コード例 #1
0
ファイル: stft.py プロジェクト: sharkinyourcomputer/agui
def Resample(S, ns, nr, p, rate):
  w_s = rfftfreq(ns, 1.0/rate)
  d_s  = 1.*rate / ns
  w_r = rfftfreq(nr, 1.0/rate)
  d_r = 1.*rate / nr
  i_wfloor = (np.floor(w_r[1:]/p / d_s) - 1).astype(np.int)
  i_wceil = (np.ceil(w_r[1:]/p / d_s) - 1).astype(np.int)
  
  w_floor = w_s[1+i_wfloor]
  w_ceil = w_s[1+i_wceil]
  a_floor0, a_ceil0 = (w_r[1:]/p - w_floor), (w_ceil - w_r[1:]/p)
  # Some freqs may be exact matches, e.g. if nr == 2*ns, every other one.
  exact = (a_floor0 == 0.0) & (a_ceil0 == 0.0)
  # Ignore the DC Component and any exact matches in the interpolation.
  R = np.hstack((np.array([S[0]]), np.zeros(w_r.size-1)))
  R[1:][exact] = S[1+i_wfloor[exact]]
  # The rest: interpolate mag/phase from the nearest sampled frequencies.
  absS, angS = np.abs(S[1:]), np.angle(S[1:])
  intrp = ~exact
  # Basic floor/ceil linear interpolation ...
  i_fl, i_cl = i_wfloor[intrp], i_wceil[intrp]
  afl0, acl0 = a_floor0[intrp], a_ceil0[intrp]
  afl = 1.0 - afl0 / (afl0 + acl0)
  acl = 1.0 - acl0 / (afl0 + acl0)
  absR = afl * absS[i_fl] + acl * absS[i_cl]
  angR = afl * angS[i_fl] + acl * angS[i_cl]
  R[1:][intrp] = absR * np.exp(1j * angR)
  assert R.size == w_r.size, (R.size, w_r.size, S.size, ns, nr)
  return R
コード例 #2
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 def test_definition(self):
     x = [0, 1, 2, 3, 4]
     assert_array_almost_equal(9*fft.rfftfreq(9), x)
     assert_array_almost_equal(9*pi*fft.rfftfreq(9, pi), x)
     x = [0, 1, 2, 3, 4, 5]
     assert_array_almost_equal(10*fft.rfftfreq(10), x)
     assert_array_almost_equal(10*pi*fft.rfftfreq(10, pi), x)
コード例 #3
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ファイル: test_helper.py プロジェクト: DavTho1983/UPress2
 def test_definition(self):
     x = [0, 1, 2, 3, 4]
     assert_array_almost_equal(9 * fft.rfftfreq(9), x)
     assert_array_almost_equal(9 * pi * fft.rfftfreq(9, pi), x)
     x = [0, 1, 2, 3, 4, 5]
     assert_array_almost_equal(10 * fft.rfftfreq(10), x)
     assert_array_almost_equal(10 * pi * fft.rfftfreq(10, pi), x)
コード例 #4
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ファイル: mode_util_base.py プロジェクト: wzmsykk/cst_tools
def mode_wavenumber_mnp(XC,YFC,XR,YFR,XZ,YFZ):
    '''
    XR length along the radius
    YFR tangental component of the field along the radius curve
    XC length along the circle
    YFC tangental component of the field along the circle curve
    XZ length along the line parallel to the z-axis
    YFZ tangental component of the field along the line curve
    '''
    #FIND M
    rpc=1
    my=YFC
    nmy=np.reshape(np.repeat(np.reshape(my,(1,len(my))),rpc,axis=0),rpc*len(my))
    #print("MY_LEN=%d,NMY_LEN=%d"%(len(my),len(nmy)))
    N=len(nmy)
    fft_y=fft.rfft(my,n=my.size)
    abs_y=np.abs(fft_y)
    freqs = fft.rfftfreq(my.size, d=1./len(my))
    M_NUM = freqs[np.argmax(abs_y)]


    #寻找N_零点个数
    nzeros=0
    ntps=len(XR) #total points
    for i in range (ntps-1):
        if YFR[i]*YFR[i+1]<0:
            nzeros+=1
        elif YFR[i]==0:
            nzeros+=1
    
    #M=0 曲线无0点 N_NUM=1
    if M_NUM==0 and nzeros==0:
        N_NUM=1
    else:
        N_NUM=nzeros


    #寻找P_零点个数? 周期个数?
    m2=np.abs(np.mean(YFZ))**2
    if YFZ[0]*YFZ[-1]>0 and YFZ[0]*YFZ[-1]/m2>0.9:
        P_NUM=0
    else:
        rpc=1
        
        my=np.concatenate([-YFZ,YFZ])
        
        

        nmy=np.reshape(np.repeat(np.reshape(my,(1,len(my))),rpc,axis=0),rpc*len(my))
        #print("MY_LEN=%d,NMY_LEN=%d"%(len(my),len(nmy)))
        fft_y=fft.rfft(my,n=my.size)
        abs_y=np.abs(fft_y)
        freqs = fft.rfftfreq(my.size, d=1./len(my))
        P_NUM = freqs[np.argmax(abs_y)]

    return M_NUM,N_NUM,P_NUM
コード例 #5
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 def complex_local_wavenumbers(self):
     """Returns local wavenumbers of complex space"""
     return (fftfreq(self.N[0], 1. / self.N[0]).astype(self.float),
             fftfreq(self.N[1],
                     1. / self.N[1])[self.complex_local_slice()[1]].astype(
                         self.float),
             rfftfreq(self.N[2], 1. / self.N[2]).astype(self.float))
コード例 #6
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ファイル: representations.py プロジェクト: choyun1/sigtools
    def __init__(self, sound, win_dur, win_overlap=0.5, win_func=hanning):
        sig_len = len(sound)
        data = sound.data
        fs = sound.fs
        win_len = floor(fs * win_dur)
        adv_len = floor(win_len * (1 - win_overlap))
        n_wins = floor((len(sound) - win_len) / adv_len)
        window = win_func(win_len)

        spec_len = win_len // 2 + 1
        stft = np.zeros((spec_len, n_wins), dtype=np.complex)
        for win_idx in range(n_wins):
            beg_idx = win_idx * adv_len
            end_idx = beg_idx + win_len

            curr_data = window * data[beg_idx:end_idx]
            stft[:, win_idx] = rfft(curr_data)

        self.t = np.linspace(0, sig_len / fs, n_wins)
        self.f = fs * rfftfreq(win_len)
        self.stft = stft
        self.mag_stft = np.abs(self.stft)
        log_mag_stft = 10 * np.log10(self.mag_stft)
        log_mag_stft[log_mag_stft == -np.inf] = np.min(
            log_mag_stft[log_mag_stft != -np.inf])
        self.log_mag_stft = log_mag_stft
        self.fs = fs
        self.sig_len = sig_len
        self.win_len = win_len
        self.adv_len = adv_len
        self.n_wins = n_wins
        self.window = window
コード例 #7
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ファイル: fft.py プロジェクト: subhylahiri/sl_py_tools
def rfftfreqs(shape: Sequence[int],
              spacings: Optional[Sequence[float]] = None) -> Tuple[np.ndarray]:
    """
    The Discrete Real Fourier Transform sample frequencies for each axis.

    Parameters
    ----------
    shape : Sequence[int]
        Axes length.
    spacings : Sequence[scalar], optional
        Sample spacings (inverse of the sampling rates). Defaults to 1.

    Returns
    -------
    freqs : Tuple[np.ndarray]
        Arrays of frequencies.

    See Also
    --------
    numpy.fft.fftfreq : for all axes except last.
    numpy.fft.rfftfreq : for last axis.
    """
    if spacings is None:
        spacings = (1., ) * len(shape)
    freqs = fftfreqs(shape[:-1], spacings[:-1])
    freqs += (fft.rfftfreq(shape[-1], spacings[-1]), )
    return freqs
コード例 #8
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    def __init__(self, lenChunk, rate, data):
        print('Building FFT Stereo chunks...')
        self._lenChunk = lenChunk
        self._rate = rate
        #Duration of a sample = lenSample (in seconds)
        lenSample = 1 / rate
        leftData, rightData = data[:, 0], data[:, 1]
        #number of chunks:
        nbChunks = int(len(leftData) / (lenChunk * rate))

        lChunksNotAveraged = array_split(leftData, nbChunks)
        rChunksNotAveraged = array_split(rightData, nbChunks)

        #
        #Computes FFT
        lChunksNotAveraged = array(list(map(fft.rfft, lChunksNotAveraged)))
        rChunksNotAveraged = array(list(map(fft.rfft, rChunksNotAveraged)))

        self._nbSamples = [
            len(lChunksNotAveraged[i]) for i in range(len(lChunksNotAveraged))
        ]
        self._freqs = [
            fft.rfftfreq(2 * self._nbSamples[i] - 1, lenSample)
            for i in range(len(lChunksNotAveraged))
        ]

        self._lChunks = lChunksNotAveraged
        self._rChunks = rChunksNotAveraged
        ####
        print('...Done')
コード例 #9
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ファイル: _noise.py プロジェクト: ucbpylegroup/QETpy
def foldpsd(psd, fs):
    """
    Return the one-sided version of the inputted two-sided psd.
    
