예제 #1
0
def stochasticModelSynth(stocEnv, H, N):
	"""
	Stochastic synthesis of a sound
	stocEnv: stochastic envelope; H: hop size; N: fft size
	returns y: output sound
	"""

	if not(UF.isPower2(N)):                                 	# raise error if N not a power of two
		raise ValueError("N is not a power of two")
 
	hN = N/2+1                                            		# positive size of fft
	No2 = N/2							# half of N
	L = stocEnv[:,0].size                                    	# number of frames
	ysize = H*(L+3)                                         	# output sound size
	y = np.zeros(ysize)                                     	# initialize output array
	ws = 2*hanning(N)                                        	# synthesis window
	pout = 0                                                 	# output sound pointer
	for l in range(L):                    
		mY = resample(stocEnv[l,:], hN)                        # interpolate to original size
		pY = 2*np.pi*np.random.rand(hN)                        # generate phase random values
		Y = np.zeros(N, dtype = complex)                       # initialize synthesis spectrum
		Y[:hN] = 10**(mY/20) * np.exp(1j*pY)                   # generate positive freq.
		Y[hN:] = 10**(mY[-2:0:-1]/20) * np.exp(-1j*pY[-2:0:-1]) # generate negative freq.
		fftbuffer = np.real(ifft(Y))                           # inverse FFT
		y[pout:pout+N] += ws*fftbuffer                         # overlap-add
		pout += H  
	y = np.delete(y, range(No2))                              # delete half of first window
	y = np.delete(y, range(y.size-No2, y.size))               # delete half of the last window 
	return y
예제 #2
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def dftAnal(x, w, N):
	"""
	Analysis of a signal using the discrete Fourier transform
	x: input signal, w: analysis window, N: FFT size 
	returns mX, pX: magnitude and phase spectrum
	"""

	if not(UF.isPower2(N)):                                 # raise error if N not a power of two
		raise ValueError("FFT size (N) is not a power of 2")

	if (w.size > N):                                        # raise error if window size bigger than fft size
		raise ValueError("Window size (M) is bigger than FFT size")

	hN = int((N/2)+1)                                            # size of positive spectrum, it includes sample 0
	hM1 = int(math.floor((w.size+1)/2))                     # half analysis window size by rounding
	hM2 = int(math.floor(w.size/2))                         # half analysis window size by floor
	fftbuffer = np.zeros(N)                                 # initialize buffer for FFT
	w = w / sum(w)                                          # normalize analysis window
	xw = x*w                                                # window the input sound
	fftbuffer[:hM1] = xw[hM2:]                              # zero-phase window in fftbuffer
	fftbuffer[-hM2:] = xw[:hM2]        
	X = fft(fftbuffer)                                      # compute FFT
	absX = abs(X[:hN])                                      # compute absolute value of positive side
	absX[absX<np.finfo(float).eps] = np.finfo(float).eps    # if zeros add epsilon to handle log
	mX = 20 * np.log10(absX)                                # magnitude spectrum of positive frequencies in dB
	X[:hN].real[np.abs(X[:hN].real) < tol] = 0.0            # for phase calculation set to 0 the small values
	X[:hN].imag[np.abs(X[:hN].imag) < tol] = 0.0            # for phase calculation set to 0 the small values         
	pX = np.unwrap(np.angle(X[:hN]))                        # unwrapped phase spectrum of positive frequencies
	return mX, pX
예제 #3
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def dft_anal(x, w, N):
    """
    Analysis of a signal using the discrete Fourier transform
    x: input signal, w: analysis window, N: FFT size
    returns mX, pX: magnitude and phase spectrum
    """

    if not utilFunctions.isPower2(N):  # raise error if N not a power of two
        raise ValueError("FFT size (N) is not a power of 2")

    if w.size > N:  # raise error if window size bigger than fft size
        raise ValueError("Window size (M) is bigger than FFT size")

