C++ Partial::end方法代码示例

// ---------------------------------------------------------------------------
//	Cropper function call operator
// ---------------------------------------------------------------------------
//	Trim a Partial by removing Breakpoints outside a specified time span.
//	Insert a Breakpoint at the boundary when cropping occurs.
//
void 
Cropper::operator()( Partial & p ) const
{
	//	crop beginning of Partial
	Partial::iterator it = p.findAfter( minTime );
	if ( it != p.begin() )    // Partial begins earlier than minTime
	{
	    if ( it != p.end() ) // Partial ends later than minTime
	    {
            Breakpoint bp = p.parametersAt( minTime );
            it = p.insert( minTime, bp );
        }
		it = p.erase( p.begin(), it );
	}
	
	//	crop end of Partial
	it = p.findAfter( maxTime );
	if ( it != p.end() ) // Partial ends later than maxTime
	{
	    if ( it != p.begin() )   // Partial begins earlier than maxTime
	    {
	    	Breakpoint bp = p.parametersAt( maxTime );
    		it = p.insert( maxTime, bp );
    		++it;   //  advance, we don't want to cut this one off
		}
		it = p.erase( it, p.end() );
	}
}

// ---------------------------------------------------------------------------
//	fixPhaseAt
//
//! Recompute phases of all Breakpoints in a Partial
//! so that the synthesize phases match the stored phases, 
//! and the synthesized phase at (nearest) the specified
//! time matches the stored (not recomputed) phase.
//! 
//! Backward phase-fixing stops if a null (zero-amplitude) Breakpoint
//! is encountered, because nulls are interpreted as phase reset points
//! in Loris. If a null is encountered, the remainder of the Partial
//! (the front part) is fixed in the forward direction, beginning at
//! the start of the Partial. Forward phase fixing is only applied 
//! to non-null (nonzero-amplitude) Breakpoints. If a null is encountered, 
//! its phase is simply left unmodified, and future phases wil be 
//! recomputed from that one.
//!
//! \param p    The Partial whose phases should be fixed.
//! \param t    The time at which phases should be made correct.
//
void fixPhaseAt( Partial & p, double t )
{
    if ( 1 < p.numBreakpoints() )
    {
        Partial::iterator pos = p.findNearest( t );
        Assert( pos != p.end() );

        fixPhaseForward( pos, --p.end() );
        fixPhaseBackward( p.begin(), pos );
    }
}

// ---------------------------------------------------------------------------
//	fixPhaseAfter
//
//! Recompute phases of all Breakpoints later than the specified time 
//! so that the synthesize phases of those later Breakpoints matches 
//! the stored phase, as long as the synthesized phase at the specified
//! time matches the stored (not recomputed) phase.
//! 
//! Phase fixing is only applied to non-null (nonzero-amplitude) Breakpoints,
//! because null Breakpoints are interpreted as phase reset points in 
//! Loris. If a null is encountered, its phase is simply left unmodified,
//! and future phases wil be recomputed from that one.
//!
//! \param p    The Partial whose phases should be fixed.
//! \param t    The time after which phases should be adjusted.
//
void fixPhaseAfter( Partial & p, double t )
{
    //  nothing to do it there are not at least
    //  two Breakpoints in the Partial   
    if ( 1 < p.numBreakpoints() )
    {
        Partial::iterator pos = p.findNearest( t );
        Assert( pos != p.end() );

        fixPhaseForward( pos, --p.end() );
    }
}

// ---------------------------------------------------------------------------
//	findContribution		(STATIC)
// ---------------------------------------------------------------------------
//	Find and return an iterator range delimiting the portion of pshort that
// 	should be spliced into the distilled Partial plong. If any Breakpoint 
//	falls in a zero-amplitude region of plong, then pshort should contribute,
//	AND its onset should be retained (!!! This is the weird part!!!). 
//  Therefore, if cbeg is not equal to cend, then cbeg is pshort.begin().
//
std::pair< Partial::iterator, Partial::iterator >
findContribution( Partial & pshort, const Partial & plong, 
				  double fadeTime, double gapTime )
{
	//	a Breakpoint can only fit in the gap if there's
	//	enough time to fade out pshort, introduce a
	//	space of length gapTime, and fade in the rest
	//	of plong (don't need to worry about the fade
	//	in, because we are checking that plong is zero
	//	at cbeg.time() + clearance anyway, so the fade 
	//	in must occur after that, and already be part of 
	//	plong):
    //
    //  WRONG if cbeg is before the start of plong.
    //  Changed so that all Partials are faded in and
    //  out before distilling, so now the clearance 
    //  need only be the gap time:
	double clearance = gapTime; // fadeTime + gapTime;
	
