def extract(file_list, train_scaler=False):
fns = np.loadtxt(file_list, dtype='str')
cur_batch_size = len(fns)
f_to_mel = filters.mel(sr=sampling_rate,
n_fft=nfft,
n_freq_bins=n_freq_bins)
print("Extracting features")
mp_func = partial(log_filterbank_energy,
output_dir="features",
sampling_rate=sampling_rate,
nfft=nfft,
n_freq_bins=n_freq_bins,
spectral_frame_length_s=spectral_frame_length_s,
frame_length_s=frame_length_s,
hop_length_s=hop_length_s,
force=False,
mel_scale=True)
feature_fns = mp_with_pbar(mp_func, fns, mp.cpu_count())
if train_scaler:
print("Training scaler")
scaler = StandardScaler()
for i, fn in tqdm(enumerate(feature_fns), total=len(feature_fns)):
with h5py.File(fn, 'r') as f:
spec = f['data']
scaler.partial_fit(spec[:, :, 0].T)
joblib.dump(scaler, "scaler.pkl")
return feature_fns
def fft_and_melscale(song,
nhop=512,
nffts=[1024, 2048, 4096],
mel_nband=80,
mel_freqlo=27.5,
mel_freqhi=16000.0,
include_zero_cross=False):
"""
fft and melscale method.
fft: nfft = [1024, 2048, 4096]; サンプルの切り取る長さを変えながらデータからnp.arrayを抽出して高速フーリエ変換を行う．
melscale: 周波数の次元を削減するとともに，log10の値を取っている．
"""
feat_channels = []
for nfft in nffts:
feats = []
window = signal.blackmanharris(nfft)
filt = mel(song.samplerate, nfft, mel_nband, mel_freqlo, mel_freqhi)
# get normal frame
frame = make_frame(song.data, nhop, nfft)
# melscaling
processedframe = fft(window * frame)[:, :nfft // 2 + 1]
processedframe = np.dot(filt, np.transpose(np.abs(processedframe)**2))
processedframe = 20 * np.log10(processedframe + 0.1)
feat_channels.append(processedframe)
if include_zero_cross:
song.zero_crossing = np.where(np.diff(np.sign(song.data)))[0]
print(song.zero_crossing)
res = np.array(feat_channels)
return res
def extract_f0_func_audiofile(audio_file, gender='M'):
floor_sp, ceil_sp = -80, 30
mel_basis = mel(16000, 1024, fmin=90, fmax=7600, n_mels=80).T
min_level = np.exp(-100 / 20 * np.log(10))
b, a = butter_highpass(30, 16000, order=5)
if gender == 'M':
lo, hi = 50, 250
elif gender == 'F':
lo, hi = 100, 600
else:
raise ValueError
prng = RandomState(0)
x, fs = sf.read(audio_file)
if(len(x.shape) >= 2):
x = x[:, 0]
if x.shape[0] % 256 == 0:
x = np.concatenate((x, np.array([1e-06])), axis=0)
y = signal.filtfilt(b, a, x)
wav = y * 0.95 + (prng.rand(y.shape[0]) - 0.5) * 1e-06
D = pySTFT(wav).T
D_mel = np.dot(D, mel_basis)
D_db = 20 * np.log10(np.maximum(min_level, D_mel)) - 16
S = (D_db + 100) / 100
f0_rapt = sptk.rapt(wav.astype(np.float32) * 32768, fs, 256, min=lo, max=hi, otype=2)
index_nonzero = (f0_rapt != -1e10)
tmp = f0_rapt[index_nonzero]
mean_f0, std_f0 = np.mean(tmp), np.std(tmp)
f0_norm = speaker_normalization(f0_rapt, index_nonzero, mean_f0, std_f0)
return S, f0_norm
def log_melsp_01(x,
sr=16000,
n_fft=1024,
hop_length=256,
n_mels=80,
fmin=80,
fmax=8000):
'''
'''
mel_basis = mel(sr, n_fft, fmin=fmin, fmax=fmax, n_mels=n_mels).T
min_level = np.exp(-100 / 20 * np.log(10))
b, a = butter_highpass(30, 16000, order=5)
#
# Remove drifting noise
y = signal.filtfilt(b, a, x)
# Ddd a little random noise for model roubstness
prng = RandomState()