    hN = (N / 2) + 1  # size of positive spectrum, it includes sample 0
    hM1 = int(math.floor((w.size + 1) / 2))  # half analysis window size by rounding
    hM2 = int(math.floor(w.size / 2))  # half analysis window size by floor
    fftbuffer = np.zeros(N)  # initialize buffer for FFT
    w = w / sum(w)  # normalize analysis window
    xw = x * w  # window the input sound
    fftbuffer[:hM1] = xw[hM2:]  # zero-phase window in fftbuffer
    fftbuffer[-hM2:] = xw[:hM2]
    X = fft(fftbuffer)  # compute FFT
    absX = abs(X[:hN])  # compute ansolute value of positive side
    absX[absX < np.finfo(float).eps] = np.finfo(float).eps  # if zeros add epsilon to handle log
    mX = 20 * np.log10(absX)  # magnitude spectrum of positive frequencies in dB
    X[:hN].real[np.abs(X[:hN].real) < tol] = 0.0  # for phase calculation set to 0 the small values
    X[:hN].imag[np.abs(X[:hN].imag) < tol] = 0.0  # for phase calculation set to 0 the small values
    pX = np.unwrap(np.angle(X[:hN]))  # unwrapped phase spectrum of positive frequencies
    return mX, pX
def stochasticModelSynth(stocEnv, H, N):
    """
	Stochastic synthesis of a sound
	stocEnv: stochastic envelope; H: hop size; N: fft size
	returns y: output sound
	"""

    if not (UF.isPower2(N)):  # raise error if N not a power of two
        raise ValueError("N is not a power of two")

    hN = N // 2 + 1  # positive size of fft
    No2 = N // 2  # half of N
    L = stocEnv[:, 0].size  # number of frames
    ysize = H * (L + 3)  # output sound size
    y = np.zeros(ysize)  # initialize output array
    ws = 2 * hanning(N)  # synthesis window
    pout = 0  # output sound pointer
    for l in range(L):
        mY = resample(stocEnv[l, :], hN)  # interpolate to original size
        pY = 2 * np.pi * np.random.rand(hN)  # generate phase random values
        Y = np.zeros(N, dtype=complex)  # initialize synthesis spectrum
        Y[:hN] = 10**(mY / 20) * np.exp(1j * pY)  # generate positive freq.
        Y[hN:] = 10**(mY[-2:0:-1] / 20) * np.exp(
            -1j * pY[-2:0:-1])  # generate negative freq.
        fftbuffer = np.real(ifft(Y))  # inverse FFT
        y[pout:pout + N] += ws * fftbuffer  # overlap-add
        pout += H
    y = np.delete(y, range(No2))  # delete half of first window
    y = np.delete(y, range(y.size - No2,
                           y.size))  # delete half of the last window
    return y
예제 #5
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def dftSynth(mX, pX, M):
    """
	Synthesis of a signal using the discrete Fourier transform
	mX: magnitude spectrum, pX: phase spectrum, M: window size
	returns y: output signal
	"""

    hN = mX.size  # size of positive spectrum, it includes sample 0
    N = (hN - 1) * 2  # FFT size
    if not (UF.isPower2(N)
            ):  # raise error if N not a power of two, thus mX is wrong
        raise ValueError("size of mX is not (N/2)+1")

    hM1 = int(math.floor((M + 1) / 2))  # half analysis window size by rounding
    hM2 = int(math.floor(M / 2))  # half analysis window size by floor
    fftbuffer = np.zeros(N)  # initialize buffer for FFT
    y = np.zeros(M)  # initialize output array
    Y = np.zeros(N, dtype=complex)  # clean output spectrum
    Y[:hN] = 10**(mX / 20) * np.exp(1j * pX)  # generate positive frequencies
    Y[hN:] = 10**(mX[-2:0:-1] / 20) * np.exp(
        -1j * pX[-2:0:-1])  # generate negative frequencies
    fftbuffer = np.real(ifft(Y))  # compute inverse FFT
    y[:hM2] = fftbuffer[-hM2:]  # undo zero-phase window
    y[hM2:] = fftbuffer[:hM1]
    return y
def stochasticModel(x, H, N, stocf):
    """
	Stochastic analysis/synthesis of a sound, one frame at a time
	x: input array sound, H: hop size, N: fft size 
	stocf: decimation factor of mag spectrum for stochastic analysis, bigger than 0, maximum of 1
	returns y: output sound
	"""
    hN = N // 2 + 1  # positive size of fft
    No2 = N // 2  # half of N
    if (hN * stocf < 3):  # raise exception if decimation factor too small
        raise ValueError("Stochastic decimation factor too small")

    if (stocf > 1):  # raise exception if decimation factor too big
        raise ValueError("Stochastic decimation factor above 1")

    if (H <= 0):  # raise error if hop size 0 or negative
        raise ValueError("Hop size (H) smaller or equal to 0")

    if not (UF.isPower2(N)):  # raise error if N not a power of twou
        raise ValueError("FFT size (N) is not a power of 2")

