librosa/core/spectrum.py
def iirt(y, sr=22050, win_length=2048, hop_length=None, center=True,
         tuning=0.0, pad_mode='reflect', flayout=None, **kwargs):
    r'''Time-frequency representation using IIR filters [1]_.

    This function will return a time-frequency representation
    using a multirate filter bank consisting of IIR filters.
    First, `y` is resampled as needed according to the provided `sample_rates`.
    Then, a filterbank with with `n` band-pass filters is designed.
    The resampled input signals are processed by the filterbank as a whole.
    (`scipy.signal.filtfilt` resp. `sosfiltfilt` is used to make the phase linear.)
    The output of the filterbank is cut into frames.
    For each band, the short-time mean-square power (STMSP) is calculated by
    summing `win_length` subsequent filtered time samples.

    When called with the default set of parameters, it will generate the TF-representation
    as described in [1]_ (pitch filterbank):

        * 85 filters with MIDI pitches [24, 108] as `center_freqs`.
        * each filter having a bandwith of one semitone.

    .. [1] Müller, Meinard.
           "Information Retrieval for Music and Motion."
           Springer Verlag. 2007.


    Parameters
    ----------
    y : np.ndarray [shape=(n,)]
        audio time series

    sr : number > 0 [scalar]
        sampling rate of `y`

    win_length : int > 0, <= n_fft
        Window length.

    hop_length : int > 0 [scalar]
        Hop length, number samples between subsequent frames.
        If not supplied, defaults to `win_length / 4`.

    center : boolean
        - If `True`, the signal `y` is padded so that frame
          `D[:, t]` is centered at `y[t * hop_length]`.
        - If `False`, then `D[:, t]` begins at `y[t * hop_length]`

    tuning : float in `[-0.5, +0.5)` [scalar]
        Tuning deviation from A440 in fractions of a bin.

    pad_mode : string
        If `center=True`, the padding mode to use at the edges of the signal.
        By default, this function uses reflection padding.

    flayout : string
        - If `ba`, the standard difference equation is used for filtering with `scipy.signal.filtfilt`.
          Can be unstable for high-order filters.
        - If `sos`, a series of second-order filters is used for filtering with `scipy.signal.sosfiltfilt`.
          Minimizes numerical precision errors for high-order filters, but is slower.

    kwargs : additional keyword arguments
        Additional arguments for `librosa.filters.semitone_filterbank()`
        (e.g., could be used to provide another set of `center_freqs` and `sample_rates`).

    Returns
    -------
    bands_power : np.ndarray [shape=(n, t), dtype=dtype]
        Short-time mean-square power for the input signal.

    Raises
    ------
    ParameterError
        If `flayout` is not None, `ba`, or `sos`.

    See Also
    --------
    librosa.filters.semitone_filterbank
    librosa.filters._multirate_fb
    librosa.filters.mr_frequencies
    librosa.core.cqt
    scipy.signal.filtfilt
    scipy.signal.sosfiltfilt

    Examples
    --------
    >>> import matplotlib.pyplot as plt
    >>> y, sr = librosa.load(librosa.util.example_audio_file())
    >>> D = np.abs(librosa.iirt(y))
    >>> librosa.display.specshow(librosa.amplitude_to_db(D, ref=np.max),
    ...                          y_axis='cqt_hz', x_axis='time')
    >>> plt.title('Semitone spectrogram')
    >>> plt.colorbar(format='%+2.0f dB')
    >>> plt.tight_layout()
    '''

    if flayout is None:
        warnings.warn('Default filter layout for `iirt` is `ba`, but will be `sos` in 0.7.',
                      FutureWarning)
        flayout = 'ba'

    elif flayout not in ('ba', 'sos'):
        raise ParameterError('Unsupported flayout={}'.format(flayout))

    # check audio input
    util.valid_audio(y)

    # Set the default hop, if it's not already specified
    if hop_length is None:
        hop_length = int(win_length // 4)

    # Pad the time series so that frames are centered
    if center:
        y = np.pad(y, int(hop_length), mode=pad_mode)

    # get the semitone filterbank
    filterbank_ct, sample_rates = semitone_filterbank(tuning=tuning, flayout=flayout, **kwargs)