    Parameters
    ----------
    psd : ndarray
        A two-sided psd to be converted to one-sided
    fs : float
        The sample rate used for the psd
            
    Returns
    -------
    f : ndarray
        The frequencies corresponding to the outputted one-sided psd
    psd_folded : ndarray
        The one-sided (folded over) psd corresponding to the inputted two-sided psd
            
    """

    psd_folded = np.copy(psd[:len(psd) // 2 + 1])
    psd_folded[1:len(psd) // 2 + (len(psd)) % 2] *= 2.0
    f = rfftfreq(len(psd), d=1.0 / fs)

    return f, psd_folded
コード例 #10
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    def create_output_signal(self, freq=1000, level=-3, sig_type='sine'):
        '''Creates the array for global out_sig.
        Should eventually handle multiple signal types.
        Currently this only builds sine waves that are periodic
        in our buffer size. It finds the closest frequency to the
        specified frequency.

        takes inputs:
            freq: frequency in Hz.
            level: signal level in dB.
            sig_type: Currently only supports sine waves.
        '''

        if self.out_enable:
            retoggle = True
            self.toggle_out()
        else:
            retoggle = False

        freq_array = fft.rfftfreq(n=self.args.buff_size,
                                  d=(1 / self.args.sample_rate))
        mag_array = np.zeros_like(freq_array)
        closest_freq_index = np.searchsorted(freq_array, freq)
        mag_array[closest_freq_index] = ((10 ** (level / 20)
                                         * self.args.buff_size / 2))
        self.out_sig = np.fft.irfft(a=mag_array, n=self.args.buff_size)
        if retoggle:
            self.toggle_out()

        return (freq_array[closest_freq_index], level, sig_type)
コード例 #11
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def stftbins(x, Nwin, Nfft=None, d=1.0):
    import numpy.fft as fft

    """
    Time and frequency bins corresponding to short-time Fourier transform.
    Call this with the same arguments as `stft`, plus one extra argument: `d`
    sample spacing, to get the time and frequency axes that the output of
    `stft` correspond to.
    Parameters
    ----------
    x : array_like
        same as `stft`
    Nwin : int
        same as `stft`
    Nfft : int, optional
        same as `stft`
    d : float, optional
        Sample spacing of `x` (or 1 / sample frequency), units of seconds.
        Default: 1.0.
    Returns
    -------
    t : ndarray
        Array of length `len(x) // Nwin`, in units of seconds, corresponding to
        the first dimension (height) of the output of `stft`.
    f : ndarray
        Array of length `Nfft`, in units of Hertz, corresponding to the second
        dimension (width) of the output of `stft`.
    """
    Nfft = Nfft or Nwin
    Nwindows = x.size // Nwin
    t = np.arange(Nwindows) * (Nwin * d)
    f = fft.rfftfreq(Nfft, d)
    return t, f
コード例 #12
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 def beam_spectrum_freq_generation(self, n_sampling_fft = None):
     """
     Frequency array of the beam spectrum
     """
     if n_sampling_fft is None:
         n_sampling_fft = len(self.profile_array)
     self._beam_spectrum_freq = fft.rfftfreq(n_sampling_fft, self.bin_size)
コード例 #13
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def spectrum(data, dt=1, IsReal=False):
    """Calculate normalized spectrum s(f) and power spectrum w(f), for a given time series of signal ``data``. For real ``data``, The corresponding power spectrum :math:`w(f)= 2|s(f)|^2`. The coefficient 2 is because input signal is assumed real and the power for positive and negative frequencies are symmetric. So the power spectrum w(f) is specified only for positive f's

    :param data: The signal data in *time* domain
    :type data: 1d array of complex (or float if ``IsReal`` is ``True``)
    :param dt: (Optional) time step size of ``data``. Default to be 1.
    :type time: float
    :param bool IsReal: specify the type of ``data``. If ``True``, float data type is expected for ``data``, and :method:`np.fft.rfft` will be used to carry out FFT on ``data``. Default to be ``False``.

    :return: tuple containing spectrum s, power spectrum w, and frequency labels f.
    :rtype: (ndarray of complex, ndarray of float, ndarray of float)
    """

    if (IsReal):
        assert (isinstance(data[0],
                           (np.float, np.float128, np.float32, np.float16))
                )  #real data is assumed to be stored in float type
        n = len(data)
        s = rfft(data) / n
        w = 2 * s * np.conj(s)
        f = rfftfreq(n, dt)
        return (s, w, f)
    else:
        assert (isinstance(data[0], (np.complex, np.complex256, np.complex64)))
        n = len(data)
        s = fft(data) / n
        w = s * np.conj(s)
        f = fftfreq(n, dt)
        return (s, w, f)
コード例 #14
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def bandpass_gaussian(data, dt, period, alpha):
    """
    Bandpassing real data (in array *data*) with a Gaussian
    filter centered at *period* whose width is controlled
    by *alpha*:

      exp[-alpha * ((f-f0)/f0)**2],

    with f the frequency and f0 = 1 / *period*.
    *dt* is the sampling interval of the data.

    @type data: L{numpy.ndarray}
    @type dt: float
    @type period: float
    @type alpha: float
    @rtype: L{numpy.ndarray}
    """
    # Fourier transform
    fft_data = rfft(data)
    # aray of frequencies
    freq = rfftfreq(len(data), d=dt)

    # bandpassing data
    f0 = 1.0 / period
    fft_data *= np.exp(-alpha * ((freq - f0) / f0)**2)

    # back to time domain
    return irfft(fft_data, n=len(data))
コード例 #15
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def apply_notches(X, notches, rate, fft=True):
    if fft:
        fs = rfftfreq(X.shape[-1], 1. / rate)
        delta = 1.
        fd = rfft(X)
    else:
        nyquist = rate / 2.
        n_taps = 1001
        gain = [1, 1, 0, 0, 1, 1]
    for notch in notches:
        if fft:
            window_mask = np.logical_and(fs > notch - delta,
                                         fs < notch + delta)
            window_size = window_mask.sum()
            window = np.hamming(window_size)
            fd[:, window_mask] = (fd[:, window_mask] *
                                  (1. - window)[np.newaxis, :])
        else:
            freq = np.array([
                0, notch - 1, notch - .5, notch + .5, notch + 1, nyquist
            ]) / nyquist
            filt = firwin2(n_taps, freq, gain)
            X = filtfilt(filt, np.array([1]), X)
    if fft:
        X = irfft(fd)
    return X
コード例 #16
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ファイル: transform.py プロジェクト: gszep/sound-detection
def stft(signal, window=128, step=65):
    """
    Perform a shoft-time fourier transform with a Hann window
    https://en.wikipedia.org/wiki/Short-time_Fourier_transform

    Paramters
    ---------
    signal : <numpy.ndarray>
        array of shape length n_points of a single-channel signal

    window : int
        number of samples per window

    step : int
        stride of window, determines overlap

    Returns
    ---------
    time,frequency,stft : <numpy.ndarray>,<numpy.ndarray>,<numpy.ndarray>
        time and frequency mesh aligned with stft spectorgram
        assuming 44100Hz sampling rate

    ---------
    """

    signal -= signal.mean()
    freq = rfftfreq(window, d=1 / 44100.0)

    # construct hann windows
    windows, time = overlap(signal, window, step)
    windows *= 0.54 - 0.46 * cos(2 * pi * arange(window) / (window - 1))

    # return rfft along window axis
    return time, freq, rfft(windows).T
コード例 #17
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def gen_features3(x_raw,feat_divisions=[1e4,6e4,1.2e5,1.6e5,2e5,2.3e5,3e5],sample_freq = 4e6):
    """
    Helper function to generate simple statistical features and fft amplitudes
    from a 150,000 row 'acoustic data' segment.
    
    x_raw is the input acoustic_data pandas series
    feat_divisions is the ranges of frequencies in Hz to store fourier coefficients for.
    sample_freq is the sampling frequency of the provided data. The default is 4MHz.
    
    returns the converted features and the names of the names of the features.
    """
    
    feature_names = ['mean', 'std', 'max', 'min', 'skew', 'kurtosis']
    feature_names += ["fft_[{:},{:}]Hz".format(int(i),int(j)) for (i,j) in zip(feat_divisions,feat_divisions[1:])]
    
    features = [x_raw.mean(),
                x_raw.std(),
                x_raw.max(),
                x_raw.min(),
                x_raw.skew(),
                x_raw.kurtosis(),
            ]

    freq = rfftfreq(len(x_raw),d=1/sample_freq)
    fft = np.abs(rfft(x_raw))
    
    for i,j in zip(feat_divisions,feat_divisions[1:]):
        start = np.argmax(freq>i)
        stop = np.argmax(freq>j)
        features.append(np.max(fft[start:stop]))
        
    return features, feature_names
コード例 #18
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ファイル: ui.py プロジェクト: ricsirke/SoundFreqPlotter
    def init_plot(self):
        x = fft.rfftfreq(self.CHUNK_SIZE) * self.SAMPLE_RATE
        x_max = x[-1]
        self.init_y = linspace(0, x_max, len(x))
        y = self.init_y
        source = ColumnDataSource(data=dict(x=x, y=y))

