to 'fmt '
if (memcmp(fmtStr, header.format.fmt, 4) != 0) return -1;
// header.data.data_field should equal to 'data'
if (memcmp(dataStr, header.data.data_field, 4) != 0) return -1;
return 0;
}
int WavInFile::read(char *buffer, int maxElems)
{
int numBytes;
uint afterDataRead;
// ensure it's 8 bit format
if (header.format.bits_per_sample != 8)
{
throw runtime_error("Error: WavInFile::read(char*, int) works only with 8bit samples.");
}
assert(sizeof(char) == 1);
numBytes = maxElems;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = header.data.data_len - dataRead;
assert(numBytes >= 0);
}
numBytes = fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
return numBytes;
}
int WavInFile::read(short *buffer, int maxElems)
{
unsigned int afterDataRead;
int numBytes;
int numElems;
if (header.format.bits_per_sample == 8)
{
// 8 bit format
char *temp = new char[maxElems];
int i;
numElems = read(temp, maxElems);
// convert from 8 to 16 bit
for (i = 0; i < numElems; i ++)
{
buffer[i] = temp[i] << 8;
}
delete[] temp;
}
else
{
// 16 bit format
assert(header.format.bits_per_sample == 16);
assert(sizeof(short) == 2);
numBytes = maxElems * 2;
afterDataRead = dataRead + numBytes;
if (afterDataRead > header.data.data_len)
{
// Don't read more samples than are marked available in header
numBytes = header.data.data_len - dataRead;
assert(numBytes >= 0);
}
numBytes = fread(buffer, 1, numBytes, fptr);
dataRead += numBytes;
numElems = numBytes / 2;
// 16bit samples, swap byte order if necessary
_swap16Buffer((unsigned short *)buffer, numElems);
}
return numElems;
}
int WavInFile::read(float *buffer, int maxElems)
{
short *temp = new short[maxElems];
int num;
int i;
double fscale;
num = read(temp, maxElems);
fscale = 1.0 / 32768.0;
// convert to floats, scale to range [-1..+1[
for (i = 0; i < num; i ++)
{
buffer[i] = (float)(fscale * (double)temp[i]);
}
delete[] temp;
return num;
}
int WavInFile::eof() const
{
// return true if all data has been read or file eof has reached
return (dataRead == header.data.data_len || feof(fptr));
}
void WavInFile::close()
{
fclose(fptr);
fptr = NULL;
}
// test if character code is between a white space ' ' and little 'z'
static int isAlpha(char c)
{
return (c >= ' ' && c <= 'z') ? 1 : 0;
}
// test if all characters are between a white space ' ' and little 'z'
static int isAlphaStr(char *str)
{
int c;
c = str[0];
while (c)
{
if (isAlpha(c) == 0) return 0;
str ++;
c = str[0];
}
return 1;
}
int WavInFile::readRIFFBlock()
{
fread(&(header.riff), sizeof(WavRiff), 1, fptr);
// swap 32bit data byte order if necessary
_swap32((unsigned int &)header.riff.package_len);
// header.riff.riff_char should equal to 'RIFF');
if (memcmp(riffStr, header.riff.riff_char, 4) != 0) return -1;
// header.riff.wave should equal to 'WAVE'
if (memcmp(waveStr, header.riff.wave, 4) != 0) return -1;
return 0;
}
int WavInFile::readHeaderBlock()
{
char label[5];
string sLabel;
// lead label string
fread(label, 1, 4, fptr);
label[4] = 0;
if (isAlphaStr(label) == 0) return -1; // not a valid label
// Decode blocks according to their label
if (strcmp(label, fmtStr) == 0)
{
int nLen, nDump;
// 'fmt ' block
memcpy(header.format.fmt, fmtStr, 4);
// read length of the format field
fread(&nLen, sizeof(int), 1, fptr);
// swap byte order if necessary
_swap32((unsigned int &)nLen); // int format_len;
header.format.format_len = nLen;
// calculate how much length differs from expected
nDump = nLen - (sizeof(header.format) - 8);
// if format_len is larger than expected, read only as much data as we've space for
if (nDump > 0)
{
nLen = sizeof(header.format) - 8;
}
// read data
fread(&(header.format.fixed), nLen, 1, fptr);
// swap byte order if necessary
_swap16((unsigned short &)header.format.fixed); // short int fixed;
_swap16((unsigned short &)header.format.channel_number); // short int channel_number;
_swap32((unsigned int &)header.format.sample_rate); // int sample_rate;
_swap32((unsigned int &)header.format.byte_rate); // int byte_rate;
_swap16((unsigned short &)header.format.byte_per_sample); // short int byte_per_sample;
_swap16((unsigned short &)header.format.bits_per_sample); // short int bits_per_sample;
// if format_len is larger than expected, skip the extra data
if (nDump > 0)
{
fseek(fptr, nDump, SEEK_CUR);
}
return 0;
}
else if (strcmp(label, dataStr) == 0)
{
// 'data' block
memcpy(header.data.data_field, dataStr, 4);
fread(&(header.data.data_len), sizeof(uint), 1, fptr);
// swap byte order if necessary
_swap32((unsigned int &)header.data.data_len);
return 1;
}
else
{
uint len, i;
uint temp;
// unknown block
// read length
fread(&len, sizeof(len), 1, fptr);
// scan through the block
for (i = 0; i < len; i ++)
{
fread(&temp, 1, 1, fptr);
if (feof(fptr)) return -1; // unexpected eof
}
}
return 0;
}
int WavInFile::readWavHeaders()
{
int res;
memset(&header, 0, sizeof(header));
res = readRIFFBlock();
if (res) return 1;
// read header blocks until data block is found
do
{
// read header blocks
res = readHeaderBlock();
if (res < 0) return 1; // error in file structure
} while (res == 0);
// check that all required tags are legal
return checkCharTags();
}
uint WavInFile::getNumChannels() const
{
return header.format.channel_number;
}
uint WavInFile::getNumBits() const
{
return header.format.bits_per_sample;
}
uint WavInFile::getBytesPerSample() const
{
return getNumChannels() * getNumBits() / 8;
}
uint WavInFile::getSampleRate() const
{
return header.format.sample_rate;
}
uint WavInFile::getDataSizeInBytes() const
{
return header.data.data_len;
}
uint WavInFile::getNumSamples() const
{
return header.data.data_len / header.format.byte_per_sample;
}
uint WavInFile::getLengthMS() const
{
uint numSamples;
uint sampleRate;
numSamples = getNumSamples();
sampleRate = getSampleRate();
assert(numSamples < UINT_MAX / 1000);
return (1000 * numSamples / sampleRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// Class WavOutFile
//
WavOutFile::WavOutFile(const char *fileName, int sampleRate, int bits, int channels)
{
bytesWritten = 0;
fptr = fopen(fileName, "wb");
if (fptr == NULL)
{
string msg = "Error : Unable to open file \"";
msg += fileName;
msg += "\" for writing.";
//pmsg = msg.c_str;
throw runtime_error(msg);
}
fillInHeader(sampleRate, bits, channels);
writeHeader();
flushTime = flushRate;
}
WavOutFile::~WavOutFile()
{
close();
}
void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
{
// fill in the 'riff' part..
// copy string 'RIFF' to riff_char
memcpy(&(header.riff.riff_char), riffStr, 4);
// package_len unknown so far
header.riff.package_len = 0;
// copy string 'WAVE' to wave
memcpy(&(header.riff.wave), waveStr, 4);
// fill in the 'format' part..
// copy string 'fmt ' to fmt
memcpy(&(header.format.fmt), fmtStr, 4);
header.format.format_len = 0x10;
header.format.fixed = 1;
header.format.channel_number = (short)channels;
header.format.sample_rate = 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;
// fill in the 'data' part..
// copy string 'data' to data_field
memcpy(&(header.data.data_field), dataStr, 4);
// data_len unknown so far
header.data.data_len = 0;
}
void WavOutFile::finishHeader()
{
// supplement the file length into the header structure
header.riff.package_len = bytesWritten + 36;
header.data.data_len = bytesWritten;
writeHeader();
}
void WavOutFile::writeHeader()
{
WavHeader hdrTemp;
// swap byte order if necessary
hdrTemp = header;
_swap32((unsigned int &)hdrTemp.riff.package_len);
_swap32((unsigned int &)hdrTemp.format.format_len);
_swap16((unsigned short &)hdrTemp.format.fixed);
_swap16((unsigned short &)hdrTemp.format.channel_number);
_swap32((unsigned int &)hdrTemp.format.sample_rate);
_swap32((unsigned int &)hdrTemp.format.byte_rate);
_swap16((unsigned short &)hdrTemp.format.byte_per_sample);
_swap16((unsigned short &)hdrTemp.format.bits_per_sample);
_swap32((unsigned int &)hdrTemp.data.data_len);
// write the supplemented header in the beginning of the file
fseek(fptr, 0, SEEK_SET);
fwrite(&hdrTemp, sizeof(hdrTemp), 1, fptr);
// jump back to the end of the file
fseek(fptr, 0, SEEK_END);
}
void WavOutFile::close()
{
finishHeader();
fclose(fptr);
fptr = NULL;
}
void WavOutFile::flush( int numElems )
{
flushTime -= numElems;
if( flushTime < 0 )
{
flushTime += flushRate;
finishHeader();
}
}
void WavOutFile::write(const char *buffer, int numElems)
{
int res;
if (header.format.bits_per_sample != 8)
{
throw runtime_error("Error: WavOutFile::write(const char*, int) accepts only 8bit samples.");
}
assert(sizeof(char) == 1);
res = fwrite(buffer, 1, numElems, fptr);
if (res != numElems)
{
throw runtime_error("Error while writing to a wav file.");
}
bytesWritten += numElems;
flush( numElems );
}
void WavOutFile::write(const short *buffer, int numElems)
{
int res;
// 16 bit samples
if (numElems < 1) return; // nothing to do
if (header.format.bits_per_sample == 8)
{
int i;
char *temp = new char[numElems];
// convert from 16bit format to 8bit format
for (i = 0; i < numElems; i ++)
{
temp[i] = buffer[i] >> 8;
}
// write in 8bit format
write(temp, numElems);
delete[] temp;
}
else
{
// 16bit format
unsigned short *pTemp = new unsigned short[numElems];
assert(header.format.bits_per_sample == 16);
// allocate temp buffer to swap byte order if necessary
memcpy(pTemp, buffer, numElems * 2);
_swap16Buffer(pTemp, numElems);
res = fwrite(pTemp, 2, numElems, fptr);
delete[] pTemp;
if (res != numElems)
{
throw runtime_error("Error while writing to a wav file.");
}
bytesWritten += 2 * numElems;
flush( numElems*2 );
}
}
void WavOutFile::write(const float *buffer, int numElems)
{
int i;
short *temp = new short[numElems];
int iTemp;
// convert to 16 bit integer
for (i = 0; i < numElems; i ++)
{
// convert to integer
iTemp = (int)(32768.0f * buffer[i]);
// saturate
if (iTemp < -32768) iTemp = -32768;
if (iTemp > 32767) iTemp = 32767;
temp[i] = (short)iTemp;
}
write(temp, numElems);
flush( numElems );
delete[] temp;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Classes for easy reading & writing of WAV sound files.
///
/// For big-endian CPU, define BIG_ENDIAN during compile-time to correctly
/// parse the WAV files with such processors.
///
/// Admittingly, more complete WAV reader routines may exist in public domain, but
/// the reason for 'yet another' one is that those generic WAV reader libraries are
/// exhaustingly large and cumbersome! Wanted to have something simpler here, i.e.
/// something that's not already larger than rest of the SoundTouch/SoundStretch program...
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.7 $
//
// $Id: WavFile.h,v 1.7 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef WAVFILE_H
#define WAVFILE_H
#include <stdio.h>
#ifndef uint
typedef unsigned int uint;
#endif
/// WAV audio file 'riff' section header
typedef struct
{
char riff_char[4];
int package_len;
char wave[4];
} WavRiff;
/// WAV audio file 'format' section header
typedef struct
{
char fmt[4];
int format_len;
short fixed;
short channel_number;
int sample_rate;
int byte_rate;
short byte_per_sample;
short bits_per_sample;
} WavFormat;
/// WAV audio file 'data' section header
typedef struct
{
char data_field[4];
uint data_len;
} WavData;
/// WAV audio file header
typedef struct
{
WavRiff riff;
WavFormat format;
WavData data;
} WavHeader;
/// Class for reading WAV audio files.
class WavInFile
{
private:
/// File pointer.
FILE *fptr;
/// Counter of how many bytes of sample data have been read from the file.
uint dataRead;
/// WAV header information
WavHeader header;
/// Read WAV file headers.
/// \return zero if all ok, nonzero if file format is invalid.
int readWavHeaders();
/// Checks WAV file header tags.
/// \return zero if all ok, nonzero if file format is invalid.
int checkCharTags();
/// Reads a single WAV file header block.
/// \return zero if all ok, nonzero if file format is invalid.
int readHeaderBlock();
/// Reads WAV file 'riff' block
int readRIFFBlock();
public:
/// Constructor: Opens the given WAV file. If the file can't be opened,
/// throws 'runtime_error' exception.
WavInFile(const char *filename);
/// 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();
/// Get sample rate.
uint getSampleRate() const;
/// Get number of bits per sample, i.e. 8 or 16.
uint getNumBits() const;
/// Get sample data size in bytes. Ahem, this should return same information as
/// 'getBytesPerSample'...
uint getDataSizeInBytes() const;
/// Get total number of samples in file.
uint getNumSamples() const;
/// Get number of bytes per audio sample (e.g. 16bit stereo = 4 bytes/sample)
uint getBytesPerSample() const;
/// Get number of audio channels in the file (1=mono, 2=stereo)
uint getNumChannels() const;
/// Get the audio file length in milliseconds
uint getLengthMS() const;
/// Reads audio samples from the WAV file. This routine works only for 8 bit samples.
/// Reads given number of elements from the file or if end-of-file reached, as many
/// elements as are left in the file.
///
/// \return Number of 8-bit integers read from the file.
int read(char *buffer, int maxElems);
/// Reads audio samples from the WAV file to 16 bit integer format. Reads given number
/// of elements from the file or if end-of-file reached, as many elements as are
/// left in the file.
///
/// \return Number of 16-bit integers read from the file.
int read(short *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Reads audio samples from the WAV file to floating point format, converting
/// sample values to range [-1,1[. Reads given number of elements from the file
/// or if end-of-file reached, as many elements as are left in the file.
///
/// \return Number of elements read from the file.
int read(float *buffer, ///< Pointer to buffer where to read data.
int maxElems ///< Size of 'buffer' array (number of array elements).
);
/// Check end-of-file.
///
/// \return Nonzero if end-of-file reached.
int eof() const;
};
/// Class for writing WAV audio files.
class WavOutFile
{
private:
/// Pointer to the WAV file
FILE *fptr;
/// WAV file header data.
WavHeader header;
/// Counter of how many bytes have been written to the file so far.
int bytesWritten;
/// number of bytes to be written before next flush.
int flushTime;
/// Fills in WAV file header information.
void fillInHeader(const uint sampleRate, const uint bits, const uint channels);
/// Finishes the WAV file header by supplementing information of amount of
/// data written to file etc
void finishHeader();
/// Writes the WAV file header.
void writeHeader();
/// Flushes the WAV file every so often -- writes header info for the current
/// data length and then returns the seek position to the end of the WAV for
/// continued writing. This method is called from each write() method.
void flush( int numElems );
/// Flush the WAVheader every 32kb written
static const int flushRate = 0x8000;
public:
/// Constructor: Creates a new WAV file. Throws a 'runtime_error' exception
/// if file creation fails.
WavOutFile(const char *fileName, ///< Filename
int sampleRate, ///< Sample rate (e.g. 44100 etc)
int bits, ///< Bits per sample (8 or 16 bits)
int channels ///< Number of channels (1=mono, 2=stereo)
);
/// Destructor: Finalizes & closes the WAV file.
~WavOutFile();
/// Write data to WAV file. This function works only with 8bit samples.
/// Throws a 'runtime_error' exception if writing to file fails.
void write(const char *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file. Throws a 'runtime_error' exception if writing to
/// file fails.
void write(const short *buffer, ///< Pointer to sample data buffer.
int numElems ///< How many array items are to be written to file.
);
/// Write data to WAV file in floating point format, saturating sample values to range
/// [-1..+1[. Throws a 'runtime_error' exception if writing to file fails.
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

