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Rewritten sound resampler.

Browse Source 
This eliminates the (optional) libresample dependency, which will clean up
building and eliminate the goofy 5k DLL.  It's also faster.

This doesn't implement "HighQuality".  That would involve changing L from
8 to 16.  I might implement that in the future, but it sounds decent to me.  (It's
a little tricky, since PolyphaseFilter::RunPolyphaseFilter needs L to be
a constant for speed.)
This commit is contained in:
Glenn Maynard
2006-02-13 04:50:28 +00:00
parent d76475ce79
commit 5d64dff234
2 changed files with 565 additions and 183 deletions
Download Patch File Download Diff File Expand all files Collapse all files
stepmania/src/RageSoundReader_Resample_Good.cpp
+549 -155
View File
@@ -1,243 +1,638 @@
/*
* This implements audio resampling, using the method described at:
* http://www.dspguru.com/info/faqs/mrfaq.htm
*
* Each conversion ratio uses some memory, but the resulting table is
* shared, so the memory overhead per stream is negligible.
*/
#include "global.h" #include "global.h"
#include "RageSoundReader_Resample_Good.h"
#include "RageLog.h" #include "RageLog.h"
#include "RageUtil.h" #include "RageUtil.h"
#include "RageSoundReader_Resample_Good.h" #include "RageMath.h"
#include "RageThreads.h"
#include "libresample/include/libresample.h" #include <numeric>
#if defined(_MSC_VER) && !defined(_XBOX)
#pragma comment(lib, "libresample/resample.lib") /* Filter length. This must be a power of 2. */
#define L 8
namespace
{
float sincf( float f )
{
if( f == 0 )
return 1;
return sinf(f)/f;
}
/* Modified Bessel function I0. From Abramowitz and Stegun "Handbook of Mathematical
* Functions", "Modified Bessel Functions I and K". */
float BesselI0( float fX )
{
float fAbsX = fabsf( fX );
if( fAbsX < 3.75f )
{
float y = fX / 3.75f;
y *= y;
float fRet = 1.0f+y*(+3.5156229f+y*(+3.0899424f+y*(+1.2067492f+y*(+0.2659732f+y*(+0.0360768f+y*+0.0045813f)))));
return fRet;
}
else
{
float y = 3.75f/fAbsX;
float fRet = (exp(fAbsX)/sqrt(fAbsX)) *
(+0.39894228f+y*(+0.01328592f+y*(+0.00225319f+y*(-0.00157565f+y*(0.00916281f+
y*(-0.02057706f+y*(+0.02635537f+y*(-0.01647633f+y*+0.00392377f))))))));
return fRet;
}
}
/*
* Kaiser window:
*
* K(n) = I0( B*sqrt(1-(n/p)^2) )
* -----------------------
* I0(B)
*
* where B is the beta parameter, p is len/2, and n is in [-len/2,+len/2].
*/
void ApplyKaiserWindow( float *pBuf, int iLen, float fBeta )
{
const float fDenom = BesselI0(fBeta);
float p = (iLen-1)/2.0f;
for( int n = 0; n < iLen; ++n )
{
float fN1 = fabsf((n-p)/p);
float fNum = fBeta * sqrtf( max(1-fN1*fN1, 0) );
fNum = BesselI0( fNum );
float fVal = fNum/fDenom;
pBuf[n] *= fVal;
}
}
void MultiplyVector( float *pStart, float *pEnd, float f )
{
for( ; pStart != pEnd; ++pStart )
*pStart *= f;
}
void GenerateSincLowPassFilter( float *pFIR, int iWinSize, float fCutoff )
{
float p = (iWinSize-1)/2.0f;
for( int n = 0; n < iWinSize; ++n )
{
float fN1 = (n-p);
float fVal = sincf(2*PI*fCutoff * fN1)*(2*fCutoff);
// printf( "n %i, %f, %f -> %f\n", n, p, fN1, fVal );
pFIR[n] = fVal;
}
#if 0
float *pFIRp = pFIR+iWinSize/2;
for(int i=-iWinSize/2;i<=iWinSize/2;i++)
{