        # TODO: range and size (toolbar), maybe could be user settings
        plot = Figure(plot_height=400,
                      plot_width=800,
                      title="freq anal",
                      tools="crosshair,pan,reset,resize,save,wheel_zoom",
                      x_range=[0, x_max],
                      y_range=[0, 15])

        rad = x_max / float(len(x))
        data = plot.circle('x',
                           'y',
                           source=source,
                           line_color=None,
                           radius=rad)
        self.data_source = data.data_source

        # TODO: maybe not the best place
        curdoc().add_root(plot)
コード例 #19
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def dfmap_fft(df, indep_var='Time'):
    if df is not None:
        df = df
    new_df = {}
    n = len(df[indep_var])
    a, b = df[indep_var][0], df[indep_var][n-1]
    for key in df:
        if key == indep_var:
            continue
        new_df[key] = rfft(df[key])
        new_df[key] = list(map(np.absolute, new_df[key]))
        new_df[key] = list(map(lambda x: 2*x/n, new_df[key]))

    # avg time between samples
    # assumes uniform sampling
    dt = (b-a) / n
    #dt = (b-a).total_seconds() / n

    # get list of frequencies from the fft
    # new_df['Freq'] = fftfreq(len(self.df[key]), dt)
    new_df['Freq'] = rfftfreq(n, dt)

    #
    new_df = pandas.DataFrame.from_dict(new_df)#.sort_values('Freq')
    #new_df = new_df[abs(new_df.Freq) > 0.4]

    return new_df
コード例 #20
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def periodogramfft(d, time, data):
    from scipy.signal import hann, periodogram
    import numpy.fft as fft
    print "d=", d

    pidx, pv, tidx, tv = peaks.turning_points(data, minidx=0)
    tts = time[pidx]
    rspl = UnivariateSpline(time, data, k=4, s=0)
    t0 = tts[1]
    tf = tts[d + 1]
    dt = time[1] - time[0]

    twin = np.arange(t0, tf, dt)
    N = len(twin)
    fs = 1 / dt
    #twin1 = np.arange(t0, tts[3], dt)
    #Nfft = len(twin1)
    #fs = N/(tf-t0)
    Nfft = 100000
    rwin = rspl(twin)
    rwin = rwin - np.mean(rwin)
    hwin = hann(len(twin))
    rwin = rwin * hwin
    #f, pgram = periodogram(hwin, fs=fs, nfft=Nfft)
    fftout = fft.rfft(rwin, 50 * len(rwin))
    fftfreq = fft.rfftfreq(50 * len(rwin), dt)
    print len(fftout)
    print len(fftfreq)
    return 2 * np.pi * fftfreq, np.abs(fftout)
コード例 #21
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def power_spectrum(da, dim=None):
    '''calculate the power spectrum of the given DataArray'''
    if not isinstance(da, xr.DataArray):
        da = xr.DataArray(da)

    if dim is None:
        dim = da.dims[-1]

    # xarray-version rfft
    da_f = xr.apply_ufunc(rfft,
                          da,
                          input_core_dims=[[dim]],
                          output_core_dims=[['freq']],
                          kwargs={'axis': -1})

    # power spectrum
    ps = da_f.real**2 + da_f.imag**2
    ps.name = 'power'

    # frequency coordinate
    ps['freq'] = xr.DataArray(rfftfreq(da[dim].size),
                              dims='freq',
                              attrs={'units': 'units: sample freq'})

    return ps
コード例 #22
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ファイル: timehistory.py プロジェクト: nprav/structpy
def low_pass_filter(signal, lp_frq, dt=0.005, time=None, zero_pad=True):
    """
    Function to low-pass filter a signal in the frequency domain,
    by setting all fourier terms larger than `lp_freq` to 0.

    :param signal: 1d list-like object with the input signal.
    :param lp_frq: Low-pass cut-off frequency.
    :param dt: float object with the timestep. Defaults to 0.005s.
                Only used if `time` is `None`.
    :param time: 1d list-like object with the input signal. Default to None.
    :param zero_pad: Boolean. If `True`, the signal is padded with zeros of
                    length `2*len(signal)`. This should be used if the input
                    is non-repetitive. IF `False`, the input is not zero-padded.
                    Hence, the fft assumes that the signal repeats.
    :return: 1D ndarray with the low-pass filtered signal.
    """

    if time is not None:
        dt = time[1] - time[0]
    if zero_pad:
        n = 3 * len(signal)
    else:
        n = len(signal)
    frq = rfftfreq(n, d=dt)
    signal_fft = rfft(signal, n=n)
    lp_fft = signal_fft * (frq <= lp_frq)
    lp_signal = irfft(lp_fft, n)[:len(signal)]
    return lp_signal
コード例 #23
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ファイル: pipeline.py プロジェクト: vrmarthi/gmrt-beam2dspec
def get_Period(dds_t, tsamp):
    N = np.int(nearest_power_of_2(dds_t.size))
    print('Padding ', dds_t.size, 'length array with ', N-dds_t.size, 'zeros.')
    DDS_T = np.abs(ft.rfft(dds_t-dds_t.mean(), N))
    nu = ft.rfftfreq(N, tsamp)
    zoom = 1
    while zoom == 1:
        pl.plot(nu, DDS_T)
        pl.xlabel('Frequency [Hz]')
        pl.ylabel('Flux')
        click = input('Ready to click ? :')
        if click == 'y':
            pl.title('Click on peak for F0')
            F0 = pl.ginput(1)
            zoom = 0
            break
        else:
            continue
            
    F0 = F0[0][0]
    P0 = 1.0 / F0
    pl.close()
    
    pl.plot(nu, DDS_T)
    pl.xlim(0.0, 12.5*F0)
    pl.xlabel('Frequency [Hz]')
    pl.ylabel('Flux')
    pl.title('First 12 harmonics')
    pl.savefig('First-12-harmonics.eps')
    pl.savefig('First-12-harmonics.png')
    return P0, F0
コード例 #24
0
ファイル: pipeline.py プロジェクト: vrmarthi/gmrt-beam2dspec
def ds_to_cs_to_fits(data, P, BW, filename):
 ntime,nchan=data.shape
 new_ntime=nearest_power_of_2(data.shape[0])
 new_nchan = nearest_power_of_2(data.shape[1])
 print(new_ntime, 'X', new_nchan, ' 2D FFT in progress...')
 cs=ft.fftshift(ft.rfft2(data, [new_ntime, new_nchan]),axes=0)
 print('FFT shift...')
 realCS=np.real(cs)
 imagCS=np.imag(cs)
 CS=np.zeros([2,cs.shape[0], cs.shape[1]])
 CS[0,:]=realCS
 CS[1,:]=imagCS
 tau = ft.rfftfreq(new_nchan,np.abs(BW)/nchan)
 fD=ft.fftshift(ft.fftfreq(new_ntime,P))*1e3
 hdu = f.PrimaryHDU(CS)
 hdu.header['CTYPE1'] ='DELAY - MICROSEC'
 hdu.header['CTYPE2'] ='DOPPLER FREQUENCY - MILLIHERTZ'
 hdu.header['CTYPE3'] ='REAL-IMAG'
 hdu.header['CRVAL1'] = tau[0]
 hdu.header['CRVAL2'] = fD[0]
 hdu.header['CRVAL3'] = 0
 hdu.header['CDELT1'] = tau[1]-tau[0]
 hdu.header['CDELT2'] = fD[1]-fD[0]
 hdu.header['CDELT3'] = 0
 hdu.header['CRPIX1'] = 0
 hdu.header['CRPIX2'] = 0
 hdu.header['CRPIX3'] = 0
 hdu.header['CROTA1'] = 0.0
 hdu.header['CROTA2'] = 0.0
 hdu.header['CROTA3'] = 0.0
 hdu.header['BSCALE'] = 1.0
 update_fits_header(hdu)
 hdu.header['INT_TIME']=P*ntime
 hdu.writeto(filename)
 return fD, tau, cs
コード例 #25
0
ファイル: TDS.py プロジェクト: camhough21/femtoPy
    def __init__(self,
                 refFile='ref.txt',
                 sampFile='samp.txt',
                 tFile='t.txt',
                 delta=1,
                 l=0.4,
                 c=0.299796):
        'load data files'
        self.refT0 = np.loadtxt(refFile)
        self.sampT0 = np.loadtxt(sampFile)
        self.sampT = np.copy(self.sampT0)
        self.t = np.loadtxt(tFile)

        'transforms'
        self.f0 = fft.rfftfreq(self.t.size, self.t[1] - self.t[0])
        self.w0 = 2 * np.pi * self.f0
        self.refF0 = fft.rfft(self.refT0)
        self.sampF0 = fft.rfft(self.sampT0)

        self.f = self.f0
        self.w = self.w0
        self.refF = self.refF0
        self.sampF = self.sampF0