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#!/bin/sh
curdir=`pwd`
echo -----------------
echo Building SoundTouch
echo -----------------
if [ $# -gt 0 ] && [ $1 = "all" ]
then
aclocal
automake -a
autoconf
./configure
make clean
make install
else
make $@
fi
if [ $? -ne 0 ]
then
exit 1
fi
#cp libZeroSPU2*.so* ${PCSX2PLUGINS}

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# -*- Autoconf -*-
# Process this file with autoconf to produce a configure script.
#AC_PREREQ([2.63])
AC_INIT([FULL-PACKAGE-NAME], [VERSION], [BUG-REPORT-ADDRESS])
AM_INIT_AUTOMAKE
AC_CONFIG_SRCDIR([BPMDetect.h])
# Checks for programs.
AC_PROG_CXX
AC_PROG_CC
AC_PROG_RANLIB
CFLAGS=
CPPFLAGS=
CXXFLAGS=
CCASFLAGS=
CFLAGS+=" -m32 "
CPPFLAGS+=" -m32 "
CXXFLAGS+=" -m32 "
CCASFLAGS+=" -m32 "
# Checks for header files.
AC_CHECK_HEADERS([limits.h memory.h stdlib.h string.h])
# Checks for typedefs, structures, and compiler characteristics.
AC_C_INLINE
AC_C_RESTRICT
AC_TYPE_SIZE_T
AC_HEADER_STDBOOL
# Checks for library functions.
AC_CHECK_FUNCS([memmove memset])
AC_CONFIG_FILES([Makefile])
AC_OUTPUT

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////////////////////////////////////////////////////////////////////////////////
///
/// A header file for detecting the Intel MMX instructions set extension.
///
/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
/// routine implementations for x86 Windows, x86 gnu version and non-x86
/// platforms, respectively.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.4 $
//
// $Id: cpu_detect.h,v 1.4 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _CPU_DETECT_H_
#define _CPU_DETECT_H_
#include "STTypes.h"
#define SUPPORT_MMX 0x0001
#define SUPPORT_3DNOW 0x0002
#define SUPPORT_ALTIVEC 0x0004
#define SUPPORT_SSE 0x0008
#define SUPPORT_SSE2 0x0010
/// Checks which instruction set extensions are supported by the CPU.
///
/// \return A bitmask of supported extensions, see SUPPORT_... defines.
uint detectCPUextensions(void);
/// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint wDisableMask);
#endif // _CPU_DETECT_H_