    # create three downsampled versions of the audio signal
    y_resampled = []

    y_srs = np.unique(sample_rates)

    for cur_sr in y_srs:
        y_resampled.append(resample(y, sr, cur_sr))

    # Compute the number of frames that will fit. The end may get truncated.
    n_frames = 1 + int((len(y) - win_length) / float(hop_length))

    bands_power = []

    for cur_sr, cur_filter in zip(sample_rates, filterbank_ct):
        factor = float(sr) / float(cur_sr)
        win_length_STMSP = int(np.round(win_length / factor))
        hop_length_STMSP = int(np.round(hop_length / factor))

        # filter the signal
        cur_sr_idx = np.flatnonzero(y_srs == cur_sr)[0]

        if flayout == 'ba':
            cur_filter_output = scipy.signal.filtfilt(cur_filter[0], cur_filter[1],
                                                      y_resampled[cur_sr_idx])
        elif flayout == 'sos':
            cur_filter_output = scipy.signal.sosfiltfilt(cur_filter,
                                                         y_resampled[cur_sr_idx])

        # frame the current filter output
        cur_frames = util.frame(np.ascontiguousarray(cur_filter_output),
                                frame_length=win_length_STMSP,
                                hop_length=hop_length_STMSP)

        bands_power.append(factor * np.sum(cur_frames**2, axis=0)[:n_frames])

    return np.asarray(bands_power)
def iirt(y, sr=22050, win_length=2048, hop_length=None, center=True,
         tuning=0.0, pad_mode='reflect', flayout=None, **kwargs):
    r'''Time-frequency representation using IIR filters [1]_.

    This function will return a time-frequency representation
    using a multirate filter bank consisting of IIR filters.
    First, `y` is resampled as needed according to the provided `sample_rates`.
    Then, a filterbank with with `n` band-pass filters is designed.
    The resampled input signals are processed by the filterbank as a whole.
    (`scipy.signal.filtfilt` resp. `sosfiltfilt` is used to make the phase linear.)
    The output of the filterbank is cut into frames.
    For each band, the short-time mean-square power (STMSP) is calculated by
    summing `win_length` subsequent filtered time samples.

    When called with the default set of parameters, it will generate the TF-representation
    as described in [1]_ (pitch filterbank):

        * 85 filters with MIDI pitches [24, 108] as `center_freqs`.
        * each filter having a bandwith of one semitone.

    .. [1] Müller, Meinard.
           "Information Retrieval for Music and Motion."
           Springer Verlag. 2007.


    Parameters
    ----------
    y : np.ndarray [shape=(n,)]
        audio time series

    sr : number > 0 [scalar]
        sampling rate of `y`

    win_length : int > 0, <= n_fft
        Window length.

    hop_length : int > 0 [scalar]
        Hop length, number samples between subsequent frames.
        If not supplied, defaults to `win_length / 4`.

    center : boolean
        - If `True`, the signal `y` is padded so that frame
          `D[:, t]` is centered at `y[t * hop_length]`.
        - If `False`, then `D[:, t]` begins at `y[t * hop_length]`

    tuning : float in `[-0.5, +0.5)` [scalar]
        Tuning deviation from A440 in fractions of a bin.

    pad_mode : string
        If `center=True`, the padding mode to use at the edges of the signal.
        By default, this function uses reflection padding.

    flayout : string
        - If `ba`, the standard difference equation is used for filtering with `scipy.signal.filtfilt`.
          Can be unstable for high-order filters.
        - If `sos`, a series of second-order filters is used for filtering with `scipy.signal.sosfiltfilt`.
          Minimizes numerical precision errors for high-order filters, but is slower.

    kwargs : additional keyword arguments
        Additional arguments for `librosa.filters.semitone_filterbank()`
        (e.g., could be used to provide another set of `center_freqs` and `sample_rates`).

    Returns
    -------
    bands_power : np.ndarray [shape=(n, t), dtype=dtype]
        Short-time mean-square power for the input signal.

    Raises
    ------
    ParameterError
        If `flayout` is not None, `ba`, or `sos`.