        'experimental transfer function'
        self.transE = self.sampF0 / self.refF0

        'constants'
        self.nAir = np.ones(self.f.size)
        self.l = l
        self.delta = delta
        self.c = c

        return
コード例 #26
0
ファイル: TDS.py プロジェクト: camhough21/femtoPy
    def trimTime(self, tMin=-10, tMax=25, delta='undefined'):
        if tMin > np.amin(self.t):
            loc1 = np.amin(np.where(self.t > tMin))
        else:
            loc1 = 0
        if tMax < np.amax(self.t):
            loc2 = np.amin(np.where(self.t > tMax))
        else:
            loc2 = self.t.size

        self.refT0 = self.refT0[loc1:loc2]
        self.sampT0 = self.sampT0[loc1:loc2]
        self.sampT = np.copy(self.sampT0)
        self.t = self.t[loc1:loc2]

        self.f0 = fft.rfftfreq(self.t.size, self.t[1] - self.t[0])
        self.w0 = 2 * np.pi * self.f0
        self.refF0 = fft.rfft(self.refT0)
        self.sampF0 = fft.rfft(self.sampT0)

        self.f = self.f0
        self.w = self.w0
        self.refF = self.refF0
        self.sampF = self.sampF0

        self.transExp = self.sampF0 / self.refF0

        self.nAir = np.ones(self.f.size)

        if delta != 'undefined':
            self.delta = delta

        return
コード例 #27
0
def FFT(signal,
        NFFT=None,
        timeName='time',
        norm=None,
        freqName='frequency',
        real=True):
    """
    Automatically take the real or complex FFT of an xarray.DataArray You must
    supply the name of the frequency axis. Assumes that time is in seconds,
    and returns frequency in kHz. 
    """
    timeAxisNum = signal.get_axis_num(timeName)
    timeAxisData = getattr(signal, timeName).data
    dt = (timeAxisData[-1] - timeAxisData[0]) / (len(timeAxisData) - 1)
    if NFFT is None:
        NFFT = len(timeAxisData)
    if real:
        freq = rfftfreq(NFFT, dt) * 1e-3
        signalFFT = rfft(signal.data, n=NFFT, axis=timeAxisNum, norm=norm)
    else:
        freq = fftshift(fftfreq(NFFT, dt)) * 1e-3
        signalFFT = fftshift(fft(signal.data,
                                 n=NFFT,
                                 axis=timeAxisNum,
                                 norm=norm),
                             axes=timeAxisNum)
    #pack things back into a dataarray
    name = signal.name
    coords = dict(signal.coords.items())
    coords.pop(timeName)
    coords[freqName] = freq
    olddims = signal.dims
    newdims = olddims[0:timeAxisNum] + (freqName, ) + olddims[timeAxisNum + 1:]
    return xr.DataArray(signalFFT, coords=coords, name=name, dims=newdims)
コード例 #28
0
def compute_ensemble_fft(x_,u_data_):

    sample_size = size(u_data_[1,:]);
    aver_size = size(u_data_[:,1]);

    if sample_size%2==0:
        normalize_factor = (sample_size/2.)+1.
    elif sample_size%2==1:
        normalize_factor = (sample_size+1.)/2.
    
    sample_rate = 1/sample_size;
    k_freq = fft.rfftfreq(sample_size, sample_rate);

    #u_x = fft.irfft(fft.rfft(u_data_[0,:]),size(x_))
    #print('x: ',size(x_),', u_x: ', size(u_x))
    #pyplot.figure()
    #pyplot.plot(x_,u_x,'--r',x_,u_data_[0,:],'-k')
    #pyplot.show()
    
    u_amp = abs(fft.rfft(u_data_[0,:]))
    for i in range(1,aver_size):
        u_amp += abs(fft.rfft(u_data_[i,:]))

    u_amp = u_amp / (aver_size*normalize_factor)

    KEnerg = 0.5 * u_amp**2.0


    return k_freq, u_amp, KEnerg
コード例 #29
0
def test_run_copy_of_notebook_code():
    np.random.seed(123)
    Fs = 100  # sampling frequency in Hz
    Ts = 1 / Fs  # sampling interval in s
    N = 1024  # number of samples
    time = arange(0, N * Ts, Ts)  # time instants
    noise_std = 0.1  # signal noise standard deviation
    # time domain signal
    x = (sin(2 * pi * Fs / 10 * time) + sin(2 * pi * Fs / 5 * time) +
         random.randn(len(time)) * noise_std)

    # Apply DFT with propagation of uncertainties
    X, UX = GUM_DFT(x, noise_std**2)
    f = fft.rfftfreq(N, Ts)  # frequency values

    figure()
    plot(time, x)
    xlim(time[0], time[-1])
    xlabel("time / s", fontsize=18)
    ylabel("signal amplitude / au", fontsize=18)

    figure()
    subplot(211)
    errorbar(f, X[:len(f)], sqrt(UX[:len(f)]))
    ylabel("real part", fontsize=18)
    xticks([])
    subplot(212)
    errorbar(f, X[len(f):], sqrt(UX[len(f):]))
    ylabel("imaginary part", fontsize=18)
    xlabel("frequency / Hz", fontsize=18)
    subplots_adjust(hspace=0.05)
コード例 #30
0
ファイル: psutils.py プロジェクト: iceseismic/SeisSuite
def bandpass_gaussian(data, dt, period, alpha):
    """
    Bandpassing real data (in array *data*) with a Gaussian
    filter centered at *period* whose width is controlled
    by *alpha*:

      exp[-alpha * ((f-f0)/f0)**2],

    with f the frequency and f0 = 1 / *period*.
    *dt* is the sampling interval of the data.

    @type data: L{numpy.ndarray}
    @type dt: float
    @type period: float
    @type alpha: float
    @rtype: L{numpy.ndarray}
    """
    # Fourier transform
    fft_data = rfft(data)
    # aray of frequencies
    freq = rfftfreq(len(data), d=dt)

    # bandpassing data
    f0 = 1.0 / period
    fft_data *= np.exp(-alpha * ((freq - f0) / f0) ** 2)

    # back to time domain
    return irfft(fft_data, n=len(data))
コード例 #31
0
ファイル: timehistory.py プロジェクト: nprav/structpy
def get_rfft(signal, dt=0.005, time=None, zero_pad=True, zero_N=5):
    """
    Function to return the real absolute fft and frequencies of
    a given signal.

    :param signal: 1d list-like object with the input signal.
    :param dt: float object with the timestep. Defaults to 0.005s.
                Only used if `time` is `None`.
    :param time: 1d list-like object with the input signal. Default to None.
    :param zero_pad: Boolean. If `True`, the signal is padded with zeros of
                    length `2*len(signal)`. This should be used if the input
                    is non-repetitive. IF `False`, the input is not zero-padded.
                    Hence, the fft assumes that the signal repeats.
    :param zero_N:  float. If `zero_pad` is True, then the total length of the 
                    signal input to the FFT is zero_N * len(signal). The 
                    extra signal content is all zeros.
    :return:
        - frq - 1d ndarray with frequencies.
        - signal_fft - 1d ndarray with absolute fft values.
    """

    if time is not None:
        dt = time[1] - time[0]
    if zero_pad:
        extra = np.ones((zero_N - 1) * len(signal)) * signal[-1]
        # )np.linspace(
        #     signal[-1], signal[0], (zero_N-1)*len(signal)
        #     )
        signal = np.append(signal, extra)
        n = zero_N * len(signal)
    else:
        n = len(signal)
    frq = rfftfreq(n, d=dt)
    signal_fft = np.abs(rfft(signal, n=n))
    return frq, signal_fft
コード例 #32
0
def make_residual_func(samples, indices, **params):
    'closure for residual func'
    fft_size = 2
    while fft_size < indices[-1]:
        fft_size *= 2
    freqs = rfftfreq(fft_size, 5)
    ind_from = int(round(1 / (params['t_max'] * freqs[1])))
    ind_to = ind_from + params['n_harm']

    def make_series(x):
        'Calculates time series from parameterized spectrum'
        nonlocal freqs, ind_from, ind_to, params
        spectrum = zeros_like(freqs, 'complex')
        sign_x0 = 0 if x[0] == 0.5 else abs(x[0] - 0.5) / (x[0] - 0.5)
        spectrum[0] = rect(sign_x0 * exp(params['scale'][0] * abs(x[0] - 0.5)),
                           0)
        for i in range(ind_from, ind_to):
            spectrum[i] = rect(
                exp(params['scale'][1] * x[1] +
                    params['slope'] * log(freqs[i])),
                params['scale'][2] * x[2 + i - ind_from])
        return irfft(spectrum)

    def residual_func(x):
        'calculates sum of squared residuals'
        nonlocal samples, indices
        series = make_series(x)
        sum_err = 0
        for position, ind in enumerate(indices):
            sum_err += (series[ind] - samples[position])**2
        return sum_err

    return make_series, residual_func
コード例 #33
0
 def test_rfftfreq_2(self):
     n_points = 10000
     try:
         res = bm.rfftfreq(n_points)
     except AttributeError as e:
         self.skipTest('Not compiled with FFTW')
     np.testing.assert_almost_equal(res, fft.rfftfreq(n_points), 8)
コード例 #34
0
ファイル: pk1d.py プロジェクト: igmhub/picca
def compute_pk_raw(delta_lambda_or_log_lambda, delta, linear_binning=False):
    """Computes the raw power spectrum

    Args:
        delta_lambda_or_log_lambda: float
            Variation of (the logarithm of) the wavelength between two pixels
        delta: array of floats
            Mean transmission fluctuation (delta field)
        linear_binning: if set then inputs need to be in AA, outputs will be 1/AA
                        else inputs will be in log(AA) and outputs in s/km

    Returns:
        The following variables
            k: the Fourier modes the Power Spectrum is measured on
            pk: the Power Spectrum
    """
    if linear_binning:  # spectral length in AA
        length_lambda = (delta_lambda_or_log_lambda * len(delta))
    else:  # spectral length in km/s
        length_lambda = (delta_lambda_or_log_lambda * constants.SPEED_LIGHT *
                         np.log(10.) * len(delta))

    # make 1D FFT
    num_pixels = len(delta)
    fft_delta = rfft(delta)

    # compute power spectrum
    pk = (fft_delta.real**2 +
          fft_delta.imag**2) * length_lambda / num_pixels**2
    k = 2 * np.pi * rfftfreq(num_pixels, length_lambda / num_pixels)

    return k, pk
コード例 #35
0
ファイル: shear.py プロジェクト: TheisEizo/LensTools
	def decompose(self,l_edges,keep_fourier=False):

		"""
		Decomposes the shear map into its E and B modes components and returns the respective power spectral densities at the specified multipole moments

		:param l_edges: Multipole bin edges
		:type l_edges: array

		:param keep_fourier: If set to True, holds the Fourier transforms of the E and B mode maps into the E and B attributes of the ShearMap instance
		:type keep_fourier: bool. 