	Partial::iterator cbeg = pshort.begin();
	while ( cbeg != pshort.end() && 
			( plong.amplitudeAt( cbeg.time() ) > 0 ||
			  plong.amplitudeAt( cbeg.time() + clearance ) > 0 ) )
	{
		++cbeg;
	}
	
	Partial::iterator cend = cbeg;
	
	// if a gap is found, find the end of the
	// range of Breakpoints that fit in that
	// gap:
	while ( cend != pshort.end() &&
			plong.amplitudeAt( cend.time() ) == 0 &&
			plong.amplitudeAt( cend.time() + clearance ) == 0 )
	{
		++cend;
	}

	// if a gap is found, and it is big enough for at
	// least one Breakpoint, then include the 
	// onset of the Partial:
	if ( cbeg != pshort.end()  )
	{
		cbeg = pshort.begin();
	}
	
	return std::make_pair( cbeg, cend );
}

// ---------------------------------------------------------------------------
//    harmonify
// ---------------------------------------------------------------------------
//! Apply the reference envelope to a Partial.
//!
//! \pre    The Partial p must be labeled with its harmonic number.
//
void Harmonifier::harmonify( Partial & p ) const
{
    //    compute absolute magnitude thresholds:
    static const double FadeRangeDB = 10;
    const double BeginFade = std::pow( 10., 0.05 * (_freqFixThresholdDb+FadeRangeDB) );
    const double Threshold = std::pow( 10., 0.05 * _freqFixThresholdDb );
    const double OneOverFadeSpan = 1. / ( BeginFade - Threshold );

    double fscale = (double)p.label() / _refPartial.label();
    
    for ( Partial::iterator it = p.begin(); it != p.end(); ++it )
    {
        Breakpoint & bp = it.breakpoint();            
                
        if ( bp.amplitude() < BeginFade )
        {
            //  alpha is the harmonic frequency weighting:
            //  when alpha is 1, the harmonic frequency is used,
            //  when alpha is 0, the breakpoint frequency is
            //  unmodified.
            double alpha = 
                std::min( ( BeginFade - bp.amplitude() ) * OneOverFadeSpan, 1. );
                
            //  alpha is scaled by the weigthing envelope
            alpha *= _weight->valueAt( it.time() );
            
            double fRef = _refPartial.frequencyAt( it.time() );
            
            bp.setFrequency( ( alpha * ( fRef * fscale ) ) + 
                             ( (1 - alpha) * bp.frequency() ) );
        }

    }
}

// ---------------------------------------------------------------------------
//	peakAmplitude
// ---------------------------------------------------------------------------
//! Return the maximum amplitude achieved by a partial. 
//!  
//! \param  p is the Partial to evaluate
//! \return the maximum (absolute) amplitude achieved by 
//!         the partial p
//
double peakAmplitude( const Partial & p )
{
    double peak = 0;
    for ( Partial::const_iterator it = p.begin();
          it != p.end();
          ++it )
    {
        peak = std::max( peak, it->amplitude() );
    }
    return peak;
}

// ---------------------------------------------------------------------------
//	fixFrequency
//
//!	Adjust frequencies of the Breakpoints in the 
//! specified Partial such that the rendered Partial 
//!	achieves (or matches as nearly as possible, within 
//!	the constraint of the maximum allowable frequency
//! alteration) the analyzed phases. 
//!
//! This just iterates over the Partial calling
//! matchPhaseFwd, should probably name those similarly.
//!
//!  \param     partial The Partial whose frequencies,
//!             and possibly phases (if the frequencies
//!             cannot be sufficiently altered to match
//!             the phases), will be recomputed.
//!  \param     maxFixPct The maximum allowable frequency 
//!             alteration, default is 0.2%.
//
void fixFrequency( Partial & partial, double maxFixPct )
{	
	if ( partial.numBreakpoints() > 1 )
	{	
		Partial::iterator next = partial.begin();
		Partial::iterator prev = next++;
		while ( next != partial.end() )		
		{
			matchPhaseFwd( prev.breakpoint(), next.breakpoint(), 
						   next.time() - prev.time(), 0.5, maxFixPct );
			prev = next++;
		}
    }
}