float ff = sinc(2*M_PI*fCutoff * (i + 0.0))*(2*fCutoff);
printf( "%i: %f\n", i, ff );
pFIRp[i]=ff;
}
for( int i=0; i < iWinSize; i++ )
printf( "sinc: %i: %f\n", i, pFIR[i] );
#endif
}
void NormalizeVector( float *pBuf, int iSize )
{
float fTotal = accumulate( &pBuf[0], &pBuf[iSize], 0.0f );
MultiplyVector( &pBuf[0], &pBuf[iSize], 1/fTotal );
}
int GCD( int i1, int i2 )
{
while(1)
{
unsigned iRem = i2 % i1;
if( iRem == 0 )
return i1;
i2 = i1;
i1 = iRem;
}
return i1;
}
}
#if 0
void RunFIRFilter( float *pIn, float *pOut, int iInputValues, float *pFIR, int iWinSize )
{
for( int i = 0; i < iInputValues; ++i )
{
float fSum = 0;
const float *pInData = &pIn[i];
for( int j = 0; j < iWinSize; ++j )
{
float in = pInData[j];
fSum += in*pFIR[j];
printf( "%i: in %f * %f, += %f\n", j, pInData[j], pFIR[j], in*pFIR[j] );
}
pOut[i] = fSum;
}
}
#endif #endif
#ifdef _XBOX template<typename T>
class AlignedBuffer
{
public:
AlignedBuffer( int iSize )
{
m_iSize = iSize;
m_pBuf = new T[m_iSize];
}
#ifdef _DEBUG AlignedBuffer( const AlignedBuffer &cpy )
#pragma comment(lib, "libresample/xboxresample/debug/xboxresample.lib") {
#else m_iSize = cpy.m_iSize;
#pragma comment(lib, "libresample/xboxresample/release/xboxresample.lib") m_pBuf = new T[m_iSize];
memcpy( m_pBuf, cpy.m_pBuf, sizeof(T)*m_iSize );
}
~AlignedBuffer()
{
delete [] m_pBuf;
}
operator T*() { return m_pBuf; }
operator const T*() const { return m_pBuf; }
private:
T& operator=( T &rhs );
int m_iSize;
T *m_pBuf;
};
struct PolyphaseFilter
{
struct State
{
State( const PolyphaseFilter &Target ):
m_fBuf( L * 2 )
{
m_iPolyIndex = Target.m_iUpFactor-1;
m_iFilled = 0;
m_iBufNext = 0;
}
int m_iPolyIndex;
int m_iFilled;
/* This buffer is duplicated. If the circular buffer is size L, the actual buffer
* is size L*2, and data at buf[N] is also at buf[N+L]. That way, we can access
* up to buf[N*2-1] without having to wrap. */
AlignedBuffer<float> m_fBuf;
int m_iBufNext;
};
PolyphaseFilter( int iUpFactor ):
m_pPolyphase( L*iUpFactor )
{
m_iUpFactor = iUpFactor;
}
void Generate( const float *pFIR );
int RunPolyphaseFilter( State &State, const float *pIn, int iSamplesIn, int iDownFactor,
float *pOut, int iSamplesOut ) const;
int GetLatency() const { return L/2; }
int NumInputsForOutputSamples( const State &State, int iOut, int iDownFactor ) const;
private:
AlignedBuffer<float> m_pPolyphase;
int m_iUpFactor;
};
/*
* Convert an FIR filter to a polyphase filter.
*
* pFIR is the input FIR filter, which has iL*iUpFactor values.
* iL is the number of real samples each output sample looks at.
* iUpFactor is the actual upsampling factor; the amount of zero-stuffing between each real sample.
* pOutput is the 2D output polyphase filter, with iL*iL values.
*
* With an upsampling factor (iUpFactor) of 3, and a sinc filter length of 12 (iL*iUpFactor),
*
* input first output sample (before decimation)
* sample second output sample
* third output sample
*
* 0 0
* 0 1 0
* 1592 2 1 0
* 0 3 2 1
* 0 4 3 2
* 1623 5 4 3
* 0 6 5 4
* 0 7 6 5
* 1682 8 7 6
* 0 9 8 7
* 0 10 9 8
* 1730 11 10 9
* 0 11 10
* 0 11
*
* first row: 2, 5, 8, 11
* second: 1, 4, 7, 10
* third: 0, 3, 6, 9
* Read a new sample after passing the last line.