		:returns: :returns: tuple -- (l -- array,P_EE,P_BB,P_EB -- arrays) = (multipole moments, EE,BB power spectra and EB cross power)

		>>> test_map = ShearMap.load("shear.fit",format=load_fits_default_shear)
		>>> l_edges = np.arange(300.0,5000.0,200.0)
		>>> l,EE,BB,EB = test_map.decompose(l_edges)

		"""

		#Perform Fourier transforms
		ft_data1 = rfft2(self.data[0])
		ft_data2 = rfft2(self.data[1])

		#Compute frequencies
		lx = rfftfreq(ft_data1.shape[0])
		ly = fftfreq(ft_data1.shape[0])

		#Safety check
		assert len(lx)==ft_data1.shape[1]
		assert len(ly)==ft_data1.shape[0]

		#Compute sines and cosines of rotation angles
		l_squared = lx[np.newaxis,:]**2 + ly[:,np.newaxis]**2
		l_squared[0,0] = 1.0

		sin_2_phi = 2.0 * lx[np.newaxis,:] * ly[:,np.newaxis] / l_squared
		cos_2_phi = (lx[np.newaxis,:]**2 - ly[:,np.newaxis]**2) / l_squared

		#Compute E and B components
		ft_E = cos_2_phi * ft_data1 + sin_2_phi * ft_data2
		ft_B = -1.0 * sin_2_phi * ft_data1 + cos_2_phi * ft_data2

		ft_E[0,0] = 0.0
		ft_B[0,0] = 0.0

		assert ft_E.shape == ft_B.shape
		assert ft_E.shape == ft_data1.shape

		#Compute and return power spectra
		l = 0.5*(l_edges[:-1] + l_edges[1:])
		P_EE = _topology.rfft2_azimuthal(ft_E,ft_E,self.side_angle.to(deg).value,l_edges)
		P_BB = _topology.rfft2_azimuthal(ft_B,ft_B,self.side_angle.to(deg).value,l_edges)
		P_EB = _topology.rfft2_azimuthal(ft_E,ft_B,self.side_angle.to(deg).value,l_edges)

		if keep_fourier:
			self.fourier_E = ft_E
			self.fourier_B = ft_B

		return l,P_EE,P_BB,P_EB
コード例 #36
0
ファイル: gammatone.py プロジェクト: HuaxingXu/pyfilterbank
def freqz_fos(b, a, order, nfft, plotfun=None):
    impulse = _create_impulse(nfft)
    response, states = fosfilter(b, a, order, impulse)
    freqresponse = rfft(np.real(response))
    frequencies = rfftfreq(nfft)
    if plotfun:
        plotfun(frequencies, freqresponse)
    return freqresponse, frequencies, response
コード例 #37
0
def fourier_analysis(data, ptree=None):
    time = data['time']
    current = data['current']
    voltage = data['voltage']

    # inspect data
    n = len(time)  # normalization factor for fft
    assert len(current) == n
    assert len(voltage) == n
    d = time[1] - time[0]  # inverse of the sampling rate
    # check sampling spacing is the same everywhere
    for i in range(n - 1):
        assert isclose(time[i + 1] - time[i], d, atol=1e-10, rtol=1e-10)

    # truncate signals
    if ptree:
        steps_per_cycle = ptree.get_int('steps_per_cycle')
        cycles = ptree.get_int('cycles')
        ignore_cycles = ptree.get_int('ignore_cycles')
        assert cycles > ignore_cycles
        assert n == cycles * steps_per_cycle
        time = time[ignore_cycles * steps_per_cycle:]
        current = current[ignore_cycles * steps_per_cycle:]
        voltage = voltage[ignore_cycles * steps_per_cycle:]
    else:
        time = time[int(n / 2):]
        current = current[int(n / 2):]
        voltage = voltage[int(n / 2):]

    n = len(time)
    assert len(current) == n
    assert len(voltage) == n

    if not _is_power_of_two(n):
        warn(
            "(cycles-ignore_cycles)*steps_per_cycles is not a "
            "power of 2 (most efficient for the fourier analysis)",
            RuntimeWarning)

    # perform the actual fourrier analaysis
    fft_current = fft.rfft(current) / n
    fft_voltage = fft.rfft(voltage) / n
    fft_frequency = fft.rfftfreq(n, d)

    # find the excited harmonics
    if ptree:
        harmonics = array(ptree.get_array_int('harmonics'))
        peak_indices = harmonics * (cycles - ignore_cycles)
    else:
        mx, mn = peakdet(absolute(fft_voltage), mean(absolute(fft_current)))
        peak_indices = int(mx[:, 0])
        mx, mn = peakdet(absolute(fft_voltage), mean(absolute(fft_current)))
        assert peak_indices == mx[:, 0]

    frequency = fft_frequency[peak_indices]
    impedance = fft_voltage[peak_indices] / fft_current[peak_indices]

    return [frequency, impedance]
コード例 #38
0
ファイル: shear.py プロジェクト: TheisEizo/LensTools
	def fromEBmodes(cls,fourier_E,fourier_B,angle=3.14*deg):

		"""
		This class method allows to build a shear map specifying its E and B mode components

		:param fourier_E: E mode of the shear map in fourier space
		:type fourier_E: numpy 2D array, must be of type np.complex128 and must have a shape that is appropriate for a real fourier transform, i.e. (N,N/2 + 1); N should be a power of 2

		:param fourier_B: B mode of the shear map in fourier space
		:type fourier_B: numpy 2D array, must be of type np.complex128 and must have a shape that is appropriate for a real fourier transform, i.e. (N,N/2 + 1); N should be a power of 2

		:param angle: Side angle of the real space map in degrees
		:type angle: float.

		:returns: the corresponding ShearMap instance

		:raises: AssertionErrors for inappropriate inputs

		"""

		assert fourier_E.dtype == np.complex128 and fourier_B.dtype == np.complex128
		assert fourier_E.shape[1] == fourier_E.shape[0]/2 + 1
		assert fourier_B.shape[1] == fourier_B.shape[0]/2 + 1
		assert fourier_E.shape == fourier_B.shape

		#Compute frequencies
		lx = rfftfreq(fourier_E.shape[0])
		ly = fftfreq(fourier_E.shape[0])

		#Safety check
		assert len(lx)==fourier_E.shape[1]
		assert len(ly)==fourier_E.shape[0]

		#Compute sines and cosines of rotation angles
		l_squared = lx[np.newaxis,:]**2 + ly[:,np.newaxis]**2
		l_squared[0,0] = 1.0

		sin_2_phi = 2.0 * lx[np.newaxis,:] * ly[:,np.newaxis] / l_squared
		cos_2_phi = (lx[np.newaxis,:]**2 - ly[:,np.newaxis]**2) / l_squared

		sin_2_phi[0,0] = 0.0
		cos_2_phi[0,0] = 0.0

		#Invert E/B modes and find the components of the shear
		ft_data1 = cos_2_phi * fourier_E - sin_2_phi * fourier_B
		ft_data2 = sin_2_phi * fourier_E + cos_2_phi * fourier_B

		#Invert Fourier transforms
		data1 = irfft2(ft_data1)
		data2 = irfft2(ft_data2)