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////////////////////////////////////////////////////////////////////////////////
///
/// gcc version of the x86 CPU detect routine.
///
/// This file is to be compiled on any platform with the GNU C compiler.
/// Compiler. Please see 'cpu_detect_x86_win.cpp' for the x86 Windows version
/// of this file.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.6 $
//
// $Id: cpu_detect_x86_gcc.cpp,v 1.6 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdexcept>
#include <string>
#include "cpu_detect.h"
#ifndef __GNUC__
#error wrong platform - this source code file is for the GNU C compiler.
#endif
using namespace std;
#include <stdio.h>
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
#ifndef __i386__
return 0; // always disable extensions on non-x86 platforms.
#else
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
asm volatile(
"\n\txor %%esi, %%esi" // clear %%esi = result register
// check if 'cpuid' instructions is available by toggling eflags bit 21
"\n\tpushf" // save eflags to stack
"\n\tpop %%eax" // load eax from stack (with eflags)
"\n\tmovl %%eax, %%ecx" // save the original eflags values to ecx
"\n\txor $0x00200000, %%eax" // toggle bit 21
"\n\tpush %%eax" // store toggled eflags to stack
"\n\tpopf" // load eflags from stack
"\n\tpushf" // save updated eflags to stack
"\n\tpop %%eax" // load from stack
"\n\txor %%edx, %%edx" // clear edx for defaulting no mmx
"\n\tcmp %%ecx, %%eax" // compare to original eflags values
"\n\tjz end" // jumps to 'end' if cpuid not present
// cpuid instruction available, test for presence of mmx instructions
"\n\tmovl $1, %%eax"
"\n\tcpuid"
// movl $0x00800000, %edx // force enable MMX
"\n\ttest $0x00800000, %%edx"
"\n\tjz end" // branch if MMX not available
"\n\tor $0x01, %%esi" // otherwise add MMX support bit
"\n\ttest $0x02000000, %%edx"
"\n\tjz test3DNow" // branch if SSE not available
"\n\tor $0x08, %%esi" // otherwise add SSE support bit
"\n\ttest3DNow:"
// test for precense of AMD extensions
"\n\tmov $0x80000000, %%eax"
"\n\tcpuid"
"\n\tcmp $0x80000000, %%eax"
"\n\tjbe end" // branch if no AMD extensions detected
// test for precense of 3DNow! extension
"\n\tmov $0x80000001, %%eax"
"\n\tcpuid"
"\n\ttest $0x80000000, %%edx"
"\n\tjz end" // branch if 3DNow! not detected
"\n\tor $0x02, %%esi" // otherwise add 3DNow support bit
"\n\tend:"
"\n\tmov %%esi, %0"
: "=r" (res)
: /* no inputs */
: "%edx", "%eax", "%ecx", "%esi" );
return res & ~_dwDisabledISA;
#endif
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Win32 version of the x86 CPU detect routine.
///
/// This file is to be compiled in Windows platform with Microsoft Visual C++
/// Compiler. Please see 'cpu_detect_x86_gcc.cpp' for the gcc compiler version
/// for all GNU platforms.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.10 $
//
// $Id: cpu_detect_x86_win.cpp,v 1.10 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#ifndef _WIN32
#error wrong platform - this source code file is exclusively for Win32 platform
#endif
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
uint res = 0;
if (_dwDisabledISA == 0xffffffff) return 0;
_asm
{
; check if 'cpuid' instructions is available by toggling eflags bit 21
;
xor esi, esi ; clear esi = result register
pushfd ; save eflags to stack
pop eax ; load eax from stack (with eflags)
mov ecx, eax ; save the original eflags values to ecx
xor eax, 0x00200000 ; toggle bit 21
push eax ; store toggled eflags to stack
popfd ; load eflags from stack
pushfd ; save updated eflags to stack
pop eax ; load from stack
xor edx, edx ; clear edx for defaulting no mmx
cmp eax, ecx ; compare to original eflags values
jz end ; jumps to 'end' if cpuid not present
; cpuid instruction available, test for presence of mmx instructions
mov eax, 1
cpuid
test edx, 0x00800000
jz end ; branch if MMX not available
or esi, SUPPORT_MMX ; otherwise add MMX support bit
test edx, 0x02000000
jz test3DNow ; branch if SSE not available
or esi, SUPPORT_SSE ; otherwise add SSE support bit
test3DNow:
; test for precense of AMD extensions
mov eax, 0x80000000
cpuid
cmp eax, 0x80000000
jbe end ; branch if no AMD extensions detected
; test for precense of 3DNow! extension
mov eax, 0x80000001
cpuid
test edx, 0x80000000
jz end ; branch if 3DNow! not detected
or esi, SUPPORT_3DNOW ; otherwise add 3DNow support bit
end:
mov res, esi
}
return res & ~_dwDisabledISA;
}

1
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link /usr/share/automake-1.10/depcomp

1
desmume/src/metaspu/SoundTouch/install-sh Normal file
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link /usr/share/automake-1.10/install-sh

1
desmume/src/metaspu/SoundTouch/missing Normal file
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link /usr/share/automake-1.10/missing

305
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////////////////////////////////////////////////////////////////////////////////
///
/// MMX optimized routines. All MMX optimized functions have been gathered into
/// this single source code file, regardless to their class or original source
/// code file, in order to ease porting the library to other compiler and
/// processor platforms.
///
/// The MMX-optimizations are programmed using MMX compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
/// is available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/06 18:52:43 $
// File revision : $Revision: 1.1 $
//
// $Id: mmx_optimized.cpp,v 1.1 2006/02/06 18:52:43 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "STTypes.h"
#ifdef ALLOW_MMX
// MMX routines available only with integer sample type
#if !(_WIN32 || __i386__ || __x86_64__)
#error "wrong platform - this source code file is exclusively for x86 platforms"
#endif
using namespace soundtouch;
//////////////////////////////////////////////////////////////////////////////
//
// implementation of MMX optimized functions of class 'TDStretchMMX'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
#include <mmintrin.h>
#include <limits.h>
// Calculates cross correlation of two buffers
long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
__m64 accu;
long corr;
uint i;
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBits);
accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples each during each
// round to improve CPU-level parallellization.
for (i = 0; i < overlapLength / 8; i ++)
{
__m64 temp;
// dictionary of instructions:
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
// _mm_add_pi32 : 2*32bit add
// _m_psrad : 32bit right-shift
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),
_mm_madd_pi16(pVec1[1], pVec2[1]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),
_mm_madd_pi16(pVec1[3], pVec2[3]));
accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));
pVec1 += 4;
pVec2 += 4;
}
// copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
// and finally store the result into the variable "corr"
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
corr = _m_to_int(accu);
// Clear MMS state
_m_empty();
return corr;
// Note: Warning about the missing EMMS instruction is harmless
// as it'll be called elsewhere.
}
void TDStretchMMX::clearCrossCorrState()
{
// Clear MMS state
_m_empty();
//_asm EMMS;
}
// MMX-optimized version of the function overlapStereo
void TDStretchMMX::overlapStereo(short *output, const short *input) const
{
const __m64 *pVinput, *pVMidBuf;
__m64 *pVdest;
__m64 mix1, mix2, adder, shifter;
uint i;
pVinput = (const __m64*)input;
pVMidBuf = (const __m64*)pMidBuffer;
pVdest = (__m64*)output;
// mix1 = mixer values for 1st stereo sample
// mix1 = mixer values for 2nd stereo sample
// adder = adder for updating mixer values after each round
mix1 = _mm_set_pi16(0, overlapLength, 0, overlapLength);
adder = _mm_set_pi16(1, -1, 1, -1);
mix2 = _mm_add_pi16(mix1, adder);
adder = _mm_add_pi16(adder, adder);
shifter = _m_from_int(overlapDividerBits);
for (i = 0; i < overlapLength / 4; i ++)
{
__m64 temp1, temp2;
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r
temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r
// temp = (temp .* mix) >> shifter
temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
// update mix += adder
mix1 = _mm_add_pi16(mix1, adder);
mix2 = _mm_add_pi16(mix2, adder);
// --- second round begins here ---
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]); // = i2l m2l i2r m2r
temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]); // = i3l m3l i3r m3r
// temp = (temp .* mix) >> shifter
temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
// update mix += adder
mix1 = _mm_add_pi16(mix1, adder);
mix2 = _mm_add_pi16(mix2, adder);
pVinput += 2;
pVMidBuf += 2;
pVdest += 2;
}
_m_empty(); // clear MMS state
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of MMX optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilterMMX::FIRFilterMMX() : FIRFilter()
{
filterCoeffsUnalign = NULL;
}
FIRFilterMMX::~FIRFilterMMX()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for MMX routine
void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new short[2 * newLength + 8];
filterCoeffsAlign = (short *)(((ulongptr)filterCoeffsUnalign + 15) & -16);
// rearrange the filter coefficients for mmx routines
for (i = 0;i < length; i += 4)
{
filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
}
}
// mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, const uint numSamples) const
{
// Create stack copies of the needed member variables for asm routines :
uint i, j;
__m64 *pVdest = (__m64*)dest;
if (length < 2) return 0;
for (i = 0; i < numSamples / 2; i ++)
{
__m64 accu1;
__m64 accu2;
const __m64 *pVsrc = (const __m64*)src;
const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
accu1 = accu2 = _mm_setzero_si64();
for (j = 0; j < lengthDiv8 * 2; j ++)
{
__m64 temp1, temp2;
temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]); // = l2 l0 r2 r0
temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]); // = l3 l1 r3 r1
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0])); // += l2*f2+l0*f0 r2*f2+r0*f0
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1])); // += l3*f3+l1*f1 r3*f3+r1*f1
temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]); // = l4 l2 r4 r2
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0])); // += l3*f2+l1*f0 r3*f2+r1*f0
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1])); // += l4*f3+l2*f1 r4*f3+r2*f1
// accu1 += l2*f2+l0*f0 r2*f2+r0*f0
// += l3*f3+l1*f1 r3*f3+r1*f1
// accu2 += l3*f2+l1*f0 r3*f2+r1*f0
// l4*f3+l2*f1 r4*f3+r2*f1
pVfilter += 2;
pVsrc += 2;
}
// accu >>= resultDivFactor
accu1 = _mm_srai_pi32(accu1, resultDivFactor);
accu2 = _mm_srai_pi32(accu2, resultDivFactor);
// pack 2*2*32bits => 4*16 bits
pVdest[0] = _mm_packs_pi32(accu1, accu2);
src += 4;
pVdest ++;
}
_m_empty(); // clear emms state
return (numSamples & 0xfffffffe) - length;
}
#endif // ALLOW_MMX