*/
void PolyphaseFilter::Generate( const float *pFIR )
{
float *pOutput=m_pPolyphase;
int iInputSize = L*m_iUpFactor;
for( int iRow = 0; iRow < m_iUpFactor; ++iRow )
{
int iInputOffset = (m_iUpFactor-iRow-1) % m_iUpFactor;
for( int iCol = 0; iCol < L; ++iCol )
{
*pOutput = pFIR[iInputOffset];
++pOutput;
iInputOffset += m_iUpFactor;
iInputOffset %= iInputSize;
}
}
}
/*
* We only want one boundary check when running the filter; either on the
* number of inputs used, or the number of outputs produced. Otherwise, we'll
* have to maintain two counters, and check two values per iteration.
*
* First, call NumInputsForOutputSamples(out), to find out how many inputs to supply to get
* the desired number of outputs. Then, pass the data, the input count
* and the output count to RunPolyphaseFilter.
*
* - When downsampling, we use the number of inputs as the boundary. For example,
* if the ratio is 1:3 (downsample x3), and the user gives us 10 samples, then we
* process until we've consumed all of the input. (This will result in exactly
* the number of samples the user asked for with NumInputsForOutputSamples.)
*
* - When upsampling, we use the number of outputs as the boundary. For example,
* if the ratio is 3:1 (upsample x3), and the user wants 8 samples to be output,
* we'll have been given 3 samples as input. Process until we've produced 8
* samples.
*
* In both cases, we have overlap. In the first, it's possible that we could
* have consumed an additional input without producing an output. In the second,
* it's possible that we could have produced an additional output without
* consuming an input.
*/
int PolyphaseFilter::RunPolyphaseFilter(
State &State,
const float *pIn, int iSamplesIn, int iDownFactor,
float *pOut, int iSamplesOut ) const
{
ASSERT( iSamplesIn >= 0 );
float *pOutOrig = pOut;
const float *pInEnd = pIn + iSamplesIn;
const float *pOutEnd = pOut + iSamplesOut;
int iFilled = State.m_iFilled;
int iPolyIndex = State.m_iPolyIndex;
while( pOut != pOutEnd )
{
if( iFilled < L )
{
if( pIn == pInEnd )
break;
State.m_fBuf[State.m_iBufNext] = *pIn;
State.m_fBuf[State.m_iBufNext + L] = *pIn;
++State.m_iBufNext;
State.m_iBufNext &= L-1;
++pIn;
++iFilled;
continue;
}
while( pOut != pOutEnd )
{
const float *pCurPoly = &m_pPolyphase[iPolyIndex*L];
const float *pInData = &State.m_fBuf[State.m_iBufNext];
float fTot = 0;
for( int j = 0; j < L; ++j )
fTot += pInData[j]*pCurPoly[j];
*pOut = fTot;
++pOut;
iPolyIndex += iDownFactor;
if( iPolyIndex >= m_iUpFactor )
break;
}
iFilled -= iPolyIndex/m_iUpFactor;
iPolyIndex %= m_iUpFactor;
}
State.m_iFilled = iFilled;
State.m_iPolyIndex = iPolyIndex;
return pOut - pOutOrig;
}
/*
* Return the number of input samples needed to produce the given number of output
* samples. This is dependent on the number of bytes in the buffer and the current
* position of the stream.
*/
int PolyphaseFilter::NumInputsForOutputSamples( const State &State, int iOut, int iDownFactor ) const
{
int iIn = 0;
int iFilled = State.m_iFilled;
int iPolyIndex = State.m_iPolyIndex;
#if 0
while( iOut > 0 )
{
if( iFilled < L )
{
int iToFill = L-iFilled;
iIn += iToFill;
iFilled += iToFill;
}
while( iFilled == L && iOut )
{
--iOut;
iPolyIndex += iDownFactor;
if( iPolyIndex >= m_iUpFactor )
break;
}
iFilled -= iPolyIndex/m_iUpFactor;
iPolyIndex %= m_iUpFactor;
}
#endif #endif
#endif if( iOut > 0 )
{
if( iFilled < L )
{
int iToFill = L-iFilled;
iIn += iToFill;
}
#include "RageTimer.h" // The -1 here is because we don't refill m_fBuf after writing the last output.
iPolyIndex += iDownFactor*(iOut-1);
iIn += iPolyIndex/m_iUpFactor;
}
#define channels source->GetNumChannels() return iIn;
}
/*
* Interface to PolyphaseFilter, providing a simple resampling interface. This handles
* reuse of PolyphaseFilters. This does not handle delay, flushing, or multiple channels.