wav = y * 0.96 + (prng.rand(y.shape[0]) - 0.5) * 1e-06
# Compute spect
D = pySTFT(wav, fft_length=n_fft, hop_length=hop_length).T
# Convert to mel and normalize
D_mel = np.dot(D, mel_basis)
D_db = 20 * np.log10(np.maximum(min_level, D_mel)) - 16
S = np.clip((D_db + 100) / 100, 0, 1)
return S.astype(np.float32)
def pncc(audio_wave,
n_fft=1024,
sr=16000,
window="hamming",
n_mels=40,
n_pncc=13,
weight_N=4,
power=2,
dct=True):
pre_emphasis_signal = scipy.signal.lfilter([1.0, -0.97], 1, audio_wave)
stft_pre_emphasis_signal = np.abs(
stft(pre_emphasis_signal, n_fft=n_fft, window=window))**power
mel_filter = np.abs(filters.mel(sr, n_fft=n_fft, n_mels=n_mels))**power
power_stft_pre_signal = np.dot(stft_pre_emphasis_signal.T, mel_filter.T)
q_ = medium_time_power_calculation(power_stft_pre_signal)
q_le = asymmetric_lawpass_filtering(q_, 0.999, 0.5)
pre_q_0 = q_ - q_le
q_0 = halfwave_rectification(pre_q_0)
q_f = asymmetric_lawpass_filtering(q_0)
q_th = temporal_masking(q_0)
r_sp = after_temporal_masking(q_th, q_f)
r_ = switch_excitation_or_non_excitation(r_sp=r_sp,
q_f=q_f,
q_le=q_le,
q_power_stft_pre_signal=q_)
s_ = weight_smoothing(r_=r_, q_=q_, N=weight_N)
t_ = time_frequency_normalization(p_=power_stft_pre_signal, s_=s_)
u_ = mean_power_normalization(t_, r_)
v_ = power_function_nonlinearity(u_)
dct_v = np.dot(filters.dct(n_pncc, v_.shape[1]), v_.T)
if dct:
return dct_v.T
else:
return v_.T
def __init__(
self,
n_fft: int = 512,
n_mels: int = 80,
sample_rate: int = 16000,
hop_length: int = 200,
f_max=8000, # default
f_min=0, # default
power=2.0, # default
win_length=None,
window='hann', # default
center=True,
pad_mode='reflect', # default
norm=None, # default for pytorch
htk=True # default for pytorch
):
self.n_fft = n_fft
self.sample_rate = sample_rate
self.pad_mode = pad_mode
self.hop_length = hop_length
self.power = power
self.win_length = n_fft
self.mel_basis = filters.mel(
sr=sample_rate,
n_fft=n_fft,
n_mels=n_mels, # mel filter
fmin=f_min, # mel filter
fmax=f_max, # mel filter
norm=norm, # mel filter
htk=htk)
self.fft_window = get_window(window, self.win_length,
fftbins=True).reshape((-1, 1))
def extract(audio_fn):
# Read and Resample the audio
try:
data, _ = librosa.core.load(audio_fn, sr=sampling_rate)
data = normalize(data)
except Exception as e:
logging.exception(e)
return None
# ensure length
if len(data) > duration:
data = data[:duration]
elif len(data) < duration:
data = np.pad(data, (duration - len(data), ),
mode='constant',
constant_values=0)
# spectrogram
f, t, Sxx = sp.signal.spectrogram(data,
fs=sampling_rate,
window=window,
nperseg=frame_length,
noverlap=overlap_length,
nfft=nfft)
if mel_scale:
# spectrogram -> log mel fb
f_to_mel = filters.mel(sr=sampling_rate,
n_fft=nfft,
n_mels=n_freq_bins)
Sxx = f_to_mel.dot(Sxx)
Sxx = np.expand_dims(np.log(1e-8 + Sxx), axis=-1)
return Sxx
def frft_MFCC(S,
fs,
n_mfcc=13,
n_mels=128,
dct_type=2,
norm='ortho',
power=2,
pic=None):
n_fft = 2 * (S.shape[0] - 1)
# Build a Mel filter
y = np.abs(S)**power
mel_basis = filters.mel(sr=fs,
n_fft=n_fft,
n_mels=n_mels,