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////////////////////////////////////////////////////////////////////////////////
///
/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
/// optimized functions have been gathered into this single source
/// code file, regardless to their class or original source code file, in order
/// to ease porting the library to other compiler and processor platforms.
///
/// The SSE-optimizations are programmed using SSE compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support SSE instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/vstudio/downloads/tools/ppack/default.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// perform a search with keywords "processor pack".
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2006/02/05 16:44:06 $
// File revision : $Revision: 1.2 $
//
// $Id: sse_optimized.cpp,v 1.2 2006/02/05 16:44:06 Olli Exp $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
using namespace soundtouch;
#ifdef ALLOW_SSE
// SSE routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'TDStretchSSE'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
#include <xmmintrin.h>
// Calculates cross correlation of two buffers
double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const
{
uint i;
__m128 vSum, *pVec2;
// 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
// unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
// This can mean up to ~ 10-fold difference (incl. part of which is
// due to skipping every second round for stereo sound though).
//
// Compile-time define ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
// for choosing if this little cheating is allowed.
#ifdef ALLOW_NONEXACT_SIMD_OPTIMIZATION
// Little cheating allowed, return valid correlation only for
// aligned locations, meaning every second round for stereo sound.
#define _MM_LOAD _mm_load_ps
if (((ulong)pV1) & 15) return -1e50; // skip unaligned locations
#else
// No cheating allowed, use unaligned load & take the resulting
// performance hit.
#define _MM_LOAD _mm_loadu_ps
#endif
// ensure overlapLength is divisible by 8
assert((overlapLength % 8) == 0);
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
// Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
pVec2 = (__m128*)pV2;
vSum = _mm_setzero_ps();
// Unroll the loop by factor of 4 * 4 operations
for (i = 0; i < overlapLength / 8; i ++)
{
// vSum += pV1[0..3] * pV2[0..3]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1),pVec2[0]));
// vSum += pV1[4..7] * pV2[4..7]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 4), pVec2[1]));
// vSum += pV1[8..11] * pV2[8..11]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 8), pVec2[2]));
// vSum += pV1[12..15] * pV2[12..15]
vSum = _mm_add_ps(vSum, _mm_mul_ps(_MM_LOAD(pV1 + 12), pVec2[3]));
pV1 += 16;
pVec2 += 4;
}
// return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
float *pvSum = (float*)&vSum;
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]);
/* This is approximately corresponding routine in C-language:
double corr;
uint i;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
corr = 0.0;
for (i = 0; i < overlapLength / 8; i ++)
{
corr += pV1[0] * pV2[0] +
pV1[1] * pV2[1] +
pV1[2] * pV2[2] +
pV1[3] * pV2[3] +
pV1[4] * pV2[4] +
pV1[5] * pV2[5] +
pV1[6] * pV2[6] +
pV1[7] * pV2[7] +
pV1[8] * pV2[8] +
pV1[9] * pV2[9] +
pV1[10] * pV2[10] +
pV1[11] * pV2[11] +
pV1[12] * pV2[12] +
pV1[13] * pV2[13] +
pV1[14] * pV2[14] +
pV1[15] * pV2[15];
pV1 += 16;
pV2 += 16;
}
*/
/* This is corresponding routine in assembler. This may be teeny-weeny bit faster
than intrinsic version, but more difficult to maintain & get compiled on multiple
platforms.
uint overlapLengthLocal = overlapLength;
float corr;
_asm
{
// Very important note: data in 'pV2' _must_ be aligned to
// 16-byte boundary!
// give prefetch hints to CPU of what data are to be needed soonish
// give more aggressive hints on pV1 as that changes while pV2 stays
// same between runs
prefetcht0 [pV1]
prefetcht0 [pV2]
prefetcht0 [pV1 + 32]
mov eax, dword ptr pV1
mov ebx, dword ptr pV2
xorps xmm0, xmm0
mov ecx, overlapLengthLocal
shr ecx, 3 // div by eight
loop1:
prefetcht0 [eax + 64] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [ebx + 32] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm1, [eax]
mulps xmm1, [ebx]
addps xmm0, xmm1
movups xmm2, [eax + 16]
mulps xmm2, [ebx + 16]
addps xmm0, xmm2
prefetcht0 [eax + 96] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [ebx + 64] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm3, [eax + 32]
mulps xmm3, [ebx + 32]
addps xmm0, xmm3
movups xmm4, [eax + 48]
mulps xmm4, [ebx + 48]
addps xmm0, xmm4
add eax, 64
add ebx, 64
dec ecx
jnz loop1
// add the four floats of xmm0 together and return the result.
movhlps xmm1, xmm0 // move 3 & 4 of xmm0 to 1 & 2 of xmm1
addps xmm1, xmm0
movaps xmm2, xmm1
shufps xmm2, xmm2, 0x01 // move 2 of xmm2 as 1 of xmm2
addss xmm2, xmm1
movss corr, xmm2
}
return (double)corr;
*/
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilterSSE::FIRFilterSSE() : FIRFilter()
{
filterCoeffsUnalign = NULL;
}
FIRFilterSSE::~FIRFilterSSE()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for SSE routine
void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for 3DNow!
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)(((unsigned long)filterCoeffsUnalign + 15) & -16);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// SSE-optimized version of the filter routine for stereo sound
uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
{
int count = (numSamples - length) & -2;
int j;
assert(count % 2 == 0);
if (count < 2) return 0;
assert((length % 8) == 0);
assert(((unsigned long)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
for (j = 0; j < count; j += 2)
{
const float *pSrc;
const __m128 *pFil;
__m128 sum1, sum2;
uint i;
pSrc = source; // source audio data
pFil = (__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 ++)
{
// Unroll loop for efficiency & calculate filter for 2*2 stereo samples
// at each pass
// sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
// sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
pSrc += 16;
pFil += 4;
}
// Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
// to sum the two hi- and lo-floats of these registers together.
// post-shuffle & add the filtered values and store to dest.
_mm_storeu_ps(dest, _mm_add_ps(
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
));
source += 4;
dest += 4;
}
// Ideas for further improvement:
// 1. If it could be guaranteed that 'source' were always aligned to 16-byte
// boundary, a faster aligned '_mm_load_ps' instruction could be used.
// 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
// boundary, a faster '_mm_store_ps' instruction could be used.
return (uint)count;
/* original routine in C-language. please notice the C-version has differently
organized coefficients though.
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
const float *pFil;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
pFil = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * pFil[0] +
ptr[2] * pFil[2] +
ptr[4] * pFil[4] +
ptr[6] * pFil[6];
sumr1 += ptr[1] * pFil[1] +
ptr[3] * pFil[3] +
ptr[5] * pFil[5] +
ptr[7] * pFil[7];
suml2 += ptr[8] * pFil[0] +
ptr[10] * pFil[2] +
ptr[12] * pFil[4] +
ptr[14] * pFil[6];
sumr2 += ptr[9] * pFil[1] +
ptr[11] * pFil[3] +
ptr[13] * pFil[5] +
ptr[15] * pFil[7];
ptr += 16;
pFil += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
/* Similar routine in assembly, again obsoleted due to maintainability
_asm
{
// Very important note: data in 'src' _must_ be aligned to
// 16-byte boundary!
mov edx, count
mov ebx, dword ptr src
mov eax, dword ptr dest
shr edx, 1
loop1:
// "outer loop" : during each round 2*2 output samples are calculated
// give prefetch hints to CPU of what data are to be needed soonish
prefetcht0 [ebx]
prefetcht0 [filterCoeffsLocal]
mov esi, ebx
mov edi, filterCoeffsLocal
xorps xmm0, xmm0
xorps xmm1, xmm1
mov ecx, lengthLocal
loop2:
// "inner loop" : during each round eight FIR filter taps are evaluated for 2*2 samples
prefetcht0 [esi + 32] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [edi + 32] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm2, [esi] // possibly unaligned load
movups xmm3, [esi + 8] // possibly unaligned load
mulps xmm2, [edi]
mulps xmm3, [edi]
addps xmm0, xmm2
addps xmm1, xmm3
movups xmm4, [esi + 16] // possibly unaligned load
movups xmm5, [esi + 24] // possibly unaligned load
mulps xmm4, [edi + 16]
mulps xmm5, [edi + 16]
addps xmm0, xmm4
addps xmm1, xmm5
prefetcht0 [esi + 64] // give a prefetch hint to CPU what data are to be needed soonish
prefetcht0 [edi + 64] // give a prefetch hint to CPU what data are to be needed soonish
movups xmm6, [esi + 32] // possibly unaligned load
movups xmm7, [esi + 40] // possibly unaligned load
mulps xmm6, [edi + 32]
mulps xmm7, [edi + 32]
addps xmm0, xmm6
addps xmm1, xmm7
movups xmm4, [esi + 48] // possibly unaligned load
movups xmm5, [esi + 56] // possibly unaligned load
mulps xmm4, [edi + 48]
mulps xmm5, [edi + 48]
addps xmm0, xmm4
addps xmm1, xmm5
add esi, 64
add edi, 64
dec ecx
jnz loop2
// Now xmm0 and xmm1 both have a filtered 2-channel sample each, but we still need
// to sum the two hi- and lo-floats of these registers together.
movhlps xmm2, xmm0 // xmm2 = xmm2_3 xmm2_2 xmm0_3 xmm0_2
movlhps xmm2, xmm1 // xmm2 = xmm1_1 xmm1_0 xmm0_3 xmm0_2
shufps xmm0, xmm1, 0xe4 // xmm0 = xmm1_3 xmm1_2 xmm0_1 xmm0_0
addps xmm0, xmm2
movaps [eax], xmm0
add ebx, 16
add eax, 16
dec edx
jnz loop1
}
*/
}
#endif // ALLOW_SSE