*/
class RageSoundResampler_Polyphase
{
public:
RageSoundResampler_Polyphase( int iSourceRate, int iDestRate )
{
int iUpFactor = iDestRate;
m_iDownFactor = iSourceRate;
{
int iGCD = GCD( iUpFactor, m_iDownFactor );
iUpFactor /= iGCD;
m_iDownFactor /= iGCD;
}
float fCutoffFrequency;
{
/*
* If we're upsampling, we want the low-pass filter to cut off at the
* nyquist frequency of the original sample.
*
* If we're downsampling, we want the low-pass filter to cut off at the
* nyquist frequency of the new sample.
*/
fCutoffFrequency = 1.0f / (2*iUpFactor);
fCutoffFrequency = min( fCutoffFrequency, 1.0f / (2*m_iDownFactor) );
LOG->Trace( "cutoff frequency %f -> %f, %f", fCutoffFrequency, 1.0f / (2*iUpFactor), 1.0f / (2*m_iDownFactor) );
}
/* Cache filter data, and reuse it without copying. All operations after creation
* are const, so this doesn't cause thread-safety problems. */
typedef map<pair<int,float>, PolyphaseFilter *> FilterMap;
static RageMutex PolyphaseFiltersLock("PolyphaseFiltersLock");
static FilterMap g_mapPolyphaseFilters;
PolyphaseFiltersLock.Lock();
pair<int,float> params( make_pair(iUpFactor, fCutoffFrequency) );
FilterMap::const_iterator it = g_mapPolyphaseFilters.find(params);
if( it == g_mapPolyphaseFilters.end() )
{
int iWinSize = L*iUpFactor;
float *pFIR = new float[iWinSize];
GenerateSincLowPassFilter( pFIR, iWinSize, fCutoffFrequency );
ApplyKaiserWindow( pFIR, iWinSize, 8 );
NormalizeVector( pFIR, iWinSize );
MultiplyVector( &pFIR[0], &pFIR[iWinSize], (float) iUpFactor );
PolyphaseFilter *pPolyphase = new PolyphaseFilter( iUpFactor );
pPolyphase->Generate( pFIR );
delete [] pFIR;
g_mapPolyphaseFilters[params] = pPolyphase;
m_pPolyphase = pPolyphase;
}
else
{
/* We already have a filter for this upsampling factor and cutoff; use it. */
m_pPolyphase = it->second;
}
PolyphaseFiltersLock.Unlock();
m_pState = new PolyphaseFilter::State( *m_pPolyphase );
}
~RageSoundResampler_Polyphase()
{
delete m_pState;
}
int Run( const float *pIn, int iSamplesIn, float *pOut, int iSamplesOut ) const
{
return m_pPolyphase->RunPolyphaseFilter( *m_pState, pIn, iSamplesIn, m_iDownFactor, pOut, iSamplesOut );
}
void Reset()
{
delete m_pState;
m_pState = new PolyphaseFilter::State( *m_pPolyphase );
}
int NumInputsForOutputSamples( int iOut ) const { return m_pPolyphase->NumInputsForOutputSamples(*m_pState, iOut, m_iDownFactor); }
int GetLatency() const { return m_pPolyphase->GetLatency(); }
RageSoundResampler_Polyphase( const RageSoundResampler_Polyphase &cpy )
{
m_pPolyphase = new PolyphaseFilter(*cpy.m_pPolyphase);
m_pState = new PolyphaseFilter::State(*cpy.m_pState);
m_iDownFactor = cpy.m_iDownFactor;
}
private:
const PolyphaseFilter *m_pPolyphase;
PolyphaseFilter::State *m_pState;
int m_iDownFactor;
};
RageSoundReader_Resample_Good::RageSoundReader_Resample_Good() RageSoundReader_Resample_Good::RageSoundReader_Resample_Good()
{ {
source = NULL; m_pSource = NULL;
empty_resamp = NULL; m_iSampleRate = -1;
samplerate = -1;
BufSamples = 0;
eof = false;
HighQuality = false;
} }
/* Call this if the input position is changed or reset. */ /* Call this if the input position is changed or reset. */
void RageSoundReader_Resample_Good::Reset() void RageSoundReader_Resample_Good::Reset()
{ {
BufSamples = 0; for( size_t iChannel = 0; iChannel < m_pSource->GetNumChannels(); ++iChannel )
eof = false; resamplers[iChannel]->Reset();
/* Flush the resampler. */
for( unsigned i = 0; i < resamplers.size(); ++i )
{
resample_channel &r = resamplers[i];
if( r.resamp )
resample_close( r.resamp );
r.resamp = resample_dup( empty_resamp );
}
} }
/* Call this if the sample factor changes. */ /* Call this if the sample factor changes. */