fmin=0.0,
fmax=None,
htk=False,
norm=1)
melspectrogram = np.dot(mel_basis, y)
S_db = lib.core.power_to_db(melspectrogram)
feature = fftpack.dct(S_db, axis=0, type=dct_type, norm=norm)[:n_mfcc]
if pic is not None:
visual.specgram(X=feature,
title='frft_mfcc',
xlabel='Time',
ylabel='frft_mfccs',
pic=pic + '_frft_mfcc')
return feature
def apply_melfb(spec, fs, n_mels=128, amin=1e-10):
fbin = spec.shape[-1]
n_fft = fbin * 2 - 2
mfb = mel(fs, n_fft, n_mels=n_mels)
spec = np.maximum(spec, amin)
mspec = np.maximum(spec @ mfb.T, amin)
return mspec
def fftandmelscaleikkatsu(song,
nhop=512,
nffts=[1024, 2048, 4096],
mel_nband=80,
mel_freqlo=27.5,
mel_freqhi=16000.0,
include_zero_cross=False):
feat_channels = []
for nfft in nffts:
feats = []
window = signal.blackmanharris(nfft)
filt = mel(song.samplerate, nfft, mel_nband, mel_freqlo, mel_freqhi)
frame = Frame(song.data, nhop, nfft)
# frame = Frame2(data, nhop, nfft, nffts[-1])
print(frame.shape)
processedframe = fft(window * frame)[:, :nfft // 2 + 1]
processedframe = np.dot(filt, np.transpose(np.abs(processedframe)**2))
processedframe = 20 * np.log10(processedframe + 0.1)
# processedframe = normalize(processedframe, axis=1, copy=False)
print(processedframe.shape)
feat_channels.append(processedframe)
if include_zero_cross:
song.zero_crossing = np.where(np.diff(np.sign(song.data)))[0]
print(song.zero_crossing)
return np.array(feat_channels)
def test_mel_filterbank(N=15):
np.random.seed(12345)
i = 0
while i < N:
fs = np.random.randint(50, 10000)
n_filters = np.random.randint(2, 20)
window_len = np.random.randint(10, 100)
norm = np.random.randint(2)
mine = mel_filterbank(window_len,
n_filters,
fs,
min_freq=0,
max_freq=None,
normalize=bool(norm))
theirs = mel(
fs,
n_fft=window_len,
n_mels=n_filters,
htk=True,
norm=norm if norm == 1 else None,
)
np.testing.assert_almost_equal(mine, theirs)
print("PASSED")
i += 1
def __init__(self, filter_length=1024, hop_length=256, win_length=1024, n_mel_channels=80, sampling_rate=22050, mel_fmin=0.0, mel_fmax=8000.0):
super(TacotronSTFT, self).__init__()
self.n_mel_channels = n_mel_channels
self.sampling_rate = sampling_rate
self.stft_fn = STFT(filter_length, hop_length, win_length)
mel_basis = mel(sampling_rate, filter_length, n_mel_channels, mel_fmin, mel_fmax)
self.mel_basis = nd.array(mel_basis)
def _wav_to_spec(self,
wav,
sample_rate,
wav_path=None,
introduce_noise=False):
"""Convert wav file to a mel spectrogram
Args:
wav (numpy array): audio data either 1-d (mono) or 2-d (stereo)
sample_rate (int): the sampling rate of the .wav (sf.read[1])
wav_path (str): Path to original wav file
note that these two variables can be loaded using:
wavfile, sample_rate = sf.read(os.path.join(input_dir, speaker, fileName))
Returns:
np.array: Mel spectrogram
"""
mel_basis = mel(Config.audio_sr,
Config.n_fft,
fmin=Config.fmin,
fmax=Config.fmax,
n_mels=Config.n_mels).T
min_level = np.exp(Config.min_level_db / 20 * np.log(10))
b, a = self._butter_highpass(30, Config.audio_sr, order=5)
# Resample wav if needed
if sample_rate != Config.audio_sr:
wav = librosa.resample(wav, sample_rate, Config.audio_sr)
print(
f"Wav file with sr {sample_rate} != {Config.audio_sr}, Now resampling to {Config.audio_sr}, then try to write to {wav_path}"
)
if wav_path:
sf.write(wav_path, wav,
Config.audio_sr) # Write downsampled file
# Remove drifting noise
wav = signal.filtfilt(b, a, wav)
# add a little random noise for model robustness
if introduce_noise:
log.info(f"Introducing random noise into wav.file")
wav = wav * 0.96 + (self._prng.rand(wav.shape[0]) - 0.5) * 1e-06
# Compute spectrogram
D = self._pySTFT(wav,
fft_length=Config.n_fft,
hop_length=Config.hop_length).T
# Convert to mel and normalize
D_mel = np.dot(D, mel_basis)
D_db = 20 * np.log10(np.maximum(
min_level, D_mel)) - Config.ref_level_db # amp to db
S = np.clip((D_db - Config.min_level_db) / -Config.min_level_db, 0,
1) # clip between 0-1
return S
def pncc(audio_wave,
n_fft=512,
sr=16000,
winlen=0.020,
winstep=0.010,
n_mels=128,
n_pncc=13,
weight_N=4,
power=2):
pre_emphasis_signal = scipy.signal.lfilter([1.0, -0.97], 1, audio_wave)
mono_wave = to_mono(pre_emphasis_signal.T)
stft_pre_emphasis_signal = np.abs(
stft(mono_wave,
n_fft=n_fft,
hop_length=int(sr * winstep),
win_length=int(sr * winlen),
window=np.ones(int(sr * winlen)),
center=False))**power
mel_filter = np.abs(filters.mel(sr, n_fft=n_fft, n_mels=n_mels))**power
power_stft_signal = np.dot(stft_pre_emphasis_signal.T, mel_filter.T)
medium_time_power = medium_time_power_calculation(power_stft_signal)
lower_envelope = asymmetric_lawpass_filtering(medium_time_power, 0.999,
0.5)
subtracted_lower_envelope = medium_time_power - lower_envelope
rectified_signal = halfwave_rectification(subtracted_lower_envelope)
floor_level = asymmetric_lawpass_filtering(rectified_signal)
temporal_masked_signal = temporal_masking(rectified_signal)
final_output = switch_excitation_or_non_excitation(temporal_masked_signal,
floor_level,
lower_envelope,
medium_time_power)
spectral_weight_smoothing = weight_smoothing(final_output,
medium_time_power,
L=n_mels)
transfer_function = time_frequency_normalization(
power_stft_signal, spectral_weight_smoothing)
normalized_power = mean_power_normalization(transfer_function,
final_output,
L=n_mels)
power_law_nonlinearity = power_function_nonlinearity(normalized_power)
dct = np.dot(power_law_nonlinearity,
filters.dct(n_pncc, power_law_nonlinearity.shape[1]).T)
return dct
def __init__(self, n_mels, sample_rate, filter_length, hop_length,
win_length=None, mel_fmin=0.0, mel_fmax=None):
super(MelSpectrogram, self).__init__()
self.stft = STFT(filter_length, hop_length, win_length)
mel_basis = mel(sample_rate, filter_length, n_mels,
mel_fmin, mel_fmax, htk=True)
mel_basis = torch.from_numpy(mel_basis).float()
self.register_buffer('mel_basis', mel_basis)
def get_filters(config: Dict):
mel_basis = mel(config["rate"],
config["window"],
fmin=90,
fmax=config["fmax"],
n_mels=config["mels"]).T
min_level = np.exp(-100 / 20 * np.log(10))
b, a = butter_highpass(30, config["rate"], order=5)
return mel_basis, min_level, b, a