		#Instantiate new shear map class
		new = cls(np.array([data1,data2]),angle)
		setattr(new,"fourier_E",fourier_E)
		setattr(new,"fourier_B",fourier_B)

		return new
コード例 #39
0
ファイル: upsample_fft_full.py プロジェクト: psoll001/phys177
def main(input_val):
    currency_pairs = ["BTC_USD_", "LTC_USD_", "LTC_BTC_", "DRK_BTC_",
                      "DOGE_BTC_", "DRK_LTC_", "DOGE_LTC_"]
    
    file_list_uber = []
    
    for pair in currency_pairs:
        file_list_uber.append(pair + "full_ubersampled.csv")
    
    for file in file_list_uber:
        uber_csv = open(file, 'r')
        csv_read = csv.DictReader(uber_csv, delimiter=',',
                                  quoting=csv.QUOTE_NONNUMERIC)
        csv_list = []
        for row in csv_read:
            csv_list.append({'price':float(row['price']),
                             'tmsp':int(row['tmsp'])})            
                    
        first_tmsp = int(csv_list[0]['tmsp'])
        last_tmsp = int(csv_list[-1]['tmsp'])
        delta_tmsp = int(last_tmsp - first_tmsp)
            
        uber_len = len(csv_list)
            
        uber_price_array = np.zeros(uber_len, dtype='float')
            
        for i in range(0,uber_len,1):
            uber_price_array[i] = csv_list[i]['price']
            
        uber_fft = fft.rfft(uber_price_array)
        uber_fft_imag = np.imag(uber_fft)
        uber_fft_real = np.real(uber_fft)
        uber_fft_mag = np.sqrt((uber_fft_real * uber_fft_real) +
                               (uber_fft_imag * uber_fft_imag))
        
        uber_fft_bins = fft.rfftfreq(uber_len, d=100)
        
        uber_csv.close()
                
        field_names = ['freq_bin', 'magnitude']
        fft_csv_name = file[0:-4] + "_fft.csv"
        
        fft_csv = open(fft_csv_name, 'w', newline = '')
                
        csvwriter = csv.DictWriter(fft_csv, delimiter=',',fieldnames=field_names,
                                   quoting=csv.QUOTE_NONNUMERIC)
        csvwriter.writeheader()
        #csvwriter.writerows(full_conditioned)
                
        for i in range(0,len(uber_fft_real),1):
            dict_d = {'freq_bin':uber_fft_bins[i],
                      'magnitude':uber_fft_mag[i]}
            csvwriter.writerow(dict_d)
                    
        fft_csv.close()        
コード例 #40
0
def getSTFT(signal, sample_spacing=1, slidingWindowSize=20, stepSize=10) :
    """
    Get short-time fourier transform with slidingWindowSize=20 (number of sample in window) and stepSize=10 (number of samle to sample from window to another)
    """
    segmentedSignal=segmentSignal(signal,slidingWindowSize,stepSize)
    segmentedSignalSTFT=[]
    for segment in segmentedSignal : segmentedSignalSTFT.append(rfft(segment))
    spectrogram=np.array(segmentedSignalSTFT)
    t=np.array([stepSize*i for i in xrange(len(segmentedSignal))])
    f=np.array(rfftfreq(slidingWindowSize,d=sample_spacing))
    return t,f,spectrogram
コード例 #41
0
 def __init__(self, args):
     self.args = args
     # Determine the frequencies the dfft calculates at.
     self.freq = fft.rfftfreq(n=args.fft_size,
                              d=(1 / args.sample_rate))
     # Setup the display:
     self.fig = plt.figure()
     self.ax = plt.axes(xlim=args.xlims, ylim=args.ylims)
     self.ax.set_xscale('log', basex=10)
     self.line, = self.ax.plot([], [])
     plt.ion()
コード例 #42
0
ファイル: filters.py プロジェクト: EdwardBetts/iclrt_tools
def freq_analyze(data, noise, dt):
    plt.figure()
    plt.plot(data)
    plt.title('data to analyze')
    
    plt.figure()
    plt.title('noise to analyze')
    plt.plot(noise)
    
    data_FFT=rfft(data)
    data_freqs=rfftfreq(len(data), dt)
    noise_FFT=rfft(noise)
    noise_freqs=rfftfreq(len(noise), dt)
    
    plt.figure()
    plt.plot(data_freqs, data_FFT, label='data FFT')
    plt.plot(noise_freqs, noise_FFT, label='noise FFT')
    plt.title('FFT of noise and data')
    plt.yscale('log')
    plt.show()
コード例 #43
0
ファイル: widgets.py プロジェクト: aagnone3/audio_analysis
    def __init__(self, **kwargs):
        super(SpectralFrameWidget, self).__init__()

        # default parameters if not specified
        for name, val in self.defaults.items():
            setattr(self, name, kwargs.get(name, val))

        self.setLabel('left', 'Amplitude', units='dB')
        self.setLabel('bottom', text='Frequency', units='kHz')
        self.getPlotItem().setRange(yRange=[-50, 40])
        self.spectrum_curve = self.plot([])
        self.freqs = rfftfreq(self.nfft) * self.fs / 1000.0
        self.frame_size = get_frame_size(self.fs, self.frame_size_ms)
        self.window = np.hamming(self.frame_size)
コード例 #44
0
def simpleSpecs(t, sig, amponly=False):
    '''
    Creates simple amplitude and phase spectrums of real functions using numpy.fft; phase is in degrees
    :param t: time (1D numpy array)
    :param sig: signal (1D numpy array)
    :param amponly: outputs only the amplitude spectrum (bool)
    :return: frequency (1D numpy array), A(f), phi(f)
    '''
    n = sig.size
    fouSig = fft.rfft(sig) / n # DFT of the signal normalized
    sample_int = t[n-1] - t[n-2] # sample interval
    fouFrq = fft.rfftfreq(n, d=sample_int)
    fouAmp = sqrt(real(fouSig)**2 + imag(fouSig)**2)
    fouPhi = angle(fouSig, deg=True)
    if amponly is True:
        return fouFrq, fouAmp
    else:
        return fouFrq, fouAmp, fouPhi
コード例 #45
0
ファイル: ui.py プロジェクト: ricsirke/SoundFreqPlotter
	def init_plot(self):
		x = fft.rfftfreq(self.CHUNK_SIZE)*self.SAMPLE_RATE
		x_max = x[-1]
		self.init_y = linspace(0, x_max, len(x))
		y = self.init_y
		source = ColumnDataSource(data=dict(x=x, y=y))
		
		# TODO: range and size (toolbar), maybe could be user settings
		plot = Figure(plot_height=400, plot_width=800, title="freq anal",
					  tools="crosshair,pan,reset,resize,save,wheel_zoom",
					  x_range=[0, x_max], y_range=[0, 15])
					  
		rad = x_max/float(len(x))
		data = plot.circle('x', 'y', source=source, line_color=None, radius=rad)
		self.data_source = data.data_source
		
		# TODO: maybe not the best place
		curdoc().add_root(plot)
コード例 #46
0
ファイル: stft.py プロジェクト: sharkinyourcomputer/agui
def testResample():
  s, rate = ReadAndConvert(os.path.join('testdata', 'foo_s80_p95.wav'))
  s = scipy.signal.resample(s, 4*len(s))
  rate = 4*rate
  ConvertAndWrite(s, rate, os.path.join('testout', 'foo_resampled.wav'))
  s = s[0 : 15*2048]
  nfft, nskip = 1024, 512
  Sws = STFT(s, nfft, nskip)
  nrfft, nrskip = 4096, 2048
  target_pitch = 880. #440.0 * 2 ** (7.0 / 12)

  Rws = []
  for nw, Sw in enumerate(Sws):
    freqs = rfftfreq(nfft, 1.0/rate)
    epitch = EstimatePitch(Sw, rate)

    if epitch == 0.0: 
      p = 1
    else:
      p = target_pitch / epitch
      print '%d: epitch=%f p=%f' % (nw, epitch, p)

    PlotPitches(np.abs(Sw), rate, name='testResampleFrame%dInMag' % nw,
                title='Spectrum of input frame %d (%f)' % (nw, epitch))
    PlotPitches(np.angle(Sw), rate, name='testResampleFrame%dInPhase' % nw,
                title='Phase of input frame %d' % nw)

    Rw = Resample(Sw, nfft, nrfft, p, rate)

    Rws.append(Rw)

    erpitch = EstimatePitch(Rw, rate)
    PlotPitches(np.abs(Rw), rate, name='testResampleFrame%dOutMag' % nw,
                title='Spectrum of output frame %d (%f)' % (nw, erpitch))
    PlotPitches(np.angle(Rw), rate, name='testResampleFrame%dOutPhase' % nw,
                title='Phase of output frame %d' % nw)
  r = STIFT(Rws, nrfft, nrskip)
  r = scipy.signal.resample(r, int(np.floor(r.size/2)))
  rate = rate/2

  ConvertAndWrite(
      r, rate,
      os.path.join('testout', 'testResample.wav'))
コード例 #47
0
ファイル: kspace.py プロジェクト: tonyyli/imapper2
def real_to_fourier(f, x, y, z):
    # Check that real-space grid spacing is all equal
    if not (_is_evenly_spaced(x) and _is_evenly_spaced(y) and _is_evenly_spaced(z)):
        raise ValueError('Sample points in real space are not evenly spaced.')
    dx = x[1]-x[0]  # Grid spacing
    dy = y[1]-y[0]
    dz = z[1]-z[0]

    ftrans  = ft.rfftn(f)

    # Wavenumber arrays
    kx        = 2*np.pi * ft.fftfreq(x.size, d=dx)
    ky        = 2*np.pi * ft.fftfreq(y.size, d=dy)
    kz        = 2*np.pi * ft.rfftfreq(z.size, d=dz) # Only last axis is halved in length when using numpy.fft.rfftn()

    # Normalize (convert DFT to continuous FT)
    ftrans *= dx*dy*dz

    return ftrans, kx, ky, kz
コード例 #48
0
ファイル: analytics.py プロジェクト: KratosCSIC/trunk
    def PlotPSD(self):
        colors = ['r']
        dt = self.Dt
        Ry = self.measurements[0][0]
        mean = np.average(Ry)
        val = np.array( [v - mean for v in Ry] )
        freq = fft.rfftfreq(len(val), dt)
        tr = fft.rfft(val)
        tr_vect = [np.real(i) for i in tr]

        psd = [(np.real(i)**2 + np.imag(i)**2)**0.5 for i in tr]

        plt.clf()
        plt.figure("psd")
        plt.plot(freq[1:],tr_vect[1:],colors[0])
        plt.xlim(0, 10.0)
        plt.xlabel(r"$Frequency \, (Hz)$")
        plt.ylabel(r"$PSD$")
        plt.savefig("psd.eps")
コード例 #49
0
ファイル: timeseries.py プロジェクト: WanduiAlbert/gwpy
    def fft(self, nfft=None):
        """Compute the one-dimensional discrete Fourier transform of
        this `TimeSeries`.