void RageSoundReader_Resample_Good::ReopenResampler() void RageSoundReader_Resample_Good::ReopenResampler()
{ {
if( empty_resamp ) for( size_t iChannel = 0; iChannel < resamplers.size(); ++iChannel )
resample_close( empty_resamp ); delete resamplers[iChannel];
empty_resamp = resample_open( HighQuality, GetFactor()-0.1f, GetFactor()+0.1f ); resamplers.clear();
for( size_t iChannel = 0; iChannel < m_pSource->GetNumChannels(); ++iChannel )
for( unsigned i = 0; i < resamplers.size(); ++i )
{ {
resample_channel &r = resamplers[i]; RageSoundResampler_Polyphase *p = new RageSoundResampler_Polyphase( m_pSource->GetSampleRate(), m_iSampleRate );
if( r.resamp ) resamplers.push_back( p );
resample_close( r.resamp );
r.resamp = resample_dup( empty_resamp );
} }
} }
void RageSoundReader_Resample_Good::Open(SoundReader *source_) void RageSoundReader_Resample_Good::Open( SoundReader *pSource )
{ {
source = source_; ASSERT(pSource);
ASSERT(source); m_pSource = pSource;
samplerate = source->GetSampleRate();
for( unsigned i = 0; i < source->GetNumChannels(); ++i )
resamplers.push_back( resample_channel() );
} }
RageSoundReader_Resample_Good::~RageSoundReader_Resample_Good() RageSoundReader_Resample_Good::~RageSoundReader_Resample_Good()
{ {
for( unsigned i = 0; i < resamplers.size(); ++i ) for( size_t iChannel = 0; iChannel < resamplers.size(); ++iChannel )
{ delete resamplers[iChannel];
if( resamplers[i].resamp ) delete m_pSource;
resample_close( resamplers[i].resamp );
}
if( empty_resamp )
resample_close( empty_resamp );
delete source;
} }
float RageSoundReader_Resample_Good::GetFactor() const void RageSoundReader_Resample_Good::SetSampleRate( int iHZ )
{ {
return float(samplerate) / source->GetSampleRate(); m_iSampleRate = iHZ;
}
void RageSoundReader_Resample_Good::SetSampleRate(int hz)
{
samplerate = hz;
ReopenResampler(); ReopenResampler();
} }
int RageSoundReader_Resample_Good::GetLength() const int RageSoundReader_Resample_Good::GetLength() const
{ {
return source->GetLength(); return m_pSource->GetLength();
} }
int RageSoundReader_Resample_Good::GetLength_Fast() const int RageSoundReader_Resample_Good::GetLength_Fast() const
{ {
return source->GetLength_Fast(); return m_pSource->GetLength_Fast();
} }
int RageSoundReader_Resample_Good::SetPosition_Accurate(int ms) int RageSoundReader_Resample_Good::SetPosition_Accurate(int ms)
{ {
Reset(); Reset();
return source->SetPosition_Accurate(ms); return m_pSource->SetPosition_Accurate(ms);
} }
int RageSoundReader_Resample_Good::SetPosition_Fast(int ms) int RageSoundReader_Resample_Good::SetPosition_Fast(int ms)
{ {
Reset(); Reset();
return source->SetPosition_Fast(ms); return m_pSource->SetPosition_Fast(ms);
} }
bool RageSoundReader_Resample_Good::FillBuf() int RageSoundReader_Resample_Good::Read( char *bufp, unsigned len )
{ {
int samples_free = BUFSIZE-BufSamples; int iChannels = resamplers.size();
if( eof ) int iBytesPerFrame = sizeof(int16_t) * iChannels;
return true;
if( !samples_free )
return true;
const int bytes_per_frame = sizeof(int16_t)*channels; int iFrames = len / iBytesPerFrame; /* bytes -> frames */
int16_t *tmpbuf = (int16_t *) alloca( BUFSIZE*bytes_per_frame ); int16_t *pBuf = (int16_t *) bufp;
int cnt = source->Read( (char *) tmpbuf, samples_free * bytes_per_frame );
int iFramesRead = 0;
if( cnt == -1 )
{ {
SetError(source->GetError()); int iFramesNeeded = resamplers[0]->NumInputsForOutputSamples(iFrames);
return false; int iBytesNeeded = iFramesNeeded * sizeof(int16_t) * iChannels;
int16_t *pTmpBuf = (int16_t *) alloca( iBytesNeeded );
ASSERT( pTmpBuf );