    w = hanning(N)  # analysis/synthesis window
    x = np.append(
        np.zeros(No2),
        x)  # add zeros at beginning to center first window at sample 0
    x = np.append(x,
                  np.zeros(No2))  # add zeros at the end to analyze last sample
    pin = No2  # initialize sound pointer in middle of analysis window
    pend = x.size - No2  # last sample to start a frame
    y = np.zeros(x.size)  # initialize output array
    while pin <= pend:
        #-----analysis-----
        xw = x[pin - No2:pin + No2] * w  # window the input sound
        X = fft(xw)  # compute FFT
        mX = 20 * np.log10(abs(
            X[:hN]))  # magnitude spectrum of positive frequencies
        stocEnv = resample(np.maximum(-200, mX),
                           int(hN * stocf))  # decimate the mag spectrum
        #-----synthesis-----
        mY = resample(stocEnv, hN)  # interpolate to original size
        pY = 2 * np.pi * np.random.rand(hN)  # generate phase random values
        Y = np.zeros(N, dtype=complex)
        Y[:hN] = 10**(mY / 20) * np.exp(1j * pY)  # generate positive freq.
        Y[hN:] = 10**(mY[-2:0:-1] / 20) * np.exp(
            -1j * pY[-2:0:-1])  # generate negative freq.
        fftbuffer = np.real(ifft(Y))  # inverse FFT
        y[pin - No2:pin + No2] += w * fftbuffer  # overlap-add
        pin += H  # advance sound pointer
    y = np.delete(y, range(No2))  # delete half of first window which was added
    y = np.delete(y,
                  range(y.size - No2,
                        y.size))  # delete half of last window which was added
    return y
예제 #7
0
def stochasticModel(x, H, N, stocf):
	"""
	Stochastic analysis/synthesis of a sound, one frame at a time
	x: input array sound, H: hop size, N: fft size 
	stocf: decimation factor of mag spectrum for stochastic analysis, bigger than 0, maximum of 1
	returns y: output sound
	"""
	hN = N/2+1                                            		# positive size of fft
	No2 = N/2							# half of N
	if (hN*stocf < 3):                                              # raise exception if decimation factor too small
		raise ValueError("Stochastic decimation factor too small")
		
	if (stocf > 1):                                          # raise exception if decimation factor too big
		raise ValueError("Stochastic decimation factor above 1")
	
	if (H <= 0):                                             # raise error if hop size 0 or negative
		raise ValueError("Hop size (H) smaller or equal to 0")
		
	if not(UF.isPower2(N)):                                  # raise error if N not a power of twou
		raise ValueError("FFT size (N) is not a power of 2")
		
	w = hanning(N)                                           # analysis/synthesis window
	x = np.append(np.zeros(No2),x)                           # add zeros at beginning to center first window at sample 0
	x = np.append(x,np.zeros(No2))                           # add zeros at the end to analyze last sample
	pin = No2                                                # initialize sound pointer in middle of analysis window       
	pend = x.size - No2                                      # last sample to start a frame
	y = np.zeros(x.size)                                     # initialize output array
	while pin<=pend:              
	#-----analysis-----             
		xw = x[pin-No2:pin+No2]*w                              # window the input sound
		X = fft(xw)                                            # compute FFT
		mX = 20 * np.log10(abs(X[:hN]))                        # magnitude spectrum of positive frequencies
		stocEnv = resample(np.maximum(-200, mX), hN*stocf)     # decimate the mag spectrum     
	#-----synthesis-----
		mY = resample(stocEnv, hN)                             # interpolate to original size
		pY = 2*np.pi*np.random.rand(hN)                        # generate phase random values
		Y = np.zeros(N, dtype = complex)
		Y[:hN] = 10**(mY/20) * np.exp(1j*pY)                   # generate positive freq.
		Y[hN:] = 10**(mY[-2:0:-1]/20) * np.exp(-1j*pY[-2:0:-1]) # generate negative freq.
		fftbuffer = np.real(ifft(Y))                           # inverse FFT
		y[pin-No2:pin+No2] += w*fftbuffer                      # overlap-add
		pin += H  					 # advance sound pointer
	y = np.delete(y, range(No2))                              # delete half of first window which was added 
	y = np.delete(y, range(y.size-No2, y.size))               # delete half of last window which was added                                            
	return y
def stochasticModelAnal(x, H, N, stocf):
    """
	Stochastic analysis of a sound
	x: input array sound, H: hop size, N: fftsize
	stocf: decimation factor of mag spectrum for stochastic analysis, bigger than 0, maximum of 1
	returns stocEnv: stochastic envelope
	"""