    See Also
    --------
    librosa.filters.semitone_filterbank
    librosa.filters._multirate_fb
    librosa.filters.mr_frequencies
    librosa.core.cqt
    scipy.signal.filtfilt
    scipy.signal.sosfiltfilt

    Examples
    --------
    >>> import matplotlib.pyplot as plt
    >>> y, sr = librosa.load(librosa.util.example_audio_file())
    >>> D = np.abs(librosa.iirt(y))
    >>> librosa.display.specshow(librosa.amplitude_to_db(D, ref=np.max),
    ...                          y_axis='cqt_hz', x_axis='time')
    >>> plt.title('Semitone spectrogram')
    >>> plt.colorbar(format='%+2.0f dB')
    >>> plt.tight_layout()
    '''

    if flayout is None:
        warnings.warn('Default filter layout for `iirt` is `ba`, but will be `sos` in 0.7.',
                      FutureWarning)
        flayout = 'ba'

    elif flayout not in ('ba', 'sos'):
        raise ParameterError('Unsupported flayout={}'.format(flayout))

    # check audio input
    util.valid_audio(y)

    # Set the default hop, if it's not already specified
    if hop_length is None:
        hop_length = int(win_length // 4)

    # Pad the time series so that frames are centered
    if center:
        y = np.pad(y, int(hop_length), mode=pad_mode)

    # get the semitone filterbank
    filterbank_ct, sample_rates = semitone_filterbank(tuning=tuning, flayout=flayout, **kwargs)

    # create three downsampled versions of the audio signal
    y_resampled = []

    y_srs = np.unique(sample_rates)

    for cur_sr in y_srs:
        y_resampled.append(resample(y, sr, cur_sr))

    # Compute the number of frames that will fit. The end may get truncated.
    n_frames = 1 + int((len(y) - win_length) / float(hop_length))

    bands_power = []

    for cur_sr, cur_filter in zip(sample_rates, filterbank_ct):
        factor = float(sr) / float(cur_sr)
        win_length_STMSP = int(np.round(win_length / factor))
        hop_length_STMSP = int(np.round(hop_length / factor))

        # filter the signal
        cur_sr_idx = np.flatnonzero(y_srs == cur_sr)[0]

        if flayout == 'ba':
            cur_filter_output = scipy.signal.filtfilt(cur_filter[0], cur_filter[1],
                                                      y_resampled[cur_sr_idx])
        elif flayout == 'sos':
            cur_filter_output = scipy.signal.sosfiltfilt(cur_filter,
                                                         y_resampled[cur_sr_idx])

        # frame the current filter output
        cur_frames = util.frame(np.ascontiguousarray(cur_filter_output),
                                frame_length=win_length_STMSP,
                                hop_length=hop_length_STMSP)

        bands_power.append(factor * np.sum(cur_frames**2, axis=0)[:n_frames])

    return np.asarray(bands_power)
def iirt ( y , sr = 22050 , win_length = 2048 , hop_length = None , center = True , tuning = 0.0 , pad_mode = 'reflect' , flayout = None , * * kwargs ) : if flayout is None : warnings . warn ( 'Default filter layout for `iirt` is `ba`, but will be `sos` in 0.7.' , FutureWarning ) flayout = 'ba' elif flayout not in ( 'ba' , 'sos' ) : raise ParameterError ( 'Unsupported flayout={}' . format ( flayout ) ) # check audio input util . valid_audio ( y ) # Set the default hop, if it's not already specified if hop_length is None : hop_length = int ( win_length // 4 ) # Pad the time series so that frames are centered if center : y = np . pad ( y , int ( hop_length ) , mode = pad_mode ) # get the semitone filterbank filterbank_ct , sample_rates = semitone_filterbank ( tuning = tuning , flayout = flayout , * * kwargs ) # create three downsampled versions of the audio signal y_resampled = [ ] y_srs = np . unique ( sample_rates ) for cur_sr in y_srs : y_resampled . append ( resample ( y , sr , cur_sr ) ) # Compute the number of frames that will fit. The end may get truncated. n_frames = 1 + int ( ( len ( y ) - win_length ) / float ( hop_length ) ) bands_power = [ ] for cur_sr , cur_filter in zip ( sample_rates , filterbank_ct ) : factor = float ( sr ) / float ( cur_sr ) win_length_STMSP = int ( np . round ( win_length / factor ) ) hop_length_STMSP = int ( np . round ( hop_length / factor ) ) # filter the signal cur_sr_idx = np . flatnonzero ( y_srs == cur_sr ) [ 0 ] if flayout == 'ba' : cur_filter_output = scipy . signal . filtfilt ( cur_filter [ 0 ] , cur_filter [ 1 ] , y_resampled [ cur_sr_idx ] ) elif flayout == 'sos' : cur_filter_output = scipy . signal . sosfiltfilt ( cur_filter , y_resampled [ cur_sr_idx ] ) # frame the current filter output cur_frames = util . frame ( np . ascontiguousarray ( cur_filter_output ) , frame_length = win_length_STMSP , hop_length = hop_length_STMSP ) bands_power . append ( factor * np . sum ( cur_frames ** 2 , axis = 0 ) [ : n_frames ] ) return np . asarray ( bands_power )
r'''Time-frequency representation using IIR filters [1]_.