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/* SPU2-X, A plugin for Emulating the Sound Processing Unit of the Playstation 2
* Developed and maintained by the Pcsx2 Development Team.
*
* Original portions from SPU2ghz are (c) 2008 by David Quintana [gigaherz]
*
* SPU2-X is free software: you can redistribute it and/or modify it under the terms
* of the GNU Lesser General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* SPU2-X is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with SPU2-X. If not, see <http://www.gnu.org/licenses/>.
*/
//#include "Global.h"
#include <math.h>
#include "types.h"
#include "SoundTouch/SoundTouch.h"
#include "SndOut.h"
//#include "SoundTouch/WavFile.h"
#include "win32/Dialogs.h"
static soundtouch::SoundTouch* pSoundTouch = NULL;
static int ts_stats_stretchblocks = 0;
static int ts_stats_normalblocks = 0;
static int ts_stats_logcounter = 0;
// data prediction amount, used to "commit" data that hasn't
// finished timestretch processing.
s32 SndBuffer::m_predictData;
// records last buffer status (fill %, range -100 to 100, with 0 being 50% full)
float SndBuffer::lastPct;
float SndBuffer::lastEmergencyAdj;
float SndBuffer::cTempo = 1;
float SndBuffer::eTempo = 1;
int SndBuffer::freezeTempo = 0;
void SndBuffer::PredictDataWrite( int samples )
{
m_predictData += samples;
}
// Calculate the buffer status percentage.
// Returns range from -1.0 to 1.0
// 1.0 = buffer overflow!
// 0.0 = buffer nominal (50% full)
// -1.0 = buffer underflow!
float SndBuffer::GetStatusPct()
{
// Get the buffer status of the output driver too, so that we can
// obtain a more accurate overall buffer status.
int drvempty = 0;
//int drvempty = mods[OutputModule]->GetEmptySampleCount(); // / 2;
//TODO
//ConLog( "Data %d >>> driver: %d predict: %d\n", data, drvempty, predictData );
float result = (float)(m_data + m_predictData - drvempty) - (m_size/2);
result /= (m_size/2);
return result;
}
void SndBuffer::UpdateTempoChange()
{
if( --freezeTempo > 0 )
{
return;
}
float statusPct = GetStatusPct();
float pctChange = statusPct - lastPct;
float tempoChange;
float emergencyAdj = 0;
float newcee = cTempo; // workspace var. for cTempo
// IMPORTANT!
// If you plan to tweak these values, make sure you're using a release build
// OUTSIDE THE DEBUGGER to test it! The Visual Studio debugger can really cause
// erratic behavior in the audio buffers, and makes the timestretcher seem a
// lot more inconsistent than it really is.
// We have two factors.
// * Distance from nominal buffer status (50% full)
// * The change from previous update to this update.
// Prediction based on the buffer change:
// (linear seems to work better here)
tempoChange = pctChange * 0.75f;
if( statusPct * tempoChange < 0.0f )
{
// only apply tempo change if it is in synch with the buffer status.
// In other words, if the buffer is high (over 0%), and is decreasing,
// ignore it. It'll just muck things up.
tempoChange = 0;
}
// Sudden spikes in framerate can cause the nominal buffer status
// to go critical, in which case we have to enact an emergency
// stretch. The following cubic formulas do that. Values near
// the extremeites give much larger results than those near 0.
// And the value is added only this time, and does not accumulate.
// (otherwise a large value like this would cause problems down the road)
// Constants:
// Weight - weights the statusPct's "emergency" consideration.
// higher values here will make the buffer perform more drastic
// compensations at the outer edges of the buffer (at -75 or +75%
// or beyond, for example).
// Range - scales the adjustment to the given range (more or less).
// The actual range is dependent on the weight used, so if you increase
// Weight you'll usually want to decrease Range somewhat to compensate.
// Prediction based on the buffer fill status:
const float statusWeight = 2.99f;
const float statusRange = 0.068f;
// "non-emergency" deadzone: In this area stretching will be strongly discouraged.
// Note: due tot he nature of timestretch latency, it's always a wee bit harder to
// cope with low fps (underruns) than it is high fps (overruns). So to help out a
// little, the low-end portions of this check are less forgiving than the high-sides.
if( cTempo < 0.965f || cTempo > 1.060f ||
pctChange < -0.38f || pctChange > 0.54f ||
statusPct < -0.32f || statusPct > 0.39f ||
eTempo < 0.89f || eTempo > 1.19f )
{
emergencyAdj = ( pow( statusPct*statusWeight, 3.0f ) * statusRange);
}
// Smooth things out by factoring our previous adjustment into this one.
// It helps make the system 'feel' a little smarter by giving it at least
// one packet worth of history to help work off of:
emergencyAdj = (emergencyAdj * 0.75f) + (lastEmergencyAdj * 0.25f );
lastEmergencyAdj = emergencyAdj;
lastPct = statusPct;
// Accumulate a fraction of the tempo change into the tempo itself.
// This helps the system run "smarter" to games that run consistently
// fast or slow by altering the base tempo to something closer to the
// game's active speed. In tests most games normalize within 2 seconds
// at 100ms latency, which is pretty good (larger buffers normalize even
// quicker).
newcee += newcee * (tempoChange+emergencyAdj) * 0.03f;
// Apply tempoChange as a scale of cTempo. That way the effect is proportional
// to the current tempo. (otherwise tempos rate of change at the extremes would
// be too drastic)
float newTempo = newcee + ( emergencyAdj * cTempo );
// ... and as a final optimization, only stretch if the new tempo is outside
// a nominal threshold. Keep this threshold check small, because it could
// cause some serious side effects otherwise. (enlarging the cTempo check above
// is usually better/safer)
if( newTempo < 0.970f || newTempo > 1.045f )
{
cTempo = (float)newcee;
if( newTempo < 0.10f ) newTempo = 0.10f;
else if( newTempo > 10.0f ) newTempo = 10.0f;
if( cTempo < 0.15f ) cTempo = 0.15f;
else if( cTempo > 7.5f ) cTempo = 7.5f;
pSoundTouch->setTempo( eTempo = (float)newTempo );
ts_stats_stretchblocks++;
/*ConLog(" * SPU2: [Nominal %d%%] [Emergency: %d%%] (baseTempo: %d%% ) (newTempo: %d%%) (buffer: %d%%)\n",
//(relation < 0.0) ? "Normalize" : "",
(int)(tempoChange * 100.0 * 0.03),
(int)(emergencyAdj * 100.0),
(int)(cTempo * 100.0),
(int)(newTempo * 100.0),
(int)(statusPct * 100.0)
);*/
}
else
{
// Nominal operation -- turn off stretching.
// note: eTempo 'slides' toward 1.0 for smoother audio and better
// protection against spikes.
if( cTempo != 1.0f )
{
cTempo = 1.0f;
eTempo = ( 1.0f + eTempo ) * 0.5f;
pSoundTouch->setTempo( eTempo );
}
else
{
if( eTempo != cTempo )
pSoundTouch->setTempo( eTempo=cTempo );
ts_stats_normalblocks++;
}
}
}
void SndBuffer::timeStretchUnderrun()
{
// timeStretcher failed it's job. We need to slow down the audio some.
cTempo -= (cTempo * 0.12f);
eTempo -= (eTempo * 0.30f);
if( eTempo < 0.1f ) eTempo = 0.1f;
pSoundTouch->setTempo( eTempo );
}
s32 SndBuffer::timeStretchOverrun()
{
// If we overran it means the timestretcher failed. We need to speed
// up audio playback.
cTempo += cTempo * 0.12f;
eTempo += eTempo * 0.40f;
if( eTempo > 7.5f ) eTempo = 7.5f;
pSoundTouch->setTempo( eTempo );
// Throw out just a little bit (two packets worth) to help
// give the TS some room to work:
return SndOutPacketSize*2;
}
static void CvtPacketToFloat( StereoOut32* srcdest )
{
StereoOutFloat* dest = (StereoOutFloat*)srcdest;
const StereoOut32* src = (StereoOut32*)srcdest;
for( uint i=0; i<SndOutPacketSize; ++i, ++dest, ++src )
*dest = (StereoOutFloat)*src;
}
// Parameter note: Size should always be a multiple of 128, thanks!
static void CvtPacketToInt( StereoOut32* srcdest, uint size )
{
//jASSUME( (size & 127) == 0 );
const StereoOutFloat* src = (StereoOutFloat*)srcdest;
StereoOut32* dest = srcdest;
for( uint i=0; i<size; ++i, ++dest, ++src )
*dest = (StereoOut32)*src;
}
void SndBuffer::timeStretchWrite()
{
bool progress = false;
// data prediction helps keep the tempo adjustments more accurate.
// The timestretcher returns packets in belated "clump" form.
// Meaning that most of the time we'll get nothing back, and then
// suddenly we'll get several chunks back at once. Thus we use
// data prediction to make the timestretcher more responsive.
PredictDataWrite( (int)( SndOutPacketSize / eTempo ) );
CvtPacketToFloat( sndTempBuffer );
pSoundTouch->putSamples( (float*)sndTempBuffer, SndOutPacketSize );
int tempProgress;
while( tempProgress = pSoundTouch->receiveSamples( (float*)sndTempBuffer, SndOutPacketSize),
tempProgress != 0 )
{
// Hint: It's assumed that pSoundTouch will return chunks of 128 bytes (it always does as
// long as the SSE optimizations are enabled), which means we can do our own SSE opts here.
CvtPacketToInt( sndTempBuffer, tempProgress );
_WriteSamples( sndTempBuffer, tempProgress );
progress = true;
}
UpdateTempoChange();
//TODO
//if( MsgOverruns() )
{
if( progress )
{
if( ++ts_stats_logcounter > 300 )
{
ts_stats_logcounter = 0;
printf( " * SPU2 > Timestretch Stats > %d%% of packets stretched.\n",
( ts_stats_stretchblocks * 100 ) / ( ts_stats_normalblocks + ts_stats_stretchblocks ) );
ts_stats_normalblocks = 0;
ts_stats_stretchblocks = 0;
}
}
}
}
void SndBuffer::soundtouchInit()
{
pSoundTouch = new soundtouch::SoundTouch();
pSoundTouch->setSampleRate(SampleRate);
pSoundTouch->setChannels(2);
pSoundTouch->setSetting( SETTING_USE_QUICKSEEK, 0 );
pSoundTouch->setSetting( SETTING_USE_AA_FILTER, 0 );
SoundtouchCfg::ApplySettings( *pSoundTouch );
pSoundTouch->setTempo(1);
// some timestretch management vars:
cTempo = 1.0;
eTempo = 1.0;
lastPct = 0;
lastEmergencyAdj = 0;
// just freeze tempo changes for a while at startup.
// the driver buffers are bogus anyway.
freezeTempo = 16;
m_predictData = 0;
}
// reset timestretch management vars, and delay updates a bit:
void SndBuffer::soundtouchClearContents()
{
if( pSoundTouch == NULL ) return;
pSoundTouch->clear();
pSoundTouch->setTempo(1);
cTempo = 1.0;
eTempo = 1.0;
lastPct = 0;
lastEmergencyAdj = 0;
freezeTempo = 16;
m_predictData = 0;
}
void SndBuffer::soundtouchCleanup()
{
//safe_delete( pSoundTouch );
delete pSoundTouch;
}