    hN = N // 2 + 1  # positive size of fft
    No2 = N // 2  # half of N
    if (hN * stocf < 3):  # raise exception if decimation factor too small
        raise ValueError("Stochastic decimation factor too small")

    if (stocf > 1):  # raise exception if decimation factor too big
        raise ValueError("Stochastic decimation factor above 1")

    if (H <= 0):  # raise error if hop size 0 or negative
        raise ValueError("Hop size (H) smaller or equal to 0")

    if not (UF.isPower2(N)):  # raise error if N not a power of two
        raise ValueError("FFT size (N) is not a power of 2")

    w = hanning(N)  # analysis window
    x = np.append(
        np.zeros(No2),
        x)  # add zeros at beginning to center first window at sample 0
    x = np.append(x,
                  np.zeros(No2))  # add zeros at the end to analyze last sample
    pin = No2  # initialize sound pointer in middle of analysis window
    pend = x.size - No2  # last sample to start a frame
    while pin <= pend:
        xw = x[pin - No2:pin + No2] * w  # window the input sound
        X = fft(xw)  # compute FFT
        mX = 20 * np.log10(abs(
            X[:hN]))  # magnitude spectrum of positive frequencies
        mY = resample(np.maximum(-200, mX),
                      int(stocf * hN))  # decimate the mag spectrum
        if pin == No2:  # first frame
            stocEnv = np.array([mY])
        else:  # rest of frames
            stocEnv = np.vstack((stocEnv, np.array([mY])))
        pin += H  # advance sound pointer
    return stocEnv
예제 #9
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def dftModel(x, w, N):
    """
	Analysis/synthesis of a signal using the discrete Fourier transform
	x: input signal, w: analysis window, N: FFT size
	returns y: output signal
	"""

    if not (UF.isPower2(N)):  # raise error if N not a power of twou
        raise ValueError("FFT size (N) is not a power of 2")

    if (w.size > N):  # raise error if window size bigger than fft size
        raise ValueError("Window size (M) is bigger than FFT size")

    if all(x == 0):  # if input array is zeros return empty output
        return np.zeros(x.size)
    hN = (N / 2) + 1  # size of positive spectrum, it includes sample 0
    hM1 = int(math.floor(
        (w.size + 1) / 2))  # half analysis window size by rounding
    hM2 = int(math.floor(w.size / 2))  # half analysis window size by floor
    fftbuffer = np.zeros(N)  # initialize buffer for FFT
    y = np.zeros(x.size)  # initialize output array
    #----analysis--------
    xw = x * w  # window the input sound
    fftbuffer[:hM1] = xw[hM2:]  # zero-phase window in fftbuffer
    fftbuffer[-hM2:] = xw[:hM2]
    X = fft(fftbuffer)  # compute FFT
    absX = abs(X[:hN])  # compute ansolute value of positive side
    absX[absX < np.finfo(float).eps] = np.finfo(
        float).eps  # if zeros add epsilon to handle log
    mX = 20 * np.log10(
        absX)  # magnitude spectrum of positive frequencies in dB
    pX = np.unwrap(np.angle(
        X[:hN]))  # unwrapped phase spectrum of positive frequencies
    #-----synthesis-----
    Y = np.zeros(N, dtype=complex)  # clean output spectrum
    Y[:hN] = 10**(mX / 20) * np.exp(1j * pX)  # generate positive frequencies
    Y[hN:] = 10**(mX[-2:0:-1] / 20) * np.exp(
        -1j * pX[-2:0:-1])  # generate negative frequencies
    fftbuffer = np.real(ifft(Y))  # compute inverse FFT
    y[:hM2] = fftbuffer[-hM2:]  # undo zero-phase window
    y[hM2:] = fftbuffer[:hM1]
    return y
예제 #10
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def stochastic_model_anal(x, H, N, stocf):
    """
    Stochastic analysis of a sound
    x: input array sound, H: hop size, N: fftsize
    stocf: decimation factor of mag spectrum for stochastic analysis, bigger than 0, maximum of 1
    returns stocEnv: stochastic envelope
    """

    hN = N / 2 + 1  # positive size of fft
    No2 = N / 2  # half of N
    if hN * stocf < 3:  # raise exception if decimation factor too small
        raise ValueError("Stochastic decimation factor too small")