// ---------------------------------------------------------------------------
//	channelize (one Partial)
// ---------------------------------------------------------------------------
//!	Label a Partial with the number of the frequency channel corresponding to
//!	the average frequency over all the Partial's Breakpoints.
//!	
//!	\param partial is the Partial to label.
//
void
Channelizer::channelize( Partial & partial ) const
{
	debugger << "channelizing Partial with " << partial.numBreakpoints() << " Breakpoints" << endl;
			
	//	compute an amplitude-weighted average channel
	//	label for each Partial:
	double ampsum = 0.;
	double weightedlabel = 0.;
	Partial::const_iterator bp;
	for ( bp = partial.begin(); bp != partial.end(); ++bp )
	{
		//	use sinusoidal amplitude:
		double a = bp.breakpoint().amplitude() * std::sqrt( 1. - bp.breakpoint().bandwidth() );
		
		//  This used to be an amplitude-weighted avg, but for many sounds, 
		//  particularly those for which the weighted avg would be very
		//  different from the simple avg, the amplitude-weighted avg
		//  emphasized the part of the sound in which the frequency estimates
		//  are least reliable (e.g. a piano tone). The unweighted 
		//  average should give more intuitive results in most cases.
				
		double f = bp.breakpoint().frequency();
		double t = bp.time();
		
		double refFreq = _refChannelFreq->valueAt( t ) / _refChannelLabel;
		// weightedlabel += a * (f / refFreq);
		weightedlabel += a * giveMeN( f, refFreq, _stretchFactor );
		ampsum += a;
	}
	
	int label;
	if ( ampsum > 0. )	
	// if ( 0 < partial.numBreakpoints() )
	{
		label = (int)((weightedlabel / ampsum) + 0.5);
	}
	else	//	this should never happen, but just in case:
	{
		label = 0;
	}
	Assert( label >= 0 );
			
	//	assign label, and remember it, but
	//	only if it is a valid (positive) 
	//	distillation label:
	partial.setLabel( label );

}

// ---------------------------------------------------------------------------
//	channelize (one Partial)
// ---------------------------------------------------------------------------
//!	Label a Partial with the number of the frequency channel corresponding to
//!	the average frequency over all the Partial's Breakpoints.
//!	
//!	\param partial is the Partial to label.
//
void
Channelizer::channelize( Partial & partial ) const
{
    using std::pow;

	debugger << "channelizing Partial with " << partial.numBreakpoints() << " Breakpoints" << endl;
			
	//	compute an amplitude-weighted average channel
	//	label for each Partial:
	//double ampsum = 0.;
	double weightedlabel = 0.;
	Partial::const_iterator bp;
	for ( bp = partial.begin(); bp != partial.end(); ++bp )
	{
				
		double f = bp.breakpoint().frequency();
		double t = bp.time();
		
        double weight = 1;
        if ( 0 != _ampWeighting )
        {
            //  This used to be an amplitude-weighted avg, but for many sounds, 
            //  particularly those for which the weighted avg would be very
            //  different from the simple avg, the amplitude-weighted avg
            //  emphasized the part of the sound in which the frequency estimates
            //  are least reliable (e.g. a piano tone). The unweighted 
            //  average should give more intuitive results in most cases.

            //	use sinusoidal amplitude:
            double a = bp.breakpoint().amplitude() * std::sqrt( 1. - bp.breakpoint().bandwidth() );                
            weight = pow( a, _ampWeighting );
        }
        
        weightedlabel += weight * computeFractionalChannelNumber( t, f );
	}
	
	int label = 0;
	if ( 0 < partial.numBreakpoints() ) //  should always be the case
	{        
		label = (int)((weightedlabel / partial.numBreakpoints()) + 0.5);
	}
	Assert( label >= 0 );
			
	//	assign label, and remember it, but
	//	only if it is a valid (positive) 
	//	distillation label:
	partial.setLabel( label );

}

// ---------------------------------------------------------------------------
//	avgFrequency
// ---------------------------------------------------------------------------
//! Return the average frequency over all Breakpoints in this Partial.
//! Return zero if the Partial has no Breakpoints.
//!  
//! \param  p is the Partial to evaluate
//! \return the average frequency (Hz) of Breakpoints in the Partial p
//
double avgFrequency( const Partial & p )
{
    double avg = 0;
    for ( Partial::const_iterator it = p.begin();
          it != p.end();
          ++it )
    {
        avg += it->frequency();
    }
    