def __init__(self, frame_stream, specfmt="dB", mels_N=12):
'''
DFTStream(frame_stream, specfmt, mels_N)
Create a stream of discrete Fourier transform (DFT) frames using the
specified sample frame stream. Only bins up to the Nyquist rate are
returned in the stream Optional arguments:
specfmt - DFT output:
"complex" - return complex DFT results
"dB" [default] - return power spectrum 20log10(magnitude)
"mag^2" - magnitude squared spectrum
"Mel" - melodic scale
mels_N - Number of Mel filters to use. Only applicable when
specfmt == "Mel".
'''
self.format_types = {"complex" : 0,
"mag^2" : 1,
"dB" : 2,
"Mel" : 3}
self.framer = frame_stream
self.frame_len = frame_stream.get_framelen_samples()
try:
self.format = self.format_types[specfmt]
except KeyError:
raise ValueError("Unknown specfmt {}. Use one of [{}]".format(
specfmt, ", ".join(self.format_types.keys())))
# Number of frequency bins is the same as the number of bins in the
# frame
self.dft_bins = self.frame_len
# Only bins up to the Nyquist rate are usable. The DFT routine that
# we are using will return up to and including the Nyuist (half bins
# plus 1 if even)
self.Nyquist_Hz = self.framer.get_Fs() / 2.0
# We add 1.1 instead of 1, see numpy.around for details which
# np.round uses.
self.bins_Nyquist = np.int(np.round((self.frame_len+1.1)/2.0))
self.window = signal.get_window("hamming", self.frame_len)
if self.format == self.format_types["Mel"]:
# Construct Mel filters
self.mel_filters = mel(self.framer.get_Fs(),
self.dft_bins, mels_N)
# Center frequencies of the Mel filters in Hz
# Returns two more than are actually used (0 Hz and Nyquist)
self.bins_Hz = mel_frequencies(mels_N+2,
fmin=0, fmax=self.Nyquist_Hz)
self.bins_Hz = self.bins_Hz[1:-1] # Remove ends
self.bins_N = len(self.bins_Hz)
else:
self.bins_Hz = np.arange(self.bins_Nyquist) / self.bins_Nyquist * self.Nyquist_Hz
self.bins_N = self.bins_Hz.shape[0]
def get_coeffs(self,A,num_ceps=13,num_filters=16,f_bins=400,fs=100,normalize=True,corr=False):
fbank = filters.mel(fs,f_bins,num_filters, norm=None)
fbank_coeffs = np.dot(fbank,A).T
cc = fftpack.dct(fbank_coeffs, type=2, norm='ortho')[:, 1 : (num_ceps + 1)]
if normalize == True:
cc -= (np.mean(cc, axis=0) + 1e-8)
fbank_coeffs -= (np.mean(fbank_coeffs, axis=0) + 1e-8)
if corr == False:
return cc
else:
return fbank_coeffs
def __init__(self, sampling_rate: int = 22050, n_fft: int = 1024, window_size: int = 1024, hop_size: int = 256,
num_mels: int = 80, fmin: float = 0., fmax: float = 8000.):
super().__init__()
self.n_fft = n_fft
self.hop_size = hop_size
self.window_size = window_size
self.pad_size = (self.n_fft - self.hop_size) // 2
mel_filter_tensor = torch.FloatTensor(mel(sampling_rate, n_fft, num_mels, fmin, fmax))
self.register_buffer('mel_filter', mel_filter_tensor)
self.register_buffer('window', torch.hann_window(window_size))
def mel_scaled_spectrogram(spectrogram: ndarray,
sr: int,
n_mels: Optional[int] = 128,
fmin: Optional[float] = 0.0,
fmax: Optional[Union[float, None]] = None,
htk: Optional[bool] = False):
"""Calculates the mel scaled version of the spectrogram.