int iBytesIn = m_pSource->Read( (char *) pTmpBuf, iBytesNeeded );
if( iBytesIn == -1 )
{
SetError( m_pSource->GetError() );
return -1;
} }
if( cnt < samples_free * bytes_per_frame ) iBytesNeeded -= iBytesIn;
eof = true; iFramesNeeded -= iBytesIn / (sizeof(int16_t) * iChannels);
cnt /= bytes_per_frame; const int iSamplesIn = iBytesIn / sizeof(int16_t);
const int iFramesIn = iSamplesIn / iChannels;
for( unsigned i = 0; i < channels; ++i ) float *pFloatBuf = (float *) alloca( iSamplesIn * sizeof(float) );
float *pFloatOut = (float *) alloca( iFrames * sizeof(float) );
for( int iChannel = 0; iChannel < iChannels; ++iChannel )
{ {
resample_channel &r = resamplers[i];
for( int s = 0; s < cnt; ++s )
r.inbuf[s+BufSamples] = tmpbuf[s*resamplers.size()+i];
}
BufSamples += cnt;
return true;
}
int RageSoundReader_Resample_Good::Read(char *bufp, unsigned len)
{
int16_t *buf = (int16_t *) bufp;
len /= sizeof(int16_t); /* bytes -> samples */
const float factor = GetFactor();
int bytes_read = 0;
while( 1 )
{ {
int samples_used = 0, samples_output = 0; int16_t *pBufIn = pTmpBuf + iChannel;
if( BufSamples ) float *pBufOut = pFloatBuf;
{ for( int i = 0; i < iSamplesIn; i += iChannels )
for( unsigned i = 0; i < channels; ++i ) *(pBufOut++) = (float) pBufIn[i];
{
resample_channel &r = resamplers[i];
ASSERT( r.resamp );
float outbuf[BUFSIZE];
samples_output = resample_process( r.resamp,
factor,
r.inbuf, BufSamples,
eof,
&samples_used,
outbuf, len/channels);
if( samples_output == -1 )
RageException::Throw( "Unexpected resample_process return value: -1" );
memmove( r.inbuf, &r.inbuf[samples_used], sizeof(float) * (BufSamples-samples_used) );
for( int s = 0; s < samples_output; ++s )
{
buf[s*channels+i] = int16_t(clamp(outbuf[s], -32768, 32767));
}
}
} }
BufSamples -= samples_used; int iGotFrames = resamplers[iChannel]->Run( pFloatBuf, iFramesIn, pFloatOut, iFrames );
ASSERT( iGotFrames <= iFrames );
if( !samples_output ) int16_t *pBufOut = pBuf + iChannel;
for( int i = 0; i < iGotFrames; ++i )
{ {
if( !len ) *pBufOut = int16_t(lrintf(clamp(pFloatOut[i], -32768, 32767)));
return bytes_read; /* buffer full */ pBufOut += iChannels;
if( eof ) }
return bytes_read; /* EOF and we're completely flushed */ if( iChannel == 0 )
if( !FillBuf() ) iFramesRead += iGotFrames;
return -1; /* source error */ }
} }
len -= samples_output*channels; return iFramesRead * iBytesPerFrame;
buf += samples_output*channels;
bytes_read += samples_output*channels*sizeof(int16_t);
}
} }
SoundReader *RageSoundReader_Resample_Good::Copy() const SoundReader *RageSoundReader_Resample_Good::Copy() const
{ {
SoundReader *new_source = source->Copy(); SoundReader *pSource = m_pSource->Copy();
RageSoundReader_Resample_Good *ret = new RageSoundReader_Resample_Good; RageSoundReader_Resample_Good *ret = new RageSoundReader_Resample_Good;
for( unsigned i = 0; i < channels; ++i ) for( size_t i = 0; i < resamplers.size(); ++i )
{ ret->resamplers.push_back( new RageSoundResampler_Polyphase(*resamplers[i]) );
const resample_channel &r = resamplers[i]; ret->m_pSource = pSource;
ASSERT( r.resamp ); ret->m_iSampleRate = m_iSampleRate;
ret->resamplers.push_back( resample_channel() );
ret->resamplers[i].resamp = resample_dup( r.resamp );
memcpy( ret->resamplers[i].inbuf, r.inbuf, sizeof(r.inbuf));
}
ret->empty_resamp = resample_dup( empty_resamp );
ret->source = new_source;
ret->HighQuality = HighQuality;
ret->samplerate = samplerate;
ret->BufSamples = BufSamples;
ret->eof = eof;
// ret->Open(new_source);
// ret->SetSampleRate(samplerate);
return ret; return ret;
} }
/* /*
* Copyright (c) 2003 Glenn Maynard * (c) 2006 Glenn Maynard
* All rights reserved. * All rights reserved.