    if ( avg != 0 )
    {
        avg /= p.numBreakpoints();
    }
    
    return avg;
}

// ---------------------------------------------------------------------------
//	timeOfPeakEnergy (static helper function)
// ---------------------------------------------------------------------------
//	Return the time at which the given Partial attains its
//	maximum sinusoidal energy.
//
static double timeOfPeakEnergy( const Partial & p )
{
	Partial::const_iterator partialIter = p.begin();
	double maxAmp = 
		partialIter->amplitude() * std::sqrt( 1. - partialIter->bandwidth() );
	double time = partialIter.time();
	
	for ( ++partialIter; partialIter != p.end(); ++partialIter ) 
	{
		double a = partialIter->amplitude() * 
					std::sqrt( 1. - partialIter->bandwidth() );
		if ( a > maxAmp ) 
		{
			maxAmp = a;
			time = partialIter.time();
		}
	}			
	
	return time;
}

Iter
find_overlapping( Partial & p, double minGapTime, Iter start, Iter end)
{
	for ( Iter it = start; it != end; ++it ) 
	{
		//	skip if other partial is already sifted out.
		if ( (*it)->label() == 0 )
			continue;
		
		//	skip the source Partial:
		//	(identity test: compare addresses)
		//	(this is a sanity check, should not happen since
		//	src should be at position end)
		Assert( (*it) != &p );

		//  test for overlap:
		if ( p.startTime() < (*it)->endTime() + minGapTime &&
			 p.endTime() + minGapTime > (*it)->startTime() )
		{
			//  Does the overlapping Partial have longer duration?
			//	(this should never be true, since the Partials
			//	are sorted by duration)
			Assert( p.duration() <= (*it)->duration() );
			
#if Debug_Loris
			debugger << "Partial starting " << p.startTime() << ", " 
					 << p.begin().breakpoint().frequency() << " ending " 
					 << p.endTime()  << ", " << (--p.end()).breakpoint().frequency() 
					 << " zapped by overlapping Partial starting " 
					 << (*it)->startTime() << ", " << (*it)->begin().breakpoint().frequency()
					 << " ending " << (*it)->endTime() << ", " 
					 << (--(*it)->end()).breakpoint().frequency()  << endl;
#endif
			return it;
		}
	}
	
	//	no overlapping Partial found:
	return end;
}

// ---------------------------------------------------------------------------
//	weightedAvgFrequency
// ---------------------------------------------------------------------------
//! Return the average frequency over all Breakpoints in this Partial, 
//! weighted by the Breakpoint amplitudes.
//! Return zero if the Partial has no Breakpoints.
//!  
//! \param  p is the Partial to evaluate
//! \return the average frequency (Hz) of Breakpoints in the Partial p
//
double weightedAvgFrequency( const Partial & p )
{
    double avg = 0;
    double ampsum = 0;
    for ( Partial::const_iterator it = p.begin();
          it != p.end();
          ++it )
    {
        avg += it->amplitude() * it->frequency();
        ampsum += it->amplitude();
    }
    
    if ( avg != 0 && ampsum != 0 )
    {
        avg /= ampsum;
    }
    else
    {
        avg = 0;
    }
    
    return avg;
}

// ---------------------------------------------------------------------------
//	fixPhaseForward
//
//! Recompute phases of all Breakpoints later than the specified time 
//! so that the synthesize phases of those later Breakpoints matches 
//! the stored phase, as long as the synthesized phase at the specified
//! time matches the stored (not recomputed) phase. Breakpoints later than
//! tend are unmodified.
//! 
//! Phase fixing is only applied to non-null (nonzero-amplitude) Breakpoints,
//! because null Breakpoints are interpreted as phase reset points in 
//! Loris. If a null is encountered, its phase is simply left unmodified,
//! and future phases wil be recomputed from that one.
//!
//! HEY Is this interesting, in general? Why would you want to do this?
//!
//! \param p    The Partial whose phases should be fixed.
//! \param tbeg The phases and frequencies of Breakpoints later than the 
//!             one nearest this time will be modified.
//! \param tend The phases and frequencies of Breakpoints earlier than the 
//!             one nearest this time will be modified. Should be greater 
//!             than tbeg, or else they will be swapped.
//
void fixPhaseForward( Partial & p, double tbeg, double tend )
{
    if ( tbeg > tend )
    {
        std::swap( tbeg, tend );
    }
    
    //  nothing to do it there are not at least
    //  two Breakpoints in the Partial   
    if ( 1 < p.numBreakpoints() )
    {
        //  find the positions nearest tbeg and tend
        Partial::iterator posbeg = p.findNearest( tbeg );
        Partial::iterator posend = p.findNearest( tend );
        