    This function will return a time-frequency representation
    using a multirate filter bank consisting of IIR filters.
    First, `y` is resampled as needed according to the provided `sample_rates`.
    Then, a filterbank with with `n` band-pass filters is designed.
    The resampled input signals are processed by the filterbank as a whole.
    (`scipy.signal.filtfilt` resp. `sosfiltfilt` is used to make the phase linear.)
    The output of the filterbank is cut into frames.
    For each band, the short-time mean-square power (STMSP) is calculated by
    summing `win_length` subsequent filtered time samples.

    When called with the default set of parameters, it will generate the TF-representation
    as described in [1]_ (pitch filterbank):

        * 85 filters with MIDI pitches [24, 108] as `center_freqs`.
        * each filter having a bandwith of one semitone.

    .. [1] Müller, Meinard.
           "Information Retrieval for Music and Motion."
           Springer Verlag. 2007.


    Parameters
    ----------
    y : np.ndarray [shape=(n,)]
        audio time series

    sr : number > 0 [scalar]
        sampling rate of `y`

    win_length : int > 0, <= n_fft
        Window length.

    hop_length : int > 0 [scalar]
        Hop length, number samples between subsequent frames.
        If not supplied, defaults to `win_length / 4`.

    center : boolean
        - If `True`, the signal `y` is padded so that frame
          `D[:, t]` is centered at `y[t * hop_length]`.
        - If `False`, then `D[:, t]` begins at `y[t * hop_length]`

    tuning : float in `[-0.5, +0.5)` [scalar]
        Tuning deviation from A440 in fractions of a bin.

    pad_mode : string
        If `center=True`, the padding mode to use at the edges of the signal.
        By default, this function uses reflection padding.

    flayout : string
        - If `ba`, the standard difference equation is used for filtering with `scipy.signal.filtfilt`.
          Can be unstable for high-order filters.
        - If `sos`, a series of second-order filters is used for filtering with `scipy.signal.sosfiltfilt`.
          Minimizes numerical precision errors for high-order filters, but is slower.

    kwargs : additional keyword arguments
        Additional arguments for `librosa.filters.semitone_filterbank()`
        (e.g., could be used to provide another set of `center_freqs` and `sample_rates`).

    Returns
    -------
    bands_power : np.ndarray [shape=(n, t), dtype=dtype]
        Short-time mean-square power for the input signal.

    Raises
    ------
    ParameterError
        If `flayout` is not None, `ba`, or `sos`.

    See Also
    --------
    librosa.filters.semitone_filterbank
    librosa.filters._multirate_fb
    librosa.filters.mr_frequencies
    librosa.core.cqt
    scipy.signal.filtfilt
    scipy.signal.sosfiltfilt

    Examples
    --------
    >>> import matplotlib.pyplot as plt
    >>> y, sr = librosa.load(librosa.util.example_audio_file())
    >>> D = np.abs(librosa.iirt(y))
    >>> librosa.display.specshow(librosa.amplitude_to_db(D, ref=np.max),
    ...                          y_axis='cqt_hz', x_axis='time')
    >>> plt.title('Semitone spectrogram')
    >>> plt.colorbar(format='%+2.0f dB')
    >>> plt.tight_layout()
r Time - frequency representation using IIR filters [ 1 ] _ .