:param spectrogram: Spectrogram to be used.
:type spectrogram: numpy.ndarray
:param sr: Sampling frequency of the original signal.
:type sr: int
:param n_mels: Amount of mel filters to use, defaults to 128.
:type n_mels: int, optional
:param fmin: Minimum frequency for mel filters, defaults to 0.0.
:type fmin: float, optional
:param fmax: Maximum frequency for mel filters. If `None`, \
sr/2.0 is used. Defaults to None
:type fmax: float|None, optional
:param htk: Use HTK formula, instead of Slaney, defaults to False.
:type htk: bool, optional
:return: Mel scaled version of the input spectrogram, with shape \
(channels, nb_mels, values) for channels >= 2, else \
(nb_mels, values).
:rtype: numpy.ndarray
"""
ndim = spectrogram.ndim
if ndim not in [2, 3]:
raise AttributeError('Input spectrogram must be of shape '
'(channels, nb_frames, frames). '
f'Current input has {ndim} dimensions. '
f'Allowed are either 2 or 3.')
n_fft = 2 * (spectrogram[ndim - 2] - 1)
mel_filters = mel(
sr=sr,
n_fft=n_fft,
n_mels=n_mels,
fmin=fmin,
fmax=fmax,
htk=htk)
if ndim == 2:
mel_spectrogram = np_dot(mel_filters, spectrogram)
else:
mel_spectrogram = np_cat([expand_dims(np_dot(mel_filters, i), 0)
for i in spectrogram], axis=0)
return mel_spectrogram
def melspectrogram(y=None,
sr=16000,
n_fft=400,
hop_length=160,
power=2.0,
**kwargs):
"""Compute a mel-scaled spectrogram.
If a spectrogram input `S` is provided, then it is mapped directly onto
the mel basis `mel_f` by `mel_f.dot(S)`.
If a time-series input `y, sr` is provided, then its magnitude spectrogram
`S` is first computed, and then mapped onto the mel scale by
`mel_f.dot(S**power)`. By default, `power=2` operates on a power spectrum.
Parameters
----------
y : np.ndarray [shape=(n,)] or None
audio time-series
sr : number > 0 [scalar]
sampling rate of `y`
n_fft : int > 0 [scalar]
length of the FFT window
hop_length : int > 0 [scalar]
number of samples between successive frames.
See `librosa.core.stft`
power : float > 0 [scalar]
Exponent for the magnitude melspectrogram.
e.g., 1 for energy, 2 for power, etc.
kwargs : additional keyword arguments
Mel filter bank parameters.
See `librosa.filters.mel` for details.