* *
* Permission is hereby granted, free of charge, to any person obtaining a * Permission is hereby granted, free of charge, to any person obtaining a
@@ -260,4 +655,3 @@ SoundReader *RageSoundReader_Resample_Good::Copy() const
* OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR * OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
* PERFORMANCE OF THIS SOFTWARE. * PERFORMANCE OF THIS SOFTWARE.
*/ */
stepmania/src/RageSoundReader_Resample_Good.h
+13 -25
View File
@@ -1,4 +1,4 @@
/* RageSoundReader_Resample_Fast - interface for libresample. */ /* RageSoundReader_Resample_Good - fast audio resampling. */
#ifndef RAGE_SOUND_READER_RESAMPLE_GOOD_H #ifndef RAGE_SOUND_READER_RESAMPLE_GOOD_H
#define RAGE_SOUND_READER_RESAMPLE_GOOD_H #define RAGE_SOUND_READER_RESAMPLE_GOOD_H
@@ -6,6 +6,8 @@
#include "RageSoundReader.h" #include "RageSoundReader.h"
#include "RageSoundReader_Resample.h" #include "RageSoundReader_Resample.h"
class RageSoundResampler_Polyphase;
/* This class changes the sampling rate of a sound. */ /* This class changes the sampling rate of a sound. */
class RageSoundReader_Resample_Good: public RageSoundReader_Resample class RageSoundReader_Resample_Good: public RageSoundReader_Resample
{ {
@@ -23,40 +25,26 @@ public:
/* Change the actual sample rate of a sound. */ /* Change the actual sample rate of a sound. */
void SetSampleRate( int hz ); void SetSampleRate( int hz );
void SetHighQuality( bool hq ) { HighQuality = hq; } void SetHighQuality( bool hq ) { }
int GetSampleRate() const { return samplerate; } int GetSampleRate() const { return m_iSampleRate; }
unsigned GetNumChannels() const { return source->GetNumChannels(); } unsigned GetNumChannels() const { return m_pSource->GetNumChannels(); }
bool IsStreamingFromDisk() const { return source->IsStreamingFromDisk(); } bool IsStreamingFromDisk() const { return m_pSource->IsStreamingFromDisk(); }
enum { BUFSIZE = 4096 };
private: private:
SoundReader *source;
bool HighQuality;
int samplerate;
void *empty_resamp;
struct resample_channel
{
resample_channel(): resamp(NULL) { }
void *resamp;
float inbuf[BUFSIZE];
};
vector<resample_channel> resamplers; /* one per channel */
int BufSamples;
bool eof;
void Reset(); void Reset();
void ReopenResampler(); void ReopenResampler();
float GetFactor() const;
bool FillBuf(); vector<RageSoundResampler_Polyphase *> resamplers; /* one per channel */
SoundReader *m_pSource;
int m_iSampleRate;
}; };
#endif #endif
/* /*
* Copyright (c) 2003 Glenn Maynard * (c) 2006 Glenn Maynard
* All rights reserved. * All rights reserved.
* *
* Permission is hereby granted, free of charge, to any person obtaining a * Permission is hereby granted, free of charge, to any person obtaining a