        Parameters
        ----------
        nfft : `int`, optional
            length of the desired Fourier transform.
            Input will be cropped or padded to match the desired length.
            If nfft is not given, the length of the `TimeSeries`
            will be used

        Returns
        -------
        out : :class:`~gwpy.spectrum.Spectrum`
            the normalised, complex-valued FFT `Spectrum`.

        See Also
        --------
        :mod:`scipy.fftpack` for the definition of the DFT and conventions
        used.

        Notes
        -----
        This method, in constrast to the :meth:`numpy.fft.rfft` method
        it calls, applies the necessary normalisation such that the
        amplitude of the output :class:`~gwpy.spectrum.Spectrum` is
        correct.
        """
        from ..spectrum import Spectrum
        if nfft is None:
            nfft = self.size
        dft = npfft.rfft(self.value, n=nfft) / nfft
        dft[1:] *= 2.0
        new = Spectrum(dft, epoch=self.epoch, channel=self.channel,
                       unit=self.unit)
        new.frequencies = npfft.rfftfreq(self.size, d=self.dx.value)
        return new
コード例 #50
0
ファイル: noise.py プロジェクト: TheisEizo/LensTools
	def _fourierMap(self,power_func,**kwargs):

		#Assert the shape of the blueprint, to tune the right size for the fourier transform
		lpix = 360.0/self.side_angle.to(deg).value
		lx = rfftfreq(self.shape[0]) * self.shape[0] * lpix
		ly = fftfreq(self.shape[0]) * self.shape[0] * lpix

		#Compute the multipole moment of each FFT pixel
		l = np.sqrt(lx[np.newaxis,:]**2 + ly[:,np.newaxis]**2)

		#Compute the power spectrum at each l and check that it is positive 
		if isinstance(power_func,np.ndarray):
			
			#Check for correct shape
			assert power_func.shape[0] == 2,"If you want an interpolated power spectrum you should pass a (l,Pl) array!"

			#Perform the interpolation
			ell,Pell = power_func
			power_interp = interpolate.interp1d(ell,Pell,**kwargs)
			Pl = power_interp(l)

		else:
			
			Pl = power_func(l,**kwargs)
		

		assert Pl[Pl>=0.0].size == Pl.size

		#Generate real and imaginary parts
		real_part = np.sqrt(0.5*Pl) * np.random.normal(loc=0.0,scale=1.0,size=l.shape) * lpix/(2.0*np.pi)
		imaginary_part = np.sqrt(0.5*Pl) * np.random.normal(loc=0.0,scale=1.0,size=l.shape) * lpix/(2.0*np.pi)

		#Get map in real space and return
		ft_map = (real_part + imaginary_part*1.0j) * l.shape[0]**2
		ft_map[0,0] = 0.0

		return ft_map
コード例 #51
0
def smooth_fft(dx, spec, sigma):
    """Basic math for FFT convolution with a gaussian kernel.

    :param dx:
        The wavelength or velocity spacing, same units as sigma

    :param sigma:
        The width of the gaussian kernel, same units as dx

    :param spec:
        The spectrum flux vector
    """
    # The Fourier coordinate
    ss = rfftfreq(len(spec), d=dx)
    # Make the fourier space taper; just the analytical fft of a gaussian
    taper = np.exp(-2 * (np.pi ** 2) * (sigma ** 2) * (ss ** 2))
    ss[0] = 0.01  # hack
    # Fourier transform the spectrum
    spec_ff = np.fft.rfft(spec)
    # Multiply in fourier space
    ff_tapered = spec_ff * taper
    # Fourier transform back
    spec_conv = np.fft.irfft(ff_tapered)
    return spec_conv
コード例 #52
0
ファイル: slab.py プロジェクト: spectralDNS/mpiFFT4py
 def complex_local_wavenumbers(self):
     """Returns local wavenumbers of complex space"""
     return (fftfreq(self.N[0], 1./self.N[0]),
             fftfreq(self.N[1], 1./self.N[1])[self.complex_local_slice()[1]],
             rfftfreq(self.N[2], 1./self.N[2]))
コード例 #53
0
ファイル: QCUS_lib.py プロジェクト: jhuguetn/WAD_Python
    def fftAnalysis(self,cs,profile,mode='sensitivity'):
        """
        Fourier analysis: find main frequencies and power therein
        """
        if cs.verbose:
            try:
                import QCUS_testing as mytest
                if mode == 'sensitivity':
                    mytest._exportProfile( profile,fname='y2profile.tsv' )
                else:
                    mytest._exportProfile( profile,fname='x2profile.tsv' )
            except:
                pass

        fdelta = .1 # peak should differ by more than this fraction of max ps

        if mode == 'sensitivity':
            cs.sens_basewavelength = 0.
            cs.sens_ylim2 = -1

        # remove average component
        ywork = profile-np.average(profile)
        nx = len(ywork)
        x = np.array(range(nx))
        
        ffty = np.fft.rfft(ywork)
        ps = np.abs(ffty)**2
        integral = np.sum(ps[self.fft_skip:]) # calculate total content, but ignore zero component
        freqs = rfftfreq(nx)
            
        # find peaks
        xy_max,xy_min = wadwrapper_lib.peakdet(ps, delta=fdelta*np.max(ps[self.fft_skip:])) # returns indices for freqs

        # sort peaks from large to small
        xy_max = sorted(xy_max,key=operator.itemgetter(1))
        xy_max = xy_max[::-1]

        # find max component which is not zero-component and represents larger than given fraction of power
        base_ix = 0   # index of base-frequency
        start_ix = 0  # index of zero-peak
        for i,(xi,yi) in enumerate(xy_max):
            if xi<=self.fft_skip: # zero-peak must be located in skip
                start_ix = xi
            if xi>self.fft_skip and base_ix<1:
                base_ix = xi 
                if mode == 'sensitivity':
                    cs.sens_basewavelength = self.pixels2mm(cs, 1./freqs[xi])
                break

        # filter-out all non-background trend components to find extend of signal
        for i in range(len(ffty)):
            if i>(base_ix+start_ix)/2:
                ffty[i] = 0.

        # locate minimum (after zero-peak) of trend
        fy = np.fft.irfft(ffty)
        ix_max = np.argmax(fy)
        ix_min = ix_max+np.argmin(fy[ix_max:])
        if cs.verbose:
            print 'max @',ix_max
            print 'min @',ix_min

        if mode == 'sensitivity':
            cs.sens_ylim2 = ix_min
        
        if cs.verbose:
            plt.figure()
            plt.plot(ywork)
            plt.plot(fy)
            if mode == 'sensitivity':
                plt.title('sensitivity,filtered')
            else:
                plt.title('uniformity,filtered')
            plt.figure()
            plt.plot(ps)
            if mode == 'sensitivity':
                plt.title('sensitivity,powerspectrum')
            else:
                plt.title('uniformity,powerspectrum')
            cs.hasmadeplots = True
コード例 #54
0
ファイル: stft.py プロジェクト: sharkinyourcomputer/agui
def PlotPitches(S, rate, dirpath='testout', name='pitches', title=''):
  freqs = rfftfreq(S.size, 1.0/rate)
  plot.LinePlotHtml(
      dirpath, name, title,
      [['freq', name]] + [[f, Sk] for f, Sk in zip(freqs, S)],
      logx=True, xticks_at_A=True)
コード例 #55
0
ファイル: stft.py プロジェクト: sharkinyourcomputer/agui
def EstimatePitch(S, rate):
  "Very simple. Maximum value of abs(spectrum)."
  w = np.argmax(np.abs(S))
  r = rfftfreq(S.size, 1.0/rate)[w]
  return r
コード例 #56
0
ファイル: fourier.py プロジェクト: mkmcc/watch-calibration
def get_period(samples, rate, plot=False):
    n = samples.shape[0]
    minfreq = 1.0 * rate / n

    # plot the last 10 seconds of raw data
    #
    if plot:
        plt.figure(0)
        plt.subplot(2,1,1)
        plt.plot(np.linspace(0,10,int(10*rate)), samples[-int(10*rate):])

        plt.xlabel('time (s)')
        plt.ylabel('amplitude')