Returns
-------
S : np.ndarray [shape=(n_mels, t)]
Mel spectrogram
"""
# Compute a magnitude spectrogram from input
S = np.abs(stft(y, n_fft=n_fft, hop_length=hop_length,
center=False))**power
# Build a Mel filter
mel_basis = filters.mel(sr, n_fft, **kwargs)
return np.dot(mel_basis, S)
def analyze(audio, time):
"""
"""
# print(time)
n_fft = int(44100 * 0.5)
mel_basis = filters.mel(44100, n_fft, n_mels=256)
# print(mel_basis.shape)
start_idx = int(44100 * time) #-n_fft/2)
spec = np.log10(np.abs(np.fft.fft(audio[start_idx:start_idx + n_fft]))**2)
spec = spec[:int(len(spec) / 2) + 1]
# print(spec.shape)
mel_spec = np.dot(mel_basis, spec)
norm = np.linalg.norm(mel_spec, ord=2)
return mel_spec / norm, norm
def pncc(audio_wave,
n_fft=1024,
sr=16000,
window="hamming",
n_mels=40,
n_pncc=13,
weight_N=4,
power=2,
dct=True):
pre_emphasis_signal = scipy.signal.lfilter([1.0, -0.97], 1, audio_wave)
stft_pre_emphasis_signal = np.abs(
stft(pre_emphasis_signal, n_fft=n_fft, window=window))**power
mel_filter = np.abs(filters.mel(sr, n_fft=n_fft, n_mels=n_mels))**power
power_stft_signal = np.dot(stft_pre_emphasis_signal.T, mel_filter.T)
medium_time_power = medium_time_power_calculation(power_stft_signal)
lower_envelope = asymmetric_lawpass_filtering(medium_time_power, 0.999,
0.5)
subtracted_lower_envelope = medium_time_power - lower_envelope
rectified_signal = halfwave_rectification(subtracted_lower_envelope)
floor_level = asymmetric_lawpass_filtering(rectified_signal)
temporal_masked_signal = temporal_masking(rectified_signal)
temporal_masked_signal = after_temporal_masking(temporal_masked_signal,
floor_level)
final_output = switch_excitation_or_non_excitation(temporal_masked_signal,
floor_level,
lower_envelope,
medium_time_power)
spectral_weight_smoothing = weight_smoothing(final_output,
medium_time_power, weight_N)
transfer_function = time_frequency_normalization(
power_stft_signal=power_stft_signal,
spectral_weight_smoothing=spectral_weight_smoothing)
normalized_power = mean_power_normalization(transfer_function,
final_output)
power_law_nonlinearity = power_function_nonlinearity(normalized_power)
dct_v = np.dot(filters.dct(n_pncc, power_law_nonlinearity.shape[1]),
power_law_nonlinearity.T)
return power_law_nonlinearity
def __init__(self, sample_rate, preemphasis, frequency, frame_length,
frame_shift, min_dbs, ref_dbs, mels_size, griff_lim_iters,
power):
self.preemphasis = preemphasis
self.n_fft = (frequency - 1) * 2
self.win_length = int(frame_length / 1e3 * sample_rate)
self.hop_length = int(frame_shift / 1e3 * sample_rate)
self.min_dbs = min_dbs
self.ref_dbs = ref_dbs
self.griff_lim_iters = griff_lim_iters
self.power = power
# Create a Filterbank matrix to combine FFT bins into Mel-frequency bins
self.mel_basis = filters.mel(sr=sample_rate,
n_fft=self.n_fft,
n_mels=mels_size)
def transform2mel(spectrogram,
samplerate,
fft_window_size,
n_mel_bands=80,
freq_min=0,
freq_max=None):
'''Transform to Mel
convert a spectrogram to a Mel scale spectrogram by grouping original frequency bins
to Mel frequency bands (using Mel filter from Librosa)
Parameters
spectrogram: input spectrogram
samplerate: samplerate of audio signal
fft_window_size: number of time window / frequency bins in the FFT analysis
n_mel_bands: number of desired Mel bands, typically 20, 40, 80 (max. 128 which is default when 'None' is provided)
freq_min: minimum frequency (Mel filters will be applied >= this frequency, but still return n_meld_bands number of bands)
freq_max: cut-off frequency (Mel filters will be applied <= this frequency, but still return n_meld_bands number of bands)