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/* Copyright 2009 DeSmuME team
This file is part of DeSmuME
DeSmuME is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
DeSmuME is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with DeSmuME; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "types.h"
#include "metaspu.h"
#include <queue>
#include <vector>
#include <assert.h>
//for pcsx2 method
//(havent bothered to get it compiling in gcc yet)
#ifdef _MSC_VER
#include "SndOut.h"
#endif
template<typename T> inline T _abs(T val)
{
if(val<0) return -val;
else return val;
}
template<typename T> inline T moveValueTowards(T val, T target, T incr)
{
incr = _abs(incr);
T delta = _abs(target-val);
if(val<target) val += incr;
else if(val>target) val -= incr;
T newDelta = _abs(target-val);
if(newDelta >= delta)
val = target;
return val;
}
class ZeromusSynchronizer : public ISynchronizingAudioBuffer
{
public:
ZeromusSynchronizer()
: mixqueue_go(false)
,
#ifdef NDEBUG
adjustobuf(200,1000)
#else
adjustobuf(22000,44000)
#endif
{
}
bool mixqueue_go;
virtual void enqueue_samples(s16* buf, int samples_provided)
{
for(int i=0;i<samples_provided;i++) {
s16 left = *buf++;
s16 right = *buf++;
adjustobuf.enqueue(left,right);
}
}
//returns the number of samples actually supplied, which may not match the number requested
virtual int output_samples(s16* buf, int samples_requested)
{
int done = 0;
if(!mixqueue_go) {
if(adjustobuf.size > 200)
mixqueue_go = true;
}
else
{
for(int i=0;i<samples_requested;i++) {
if(adjustobuf.size==0) {
mixqueue_go = false;
break;
}
done++;
s16 left, right;
adjustobuf.dequeue(left,right);
*buf++ = left;
*buf++ = right;
}
}
return done;
}
private:
class Adjustobuf
{
public:
Adjustobuf(int _minLatency, int _maxLatency)
: size(0)
, minLatency(_minLatency)
, maxLatency(_maxLatency)
{
rollingTotalSize = 0;
targetLatency = (maxLatency + minLatency)/2;
rate = 1.0f;
cursor = 0.0f;
curr[0] = curr[1] = 0;
kAverageSize = 80000;
}
float rate, cursor;
int minLatency, targetLatency, maxLatency;
std::queue<s16> buffer;
int size;
s16 curr[2];
std::queue<int> statsHistory;
void enqueue(s16 left, s16 right)
{
buffer.push(left);
buffer.push(right);
size++;
}
s64 rollingTotalSize;
u32 kAverageSize;
void addStatistic()
{
statsHistory.push(size);
rollingTotalSize += size;
if(statsHistory.size()>kAverageSize)
{
rollingTotalSize -= statsHistory.front();
statsHistory.pop();
float averageSize = (float)(rollingTotalSize / kAverageSize);
//static int ctr=0; ctr++; if((ctr&127)==0) printf("avg size: %f curr size: %d rate: %f\n",averageSize,size,rate);
{
float targetRate;
if(averageSize < targetLatency)
{
targetRate = 1.0f - (targetLatency-averageSize)/kAverageSize;
}
else if(averageSize > targetLatency) {
targetRate = 1.0f + (averageSize-targetLatency)/kAverageSize;
} else targetRate = 1.0f;
//rate = moveValueTowards(rate,targetRate,0.001f);
rate = targetRate;
}
}
}
void dequeue(s16& left, s16& right)
{
left = right = 0;
addStatistic();
if(size==0) { return; }
cursor += rate;
while(cursor>1.0f) {
cursor -= 1.0f;
if(size>0) {
curr[0] = buffer.front(); buffer.pop();
curr[1] = buffer.front(); buffer.pop();
size--;
}
}
left = curr[0];
right = curr[1];
}
} adjustobuf;
};
class NitsujaSynchronizer : public ISynchronizingAudioBuffer
{
private:
struct ssamp
{
s16 l, r;
ssamp() {}
ssamp(s16 ll, s16 rr) : l(ll), r(rr) {}
};
std::vector<ssamp> sampleQueue;
// returns values going between 0 and y-1 in a saw wave pattern, based on x
static FORCEINLINE int pingpong(int x, int y)
{
x %= 2*y;
if(x >= y)
x = 2*y - x - 1;
return x;
// in case we want to switch to odd buffer sizes for more sharpness
//x %= 2*(y-1);
//if(x >= y)
// x = 2*(y-1) - x;
//return x;
}
static FORCEINLINE ssamp crossfade (ssamp lhs, ssamp rhs, int cur, int start, int end)
{
if(cur <= start)
return lhs;
if(cur >= end)
return rhs;
// in case we want sine wave interpolation instead of linear here
//float ang = 3.14159f * (float)(cur - start) / (float)(end - start);
//cur = start + (int)((1-cosf(ang))*0.5f * (end - start));
int inNum = cur - start;
int outNum = end - cur;
int denom = end - start;
int lrv = ((int)lhs.l * outNum + (int)rhs.l * inNum) / denom;
int rrv = ((int)lhs.r * outNum + (int)rhs.r * inNum) / denom;
return ssamp(lrv,rrv);
}
static FORCEINLINE void emit_sample(s16*& outbuf, ssamp sample)
{
*outbuf++ = sample.l;
*outbuf++ = sample.r;
}
static FORCEINLINE void emit_samples(s16*& outbuf, const ssamp* samplebuf, int samples)
{
for(int i=0;i<samples;i++)
emit_sample(outbuf,samplebuf[i]);
}
public:
NitsujaSynchronizer()
{}
virtual void enqueue_samples(s16* buf, int samples_provided)
{
for(int i=0;i<samples_provided;i++)
{
sampleQueue.push_back(ssamp(buf[0],buf[1]));
buf += 2;
}
}
virtual int output_samples(s16* buf, int samples_requested)
{
int audiosize = samples_requested;
int queued = sampleQueue.size();
// truncate input and output sizes to multiples of 8 because I am too lazy to deal with odd numbers
audiosize &= ~7;
queued &= ~7;
if(queued > 0x200 && audiosize > 0) // is there any work to do?
{
// are we going at normal speed?
// or more precisely, are the input and output queues/buffers of similar size?
if(queued > 900 || audiosize > queued * 2)
{
// not normal speed. we have to resample it somehow in this case.
if(audiosize <= queued)
{
// fast forward speed
// this is the easy case, just crossfade it and it sounds ok
for(int i = 0; i < audiosize; i++)
{
int j = i + queued - audiosize;
ssamp outsamp = crossfade(sampleQueue[i],sampleQueue[j], i,0,audiosize);
emit_sample(buf,outsamp);
}
}
else
{
// slow motion speed
// here we take a very different approach,
// instead of crossfading it, we select a single sample from the queue
// and make sure that the index we use to select a sample is constantly moving
// and that it starts at the first sample in the queue and ends on the last one.
//
// hopefully the index doesn't move discontinuously or we'll get slight crackling
// (there might still be a minor bug here that causes this occasionally)
//
// here's a diagram of how the index we sample from moves:
//
// queued (this axis represents the index we sample from. the top means the end of the queue)
// ^
// | --> audiosize (this axis represents the output index we write to, right meaning forward in output time/position)
// | A C C end
// A A B C C C
// A A A B C C C
// A A A B C C
// A A C
// start
//
// yes, this means we are spending some stretches of time playing the sound backwards,
// but the stretches are short enough that this doesn't sound weird.
// this lets us avoid most crackling problems due to the endpoints matching up.
// TODO: it might help to calculate the approximate fundamental frequency
// and reduce either buffer size such that the reflections line up with it.
int midpointX = audiosize >> 1;
int midpointY = queued >> 1;
// all we need to do here is calculate the X position of the leftmost "B" in the above diagram.
// TODO: we should calculate it with a simple equation like
// midpointXOffset = min(something,somethingElse);
// but it's a little difficult to work it out exactly
// so here's a stupid search for the value for now:
int prevA = 999999;
int midpointXOffset = queued/2;
while(true)
{
int a = abs(pingpong(midpointX - midpointXOffset, queued) - midpointY) - midpointXOffset;
if(((a > 0) != (prevA > 0) || (a < 0) != (prevA < 0)) && prevA != 999999)
{
if((a + prevA)&1) // there's some sort of off-by-one problem with this search since we're moving diagonally...
midpointXOffset++; // but this fixes it most of the time...
break; // found it
}
prevA = a;
midpointXOffset--;
if(midpointXOffset < 0)
{
midpointXOffset = 0;
break; // failed to find it. the two sides probably meet exactly in the center.
}
}
int leftMidpointX = midpointX - midpointXOffset;
int rightMidpointX = midpointX + midpointXOffset;
int leftMidpointY = pingpong(leftMidpointX, queued);
int rightMidpointY = (queued-1) - pingpong((int)audiosize-1 - rightMidpointX + queued*2, queued);
// output the left almost-half of the sound (section "A")
for(int x = 0; x < leftMidpointX; x++)
{
int i = pingpong(x, queued);
emit_sample(buf,sampleQueue[i]);
}
// output the middle stretch (section "B")
int y = leftMidpointY;
int dyMidLeft = (leftMidpointY < midpointY) ? 1 : -1;
int dyMidRight = (rightMidpointY > midpointY) ? 1 : -1;
for(int x = leftMidpointX; x < midpointX; x++, y+=dyMidLeft)
emit_sample(buf,sampleQueue[y]);
for(int x = midpointX; x < rightMidpointX; x++, y+=dyMidRight)
emit_sample(buf,sampleQueue[y]);
// output the end of the queued sound (section "C")
for(int x = rightMidpointX; x < audiosize; x++)
{
int i = (queued-1) - pingpong((int)audiosize-1 - x + queued*2, queued);
emit_sample(buf,sampleQueue[i]);
}
} //end else
sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + queued);
return audiosize;
}
else
{
// normal speed
// just output the samples straightforwardly.
//
// at almost-full speeds (like 50/60 FPS)
// what will happen is that we rapidly fluctuate between entering this branch
// and entering the "slow motion speed" branch above.
// but that's ok! because all of these branches sound similar enough that we can get away with it.
// so the two cases actually complement each other.
if(audiosize >= queued)
{
emit_samples(buf,&sampleQueue[0],queued);
sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin() + queued);
return queued;
}
else
{
emit_samples(buf,&sampleQueue[0],audiosize);
sampleQueue.erase(sampleQueue.begin(), sampleQueue.begin()+audiosize);
return audiosize;
}
} //end normal speed
} //end if there is any work to do
else
{
return 0;
}
} //output_samples
private:
}; //NitsujaSynchronizer
#ifdef _MSC_VER
class PCSX2Synchronizer : public ISynchronizingAudioBuffer
{
public:
std::queue<s16> readySamples;
PCSX2Synchronizer()
{
SndBuffer::Init();
}
virtual void enqueue_samples(s16* buf, int samples_provided)
{
for(int i=0;i<samples_provided;i++)
{
StereoOut32 so32(buf[0],buf[1]);
SndBuffer::Write(so32);
buf++;
buf++;
}
}
virtual int output_samples(s16* buf, int samples_requested)
{
for(int i=0;i<samples_requested;i++) {
if(readySamples.size()==0) {
//SndOutPacketSize
StereoOut16 temp[SndOutPacketSize*2];
SndBuffer::ReadSamples( temp );
for(int i=0;i<SndOutPacketSize;i++) {
readySamples.push(temp[i].Left);
readySamples.push(temp[i].Right);
}
}
*buf++ = readySamples.front(); readySamples.pop();
*buf++ = readySamples.front(); readySamples.pop();
}
return samples_requested;
}
};
#endif
ISynchronizingAudioBuffer* metaspu_construct(ESynchMethod method)
{
switch(method)
{
case ESynchMethod_N: return new NitsujaSynchronizer();
case ESynchMethod_Z: return new ZeromusSynchronizer();
#ifdef _MSC_VER
case ESynchMethod_P: return new PCSX2Synchronizer();
#endif
default: return NULL;
}
}