    if stocf > 1:  # raise exception if decimation factor too big
        raise ValueError("Stochastic decimation factor above 1")

    if H <= 0:  # raise error if hop size 0 or negative
        raise ValueError("Hop size (H) smaller or equal to 0")

    if not (UF.isPower2(N)):  # raise error if N not a power of two
        raise ValueError("FFT size (N) is not a power of 2")

    w = hanning(N)  # analysis window
    x = np.append(np.zeros(No2), x)  # add zeros at beginning to center first window at sample 0
    x = np.append(x, np.zeros(No2))  # add zeros at the end to analyze last sample
    pin = No2  # initialize sound pointer in middle of analysis window
    pend = x.size - No2  # last sample to start a frame

    stocEnv = None
    while pin <= pend:
        xw = x[pin - No2:pin + No2] * w  # window the input sound
        X = fft(xw)  # compute FFT
        mX = 20 * np.log10(abs(X[:hN]))  # magnitude spectrum of positive frequencies
        mY = resample(np.maximum(-200, mX), stocf * hN)  # decimate the mag spectrum
        if pin == No2:  # first frame
            stocEnv = np.array([mY])
        else:  # rest of frames
            stocEnv = np.vstack((stocEnv, np.array([mY])))
        pin += H  # advance sound pointer
    return stocEnv
예제 #11
0
def dft_synth(mX, pX, M):
    """
    Synthesis of a signal using the discrete Fourier transform
    mX: magnitude spectrum, pX: phase spectrum, M: window size
    returns y: output signal
    """

    hN = mX.size  # size of positive spectrum, it includes sample 0
    N = (hN - 1) * 2  # FFT size
    if not utilFunctions.isPower2(N):  # raise error if N not a power of two, thus mX is wrong
        raise ValueError("size of mX is not (N/2)+1")

    hM1 = int(math.floor((M + 1) / 2))  # half analysis window size by rounding
    hM2 = int(math.floor(M / 2))  # half analysis window size by floor
    y = np.zeros(M)  # initialize output array
    Y = np.zeros(N, dtype=complex)  # clean output spectrum
    Y[:hN] = 10 ** (mX / 20) * np.exp(1j * pX)  # generate positive frequencies
    Y[hN:] = 10 ** (mX[-2:0:-1] / 20) * np.exp(-1j * pX[-2:0:-1])  # generate negative frequencies
    fftbuffer = np.real(ifft(Y))  # compute inverse FFT
    y[:hM2] = fftbuffer[-hM2:]  # undo zero-phase window
    y[hM2:] = fftbuffer[:hM1]
    return y
예제 #12
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def dftModel(x, w, N):
	"""
	Analysis/synthesis of a signal using the discrete Fourier transform
	x: input signal, w: analysis window, N: FFT size
	returns y: output signal
	"""

	if not(UF.isPower2(N)):                                 # raise error if N not a power of twou
		raise ValueError("FFT size (N) is not a power of 2")

	if (w.size > N):                                        # raise error if window size bigger than fft size
		raise ValueError("Window size (M) is bigger than FFT size")

	if all(x==0):                                           # if input array is zeros return empty output
		return np.zeros(x.size)
	hN = (N/2)+1                                            # size of positive spectrum, it includes sample 0
	hM1 = int(math.floor((w.size+1)/2))                     # half analysis window size by rounding
	hM2 = int(math.floor(w.size/2))                         # half analysis window size by floor
	fftbuffer = np.zeros(N)                                 # initialize buffer for FFT
	y = np.zeros(x.size)                                    # initialize output array
	#----analysis--------
	xw = x*w                                                # window the input sound
	fftbuffer[:hM1] = xw[hM2:]                              # zero-phase window in fftbuffer
	fftbuffer[-hM2:] = xw[:hM2]        
	X = fft(fftbuffer)                                      # compute FFT
	absX = abs(X[:hN])                                      # compute ansolute value of positive side
	absX[absX<np.finfo(float).eps] = np.finfo(float).eps    # if zeros add epsilon to handle log
	mX = 20 * np.log10(absX)                                # magnitude spectrum of positive frequencies in dB     
	pX = np.unwrap(np.angle(X[:hN]))                        # unwrapped phase spectrum of positive frequencies
	#-----synthesis-----
	Y = np.zeros(N, dtype = complex)                        # clean output spectrum
	Y[:hN] = 10**(mX/20) * np.exp(1j*pX)                    # generate positive frequencies
	Y[hN:] = 10**(mX[-2:0:-1]/20) * np.exp(-1j*pX[-2:0:-1]) # generate negative frequencies
	fftbuffer = np.real(ifft(Y))                            # compute inverse FFT
	y[:hM2] = fftbuffer[-hM2:]                              # undo zero-phase window
	y[hM2:] = fftbuffer[:hM1]
	return y
예제 #13
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def sineModelMultiRes(x, f, windows, Nsizes, fBands, t):
	'''
		For each audio frame
		1. compute three different DFTs with three different window sizes (which are input parameters)
		2. compute the sinusoid peaks for each of the DFTs.
		3. choose the peaks from these three DFTs depending on the band to which the frequency of the peak belongs
		x: input array sound, windows: analysis windows [w1, w2, w3]
		Nsizes: FFT sizes[N1, N2, N3], t: threshold in negative dB
		fBands: frequency band edges [B1, B2, B3]
	'''