    # apply a threshold to cut out the noise
    #
    thresh = 3.0 * np.std(samples)
    samples = np.fabs(samples)

    idx = samples < thresh
    samples[idx] = 0.0


    # fourier transform the data to identify frequencies
    # ... window it with a gaussian envelope to make gaussian-shaped peaks
    # ... these can be accurately fit with gaussian profiles to centroid
    # ... precise frequency
    #
    w = signal.gaussian(samples.shape[0], std=samples.shape[0]/7)

    f1 = rfft(samples*w)
    f1 = np.abs(f1)
    f1 = f1 / np.max(f1)

    freqs = rfftfreq(samples.shape[0], 1.0/rate)


    # look in the region between 0.5 and 10 Hz.  if your clock doesn't
    # tick in the frequency range, you may be in trouble
    #
    lo = np.argmax(freqs >  0.5)    # or just calculate from minfreq...
    hi = np.argmax(freqs > 10.0)

    f1    = f1[lo:hi]
    freqs = freqs[lo:hi]


    # arbitrary threshold here...
    #
    tmp = np.select([f1>0.5*np.max(f1)], [f1])
    peaks = signal.find_peaks_cwt(tmp, np.arange(3,5))


    # plot the fourier transform, along with identified peaks
    #
    if plot:
        plt.subplot(2,2,3)
        plt.plot(freqs,f1)
        plt.scatter(freqs[peaks],f1[peaks],s=50)

        plt.xlim([0,10])
        plt.ylim([0,1.1])

        plt.xlabel('frequency (Hz)')
        plt.ylabel('power')


    # get a crude guess to the frequency, and cut the data around it
    #
    fguess = np.min(peaks)

    lo = fguess-10
    hi = fguess+10

    f1    = f1[lo:hi]
    freqs = freqs[lo:hi]

    fguess = np.argsort(f1)[-1] # find_peaks() doesn't always get the
                                # exact location


    # update frequency guess based on gaussian interpolation
    # (ie, fit a parabola to log(f1))
    #
    alpha = log(f1[fguess-1])
    beta  = log(f1[fguess])
    gamma = log(f1[fguess+1])

    a = 0.5 * (alpha - 2*beta + gamma)
    p = 0.5 * (alpha - gamma) / (alpha - 2*beta + gamma)
    b = beta - a * p**2


    # new plot, zoomed in on the peak, with a Gaussian fit to the profile
    #
    if plot:
        xs = np.arange(0,f1.shape[0],0.1)
        model = np.exp(a*(xs-p-fguess)**2 + b)

        plt.subplot(2,2,4)
        plt.plot(freqs,f1)
        plt.plot(xs*minfreq+freqs[0],model)
        plt.scatter(freqs[fguess],f1[fguess])
        plt.scatter(minfreq*(p+fguess) + freqs[0],np.max(model), color='r')

        plt.ylim([0,1.1])
        plt.xlim([np.min(freqs),np.max(freqs)])

        plt.xlabel('frequency (Hz)')
        plt.ylabel('power')


    fguessrefined = freqs[fguess] + p * minfreq

    return freqs[fguess], fguessrefined
コード例 #57
0
ファイル: fourier.py プロジェクト: bendudson/pyxpad
def fftp(item):
    """
    Calculate amplitude and phase as a function of frequency
    
    Inputs
    ------

    item  - an XPadDataItem object
    
    Returns
    -------
    
    amplitude, phase pair of XPadDataItem objects
    
    """

    if len(item.dim) != 1:
        raise ValueError("fftp can only operate on 1D traces currently")
    
    # Calculate FFT
    data = rfft(item.data)*(1./len(item.data))
    
    # Create a dimension
    dim = XPadDataDim()
    
    dim.name = "Frequency"
    
    step = (item.dim[0].data[1] - item.dim[0].data[0])
    dim.data = rfftfreq(len(item.data), step)

    dim.units = "1/"+item.dim[0].units
    if item.dim[0].units in ["s", "S", "sec", "Sec", "SEC"]:
        dim.data /= 1000.
        dim.units = "kHz"
    
    # Calculate the amplitude
    amp = XPadDataItem()
    if item.name != "":
        amp.name = "AMP( "+item.name+" )"
    amp.source = item.source
    if item.label != "":
        amp.label = "AMP( "+item.label+" )"
    amp.units = item.units

    amp.data = abs(data)
    
    amp.dim = [dim]
    
    # Calculate the phase
    phase = XPadDataItem()
    if item.name != "":
        phase.name = "PHASE( "+item.name+" )"
    phase.source = item.source
    if item.label != "":
        phase.label = "PHASE( "+item.label+" )"
    phase.units = "Radians"
    
    phase.data = arctan2(data.real, data.imag)

    a = phase.data - 2*pi
    for i in range(1,len(phase.data)):
        if abs(phase.data[i-1] - a[i]) < 1.:
            phase.data[i] = a[i]
            
    a = phase.data + 2*pi
    for i in range(1,len(phase.data)):
        if abs(phase.data[i-1] - a[i]) < 1.:
            phase.data[i] = a[i]
    
    phase.dim = [dim]
    
    return amp, phase
コード例 #58
0
ファイル: impedance.py プロジェクト: kiliakis/BLonD
    def __init__(self, Slices, impedance_source_list, frequency_resolution_input = None, 
                 freq_res_option = 'round', n_turns_memory = 0, recalculation_impedance = False, save_individual_voltages = False):
    

        #: *Copy of the Slices object in order to access the profile info.*
        self.slices = Slices
        
        #: *Impedance sources inputed as a list (eg: list of BBResonators objects)*
        self.impedance_source_list = impedance_source_list
        
        #: *Input frequency resolution in [Hz], the beam profile sampling for the spectrum
        #: will be adapted according to the freq_res_option.*
        self.frequency_resolution_input = frequency_resolution_input
        
        #: *Number of turns to be considered as memory for induced voltage calculation.*
        self.n_turns_memory = n_turns_memory
        
        #: *Length of one slice.*
        time_resolution = (self.slices.bin_centers[1] - self.slices.bin_centers[0])
        
        self.recalculation_impedance = recalculation_impedance
        
        if n_turns_memory==0:
            
            if self.frequency_resolution_input == None:
                self.n_fft_sampling = self.slices.n_slices
            else:    
                self.freq_res_option = freq_res_option
                if self.freq_res_option is 'round':
                    self.n_fft_sampling = next_regular(int(np.round(1/(self.frequency_resolution_input * time_resolution))))
                elif self.freq_res_option is 'ceil':
                    self.n_fft_sampling = next_regular(int(np.ceil(1/(self.frequency_resolution_input * time_resolution))))
                elif self.freq_res_option is 'floor':
                    self.n_fft_sampling = next_regular(int(np.floor(1/(self.frequency_resolution_input * time_resolution))))
                else:
                    raise RuntimeError('The input freq_res_option is not recognized')
                
                if self.n_fft_sampling < self.slices.n_slices:
                    print 'The input frequency resolution step is too big, and the whole \
                           bunch is not sliced... The number of sampling points for the \
                           FFT is corrected in order to sample the whole bunch (and \
                           you might consider changing the input in order to have \
                           a finer resolution).'
                    self.n_fft_sampling = next_regular(self.slices.n_slices)
                
            #: *Real frequency resolution in [Hz], according to the obtained n_fft_sampling.*
            self.frequency_resolution = 1 / (self.n_fft_sampling * time_resolution)
            
            #: *Frequency array of the impedance in [Hz]*
            self.frequency_array = rfftfreq(self.n_fft_sampling, self.slices.bin_centers[1] - self.slices.bin_centers[0])
                                           
            #: *Total impedance array of all sources in* [:math:`\Omega`]
            self.total_impedance = 0
            
            self.sum_impedances(self.frequency_array)
            
            self.save_individual_voltages = save_individual_voltages
            if self.save_individual_voltages:
                self.len_impedance_source_list = len(impedance_source_list)
                self.matrix_save_individual_impedances = np.zeros((self.len_impedance_source_list, len(self.frequency_array)))
                self.matrix_save_individual_voltages = np.zeros((self.len_impedance_source_list, self.slices.n_slices))
                for i in self.len_impedance_source_list:
                    self.matrix_save_individual_impedances[i,:] = impedance_source_list[i].impedance
            
            #: *Induced voltage from the sum of the wake sources in [V]*
            self.induced_voltage = 0
        
        else:
            
            self.n_turns_memory = n_turns_memory
            self.len_array_memory = (self.n_turns_memory+1) * self.slices.n_slices
            self.len_array_memory_extended = (self.n_turns_memory+2) * self.slices.n_slices
            self.n_points_fft = next_regular(self.len_array_memory_extended)
            self.frequency_array_memory = rfftfreq(self.n_points_fft, time_resolution)
            self.total_impedance_memory = np.zeros(self.frequency_array_memory.shape) + 0j
            self.time_array_memory = self.slices.bin_centers
            for i in range(1, self.n_turns_memory+1):
                self.time_array_memory = np.concatenate((self.time_array_memory, self.slices.bin_centers+(self.slices.edges[-1]-self.slices.edges[0])*i))
            
            for imped_object in self.impedance_source_list:
                imped_object.imped_calc(self.frequency_array_memory)
                self.total_impedance_memory += imped_object.impedance
コード例 #59
0
def getFFT(signal, sample_spacing=1) :
    """
    get (frequency, complex coefficient) list representing the spectrum of the signal (list of value sampled with the sample_spacing (1/sample_rate) (every 1s))
    """
    return zip(rfftfreq(len(signal),d=sample_spacing),rfft(signal))