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/* Copyright 2009 DeSmuME team
This file is part of DeSmuME
DeSmuME is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
DeSmuME is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with DeSmuME; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
//-------------------------
//this file contains the METASPU system
//which is designed to handle the task of audio synchronization
//and is designed to be as portable between multiple emulators
//-------------------------
#ifndef _METASPU_H_
#define _METASPU_H_
#include <algorithm>
template< typename T >
static FORCEINLINE void Clampify( T& src, T min, T max )
{
src = std::min( std::max( src, min ), max );
}
template< typename T >
static FORCEINLINE T GetClamped( T src, T min, T max )
{
return std::min( std::max( src, min ), max );
}
class ISynchronizingAudioBuffer
{
public:
virtual void enqueue_samples(s16* buf, int samples_provided) = 0;
//returns the number of samples actually supplied, which may not match the number requested
virtual int output_samples(s16* buf, int samples_requested) = 0;
};
enum ESynchMode
{
ESynchMode_DualSynchAsynch,
ESynchMode_Synchronous
};
enum ESynchMethod
{
ESynchMethod_N, //nitsuja's
ESynchMethod_Z, //zero's
ESynchMethod_P, //PCSX2 spu2-x
};
ISynchronizingAudioBuffer* metaspu_construct(ESynchMethod method);
#endif