	# raise error if N not a power of two
	if not all(UF.isPower2(item) for item in Ns):                                 
		raise ValueError("All FFT size (N) must be a power of 2")
	
	# raise error if window size bigger than fft size
	if (w.size > N):                                        
		raise ValueError("Window size (M) is bigger than FFT size")
	
	# Check if all sizes in Window are ints
	if not  all(isinstance(item, int) for item in windows):
		raise ValueError("All window sizes must be ints")

	# Check if all FFT sizes are ints
	if not  all(isinstance(item, int) for item in Ns):
		raise ValueError("All FFT sizes must be ints")
	
	# Check if all freq bands are ints
	if not  all(isinstance(item, int) for item in Ns):
		raise ValueError("All freq Bands must be ints")

	# global synthsis params
	Ns = 512                                                # FFT size for synthesis (even)
	H = Ns//4                                               # Hop size used for analysis and synthesis
	hNs = Ns//2                                             # half of synthesis FFT size
	yw = np.zeros(Ns)                                       # initialize output sound frame
	y = np.zeros(x.size)                                    # initialize output array
	sw = np.zeros(Ns)                                       # initialize synthesis window
	ow = triang(2*H)                                        # triangular window
	sw[hNs-H:hNs+H] = ow                                    # add triangular window
	bh = blackmanharris(Ns)                                 # blackmanharris window
	bh = bh / sum(bh)                                       # normalized blackmanharris window
	sw[hNs-H:hNs+H] = sw[hNs-H:hNs+H] / bh[hNs-H:hNs+H]     # normalized synthesis window

	for i in range(3):
		w = windows[i]
		N = Nsizes[i]
		Bmin, Bmax = fBands[i]
		hM1 = int(math.floor((w.size+1)/2))                     # half analysis window size by rounding
		hM2 = int(math.floor(w.size/2))
		pin = max(hNs, hM1)                                     # init sound pointer in middle of anal window       
		pend = x.size - max(hNs, hM1)                           # last sample to start a frame
		w = w / sum(w)											# normalize analysis window
		while pin<pend:                                         # while input sound pointer is within sound 
			#-----analysis-----             
			x1 = x[pin-hM1:pin+hM2]                               # select frame
			mX, pX = DFT.dftAnal(x1, w, N)                        # compute dft
			ploc = UF.peakDetection(mX, t)                        # detect locations of peaks
			iploc, ipmag, ipphase = UF.peakInterp(mX, pX, ploc)   # refine peak values by interpolation
			ipfreq = fs*iploc/float(N)                            # convert peak locations to Hertz
			
			# Filter indexes within the frequency bands Bmin and Bmax
			ipmag = ipmag[np.logical_and(ipfreq >= Bmin, ipfreq < Bmax)]
            ipphase = ipphase[np.logical_and(ipfreq >= Bmin, ipfreq < Bmax)]
            ipfreq = ipfreq[np.logical_and(ipfreq >= Bmin, ipfreq < Bmax)]
			#-----synthesis-----
			Y = UF.genSpecSines(ipfreq, ipmag, ipphase, Ns, fs)   # generate sines in the spectrum         
			fftbuffer = np.real(ifft(Y))                          # compute inverse FFT
			yw[:hNs-1] = fftbuffer[hNs+1:]                        # undo zero-phase window
			yw[hNs-1:] = fftbuffer[:hNs+1] 
			y[pin-hNs:pin+hNs] += sw*yw                           # overlap-add and apply a synthesis window
			pin += H