Returns:
mel_spectrogram: Mel spectrogram: np.array of shape(n_mel_bands,frames) maintaining the number of frames in the original spectrogram
'''
from librosa.filters import mel
# Syntax: librosa.filters.mel(sr, n_fft, n_mels=128, fmin=0.0, fmax=None, htk=False)
mel_basis = mel(samplerate,
fft_window_size,
n_mels=n_mel_bands,
fmin=freq_min,
fmax=freq_max)
freq_bin_max = mel_basis.shape[1] # will be fft_window_size / 2 + 1
# IMPLEMENTATION WITH FOR LOOP
# initialize Mel Spectrogram matrix
#n_mel_bands = mel_basis.shape[0] # get the number of bands from result in case 'None' was specified as parameter
#mel_spectrogram = np.empty((n_mel_bands, frames))
#for i in range(frames): # stepping through the wave segment, building spectrum for each window
# mel_spectrogram[:,i] = np.dot(mel_basis,spectrogram[0:freq_bin_max,i])
# IMPLEMENTATION WITH DOT PRODUCT (15% faster)
# multiply the mel filter of each band with the spectogram frame (dot product executes it on all frames)
# filter will be adapted in a way so that frequencies beyond freq_max will be discarded
mel_spectrogram = np.dot(mel_basis, spectrogram[0:freq_bin_max, :])
return (mel_spectrogram)
def __init__(self, dlnet_config: dict, ds_config: str):
"""
Init wrapper object. Reads DL Network config and dataset config.
Parameters
----------
dlnet_config : dict
Config for DL Network and preprocessing
ds_config : str
Path to Dataset config to extract classes
"""
# Set config:
self.config = dlnet_config
# Set random seed:
random.seed = self.config['random_seed']
# Classes
if self.config['binary']:
self.config['classes'] = ['compressed_wav', 'uncompr_wav']
else:
self.config['classes'] = self.get_classes_from_dataset(ds_config)
# Input shape and filter settings:
if self.config['calculate_mel']:
# Mel filter init:
self._mel_filter = filters.mel(self.config['sr'],
self.config['n_fft'],
n_mels=dlnet_config['n_mels'],
norm='slaney')
self.config['input_shape'] = (self.config['n_mels'],
self.config['n_frames'], 1)
elif self.config['filter_signal']:
# Crop spectrogram
# frequency array
self._freqs = np.fft.rfftfreq(self.config['n_fft'],
d=1 / self.config['sr'])
# cutoff frequency bin at cutoff frequency
self._cutoff_bin = int(
np.argmin(np.abs(self._freqs -
self.config['filter_config'][1])))
self.config['input_shape'] = (
int(len(self._freqs) - self._cutoff_bin),
self.config['n_frames'], 1)
else:
self.config['input_shape'] = (int(self.config['n_fft'] / 2 + 1),
self.config['n_frames'], 1)
def __init__(self,
flows,
n_group,
sr,
window_size,
n_mels,
hp,
use_conv1x1=False):
super().__init__()
self.flows = flows
self.n_group = n_group
self.win_size = window_size
self.hop_size = hp.audio.hop_length
self.n_mels = n_mels
self.sr = sr
self.sub_sr = self.hop_size // n_group
self.upsampler = nn.Sequential(
nn.ConvTranspose1d(n_mels,
n_mels,
self.sub_sr * 2 + 1,
self.sub_sr,
padding=self.sub_sr), nn.LeakyReLU(0.4, True))
self.upsampler.apply(add_weight_norms)
self.WNs = nn.ModuleList()
if use_conv1x1:
self.invconv1x1 = nn.ModuleList()
# Set up layers with the right sizes based on how many dimensions
# have been output already
for k in range(flows):
self.WNs.append(
WN2D(n_group, n_mels, hp.model.dilation_channels,
hp.model.residual_channels, hp.model.skip_channels))
if use_conv1x1:
self.invconv1x1.append(
InvertibleConv1x1(n_group, memory_efficient=False))
filters = mel(sr, window_size, n_mels, fmax=8000)
self.filter_idx = np.nonzero(filters)
self.register_buffer('filter_value',
torch.Tensor(filters[self.filter