134
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/* SPU2-X, A plugin for Emulating the Sound Processing Unit of the Playstation 2
* Developed and maintained by the Pcsx2 Development Team.
*
* Original portions from SPU2ghz are (c) 2008 by David Quintana [gigaherz]
*
* SPU2-X is free software: you can redistribute it and/or modify it under the terms
* of the GNU Lesser General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* SPU2-X is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with SPU2-X. If not, see <http://www.gnu.org/licenses/>.
*/
//TODO - need to make a generic way for saving these configs and add in the windows
//rc configurator
#include "types.h"
//#include "Global.h"
#include "Dialogs.h"
#include "../metaspu.h"
#include "../SoundTouch/SoundTouch.h"
static int SequenceLenMS = 63;
static int SeekWindowMS = 16;
static int OverlapMS = 7;
// Timestretch Slider Bounds, Min/Max
static const int SequenceLen_Min = 50;
static const int SequenceLen_Max = 90;
static const int SeekWindow_Min = 10;
static const int SeekWindow_Max = 32;
static const int Overlap_Min = 3;
static const int Overlap_Max = 25;
void SoundtouchCfg::ApplySettings( soundtouch::SoundTouch& sndtouch )
{
sndtouch.setSetting( SETTING_SEQUENCE_MS, SequenceLenMS );
sndtouch.setSetting( SETTING_SEEKWINDOW_MS, SeekWindowMS );
sndtouch.setSetting( SETTING_OVERLAP_MS, OverlapMS );
}
static void ClampValues()
{
Clampify( SequenceLenMS, SequenceLen_Min, SequenceLen_Max );
Clampify( SeekWindowMS, SeekWindow_Min, SeekWindow_Max );
Clampify( OverlapMS, Overlap_Min, Overlap_Max );
}
void SoundtouchCfg::ReadSettings()
{
/* SequenceLenMS = CfgReadInt( L"SOUNDTOUCH", L"SequenceLengthMS", 50 );
SeekWindowMS = CfgReadInt( L"SOUNDTOUCH", L"SeekWindowMS", 15 );
OverlapMS = CfgReadInt( L"SOUNDTOUCH", L"OverlapMS", 25 );
ClampValues(); */
}
void SoundtouchCfg::WriteSettings()
{
//CfgWriteInt( L"SOUNDTOUCH", L"SequenceLengthMS", SequenceLenMS );
//CfgWriteInt( L"SOUNDTOUCH", L"SeekWindowMS", SeekWindowMS );
//CfgWriteInt( L"SOUNDTOUCH", L"OverlapMS", OverlapMS );
}
//BOOL CALLBACK SoundtouchCfg::DialogProc(HWND hWnd,UINT uMsg,WPARAM wParam,LPARAM lParam)
//{
// int wmId,wmEvent;
// wchar_t temp[384]={0};
//
// switch(uMsg)
// {
// case WM_PAINT:
// return FALSE;
//
// case WM_INITDIALOG:
// {
// INIT_SLIDER( IDC_SEQLEN_SLIDER, SequenceLen_Min, SequenceLen_Max, 20, 5, 1 );
// INIT_SLIDER( IDC_SEEKWIN_SLIDER, SeekWindow_Min, SeekWindow_Max, 5, 2, 1 );
// INIT_SLIDER( IDC_OVERLAP_SLIDER, Overlap_Min, Overlap_Max, 3, 2, 1 );
//
// SendDialogMsg( hWnd, IDC_SEQLEN_SLIDER, TBM_SETPOS, TRUE, SequenceLenMS );
// SendDialogMsg( hWnd, IDC_SEEKWIN_SLIDER, TBM_SETPOS, TRUE, SeekWindowMS );
// SendDialogMsg( hWnd, IDC_OVERLAP_SLIDER, TBM_SETPOS, TRUE, OverlapMS );
// }
//
// case WM_COMMAND:
// wmId = LOWORD(wParam);
// wmEvent = HIWORD(wParam);
// // Parse the menu selections:
// if( wmId == IDOK )
// {
// SequenceLenMS = (int)SendDialogMsg( hWnd, IDC_SEQLEN_SLIDER, TBM_GETPOS, 0, 0 );
// SeekWindowMS = (int)SendDialogMsg( hWnd, IDC_SEEKWIN_SLIDER, TBM_GETPOS, 0, 0 );
// OverlapMS = (int)SendDialogMsg( hWnd, IDC_OVERLAP_SLIDER, TBM_GETPOS, 0, 0 );
//
// ClampValues();
// WriteSettings();
// EndDialog(hWnd,0);
// }
// else if( wmId == IDCANCEL )
// {
// EndDialog(hWnd,0);
// }
// break;
//
// case WM_HSCROLL:
// DoHandleScrollMessage( hWnd, wParam, lParam );
// break;
//
// default:
// return FALSE;
// }
// return TRUE;
//}
void SoundtouchCfg::OpenDialog( HWND hWnd )
{
//INT_PTR ret;
//ret = DialogBox( hInstance, MAKEINTRESOURCE(IDD_CONFIG_SOUNDTOUCH), hWnd, (DLGPROC)DialogProc );
//if(ret==-1)
//{
// MessageBoxEx(GetActiveWindow(), L"Error Opening the Soundtouch advanced dialog.", L"OMG ERROR!", MB_OK, 0);
// return;
//}
//ReadSettings();
}

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/* SPU2-X, A plugin for Emulating the Sound Processing Unit of the Playstation 2
* Developed and maintained by the Pcsx2 Development Team.
*
* Original portions from SPU2ghz are (c) 2008 by David Quintana [gigaherz]
*
* SPU2-X is free software: you can redistribute it and/or modify it under the terms
* of the GNU Lesser General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* SPU2-X is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with SPU2-X. If not, see <http://www.gnu.org/licenses/>.
*/
#pragma once
#ifdef _WIN32
#include <windows.h>
#endif
#include "../SoundTouch/SoundTouch.h"
//#ifdef _WIN32
//# include "WinConfig.h"
//#else
//# include "LnxConfig.h"
//#endif
//
//namespace DebugConfig
//{
// extern void ReadSettings();
// extern void WriteSettings();
// extern void OpenDialog();
// extern void EnableControls( HWND hWnd );
//}
//
namespace SoundtouchCfg
{
extern void ReadSettings();
extern void WriteSettings();
extern void OpenDialog( HWND hWnd );
extern BOOL CALLBACK DialogProc(HWND hWnd,UINT uMsg,WPARAM wParam,LPARAM lParam);
extern void ApplySettings( soundtouch::SoundTouch& sndtouch );
}
//
//extern int SendDialogMsg( HWND hwnd, int dlgId, UINT code, WPARAM wParam, LPARAM lParam);
//extern HRESULT GUIDFromString( const char *str, LPGUID guid );
//
//extern void AssignSliderValue( HWND idcwnd, HWND hwndDisplay, int value );
//extern void AssignSliderValue( HWND hWnd, int idc, int editbox, int value );
//extern int GetSliderValue( HWND hWnd, int idc );
//extern BOOL DoHandleScrollMessage( HWND hwndDisplay, WPARAM wParam, LPARAM lParam );
//
//extern bool CfgFindName( const TCHAR *Section, const TCHAR* Name);
//
//extern void CfgWriteBool(const TCHAR* Section, const TCHAR* Name, bool Value);
//extern void CfgWriteInt(const TCHAR* Section, const TCHAR* Name, int Value);
//extern void CfgWriteStr(const TCHAR* Section, const TCHAR* Name, const wstring& Data);
//
//extern bool CfgReadBool(const TCHAR *Section,const TCHAR* Name, bool Default);
//extern void CfgReadStr(const TCHAR* Section, const TCHAR* Name, wstring& Data, int DataSize, const TCHAR* Default);
//extern void CfgReadStr(const TCHAR* Section, const TCHAR* Name, TCHAR* Data, int DataSize, const TCHAR* Default);
//extern int CfgReadInt(const TCHAR* Section, const TCHAR* Name,int Default);
//
//
//// Items Specific to DirectSound
//#define STRFY(x) #x
//#define verifyc(x) Verifyc(x,STRFY(x))
//
//extern void Verifyc(HRESULT hr, const char* fn);
//
//struct ds_device_data
//{
// std::wstring name;
// GUID guid;
// bool hasGuid;
//};
//

128
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@ -1,4 +1,4 @@
<?xml version="1.0" encoding="shift_jis"?>
<?xml version="1.0" encoding="Windows-1252"?>
<VisualStudioProject
ProjectType="Visual C++"
Version="8.00"
@ -274,7 +274,7 @@
/>
<Tool
Name="VCLinkerTool"
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@ -109,193 +109,6 @@
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3
desmume/src/windows/main.cpp
View File

@ -4831,6 +4831,7 @@ static void SoundSettings_updateSynchMode(HWND hDlg)
EnableWindow(GetDlgItem(hDlg,IDC_GROUP_SYNCHMETHOD),en);
EnableWindow(GetDlgItem(hDlg,IDC_SYNCHMETHOD_N),en);
EnableWindow(GetDlgItem(hDlg,IDC_SYNCHMETHOD_Z),en);
EnableWindow(GetDlgItem(hDlg,IDC_SYNCHMETHOD_P),en);
}
static LRESULT CALLBACK SoundSettingsDlgProc(HWND hDlg, UINT uMsg, WPARAM wParam, LPARAM lParam)
@ -4865,6 +4866,7 @@ static LRESULT CALLBACK SoundSettingsDlgProc(HWND hDlg, UINT uMsg, WPARAM wParam
//update the synch method
CheckDlgItem(hDlg,IDC_SYNCHMETHOD_N,snd_synchmethod==0);
CheckDlgItem(hDlg,IDC_SYNCHMETHOD_Z,snd_synchmethod==1);
CheckDlgItem(hDlg,IDC_SYNCHMETHOD_P,snd_synchmethod==2);
//setup interpolation combobox
SendDlgItemMessage(hDlg, IDC_SPU_INTERPOLATION_CB, CB_RESETCONTENT, 0, 0);
@ -4948,6 +4950,7 @@ static LRESULT CALLBACK SoundSettingsDlgProc(HWND hDlg, UINT uMsg, WPARAM wParam
//save the synch method
if(IsDlgCheckboxChecked(hDlg,IDC_SYNCHMETHOD_N)) snd_synchmethod = 0;
if(IsDlgCheckboxChecked(hDlg,IDC_SYNCHMETHOD_Z)) snd_synchmethod = 1;
if(IsDlgCheckboxChecked(hDlg,IDC_SYNCHMETHOD_P)) snd_synchmethod = 2;
WritePrivateProfileInt("Sound", "SynchMethod", snd_synchmethod, IniName);
SPU_SetSynchMode(snd_synchmode, snd_synchmethod);

1
desmume/src/windows/resource.h
View File

@ -351,6 +351,7 @@
#define IDC_GROUP_SYNCHMETHOD 1017
#define IDC_AUTOUPDATE_ASM 1018
#define IDC_BGMAP_PRIO 1018
#define IDC_SYNCHMETHOD_P 1018
#define IDC_BGMAP_PAL 1019
#define IDC_VISIBLE 1019
#define IDC_BGMAP_SIZE 1020

BIN
desmume/src/windows/resources.rc
View File

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