        //  if the positions are different, and tend is
        //  the end, back it up
        if ( posbeg != posend && posend == p.end() )
        {
            --posend;
        }
        fixPhaseForward( posbeg, posend );
    }
}

// ---------------------------------------------------------------------------
//  synthesize
// ---------------------------------------------------------------------------
//! Synthesize a bandwidth-enhanced sinusoidal Partial. Zero-amplitude
//! Breakpoints are inserted at either end of the Partial to reduce
//! turn-on and turn-off artifacts, as described above. The synthesizer
//! will resize the buffer as necessary to accommodate all the samples,
//! including the fade out. Previous contents of the buffer are not
//! overwritten. Partials with start times earlier than the Partial fade
//! time will have shorter onset fades. Partials are not rendered at
//! frequencies above the half-sample rate. 
//!
//! \param  p The Partial to synthesize.
//! \return Nothing.
//! \pre    The partial must have non-negative start time.
//! \post   This Synthesizer's sample buffer (vector) has been 
//!         resized to accommodate the entire duration of the 
//!         Partial, p, including fade out at the end.
//! \throw  InvalidPartial if the Partial has negative start time.
//  
void
Synthesizer::synthesize( Partial p ) 
{
    if ( p.numBreakpoints() == 0 )
    {
        debugger << "Synthesizer ignoring a partial that contains no Breakpoints" << endl;
        return;
    }
    
    if ( p.startTime() < 0 )
    {
        Throw( InvalidPartial, "Tried to synthesize a Partial having start time less than 0." );
    }

    debugger << "synthesizing Partial from " << p.startTime() * m_srateHz 
             << " to " << p.endTime() * m_srateHz << " starting phase "
             << p.initialPhase() << " starting frequency " 
             << p.first().frequency() << endl;
             
    //  better to compute this only once:
    const double OneOverSrate = 1. / m_srateHz;
    
             
    //  use a Resampler to quantize the Breakpoint times and 
    //  correct the phases:
    Resampler quantizer( OneOverSrate );
    quantizer.setPhaseCorrect( true );
    quantizer.quantize( p );
    

    //  resize the sample buffer if necessary:
    typedef unsigned long index_type;
    index_type endSamp = index_type( ( p.endTime() + m_fadeTimeSec ) * m_srateHz );
    if ( endSamp+1 > m_sampleBuffer->size() )
    {
        //  pad by one sample:
        m_sampleBuffer->resize( endSamp+1 );
    }
    
    //  compute the starting time for synthesis of this Partial,
    //  m_fadeTimeSec before the Partial's startTime, but not before 0:
    double itime = ( m_fadeTimeSec < p.startTime() ) ? ( p.startTime() - m_fadeTimeSec ) : 0.;
    index_type currentSamp = index_type( (itime * m_srateHz) + 0.5 );   //  cheap rounding
    
    //  reset the oscillator:
    //  all that really needs to happen here is setting the frequency
    //  correctly, the phase will be reset again in the loop over 
    //  Breakpoints below, and the amp and bw can start at 0.
    m_osc.resetEnvelopes( BreakpointUtils::makeNullBefore( p.first(), p.startTime() - itime ), m_srateHz );

    //  cache the previous frequency (in Hz) so that it
    //  can be used to reset the phase when necessary
    //  in the sample computation loop below (this saves
    //  having to recompute from the oscillator's radian
    //  frequency):
    double prevFrequency = p.first().frequency();   
    
    //  synthesize linear-frequency segments until 
    //  there aren't any more Breakpoints to make segments:
    double * bufferBegin = &( m_sampleBuffer->front() );
    for ( Partial::const_iterator it = p.begin(); it != p.end(); ++it )
    {
        index_type tgtSamp = index_type( (it.time() * m_srateHz) + 0.5 );   //  cheap rounding
        Assert( tgtSamp >= currentSamp );
        
        //  if the current oscillator amplitude is
        //  zero, and the target Breakpoint amplitude
        //  is not, reset the oscillator phase so that
        //  it matches exactly the target Breakpoint 
        //  phase at tgtSamp:
        if ( m_osc.amplitude() == 0. )
        {
            //  recompute the phase so that it is correct
            //  at the target Breakpoint (need to do this
            //  because the null Breakpoint phase was computed
            //  from an interval in seconds, not samples, so
            //  it might be inaccurate):
            //
            //  double favg = 0.5 * ( prevFrequency + it.breakpoint().frequency() );
            //  double dphase = 2 * Pi * favg * ( tgtSamp - currentSamp ) / m_srateHz;
//.........这里部分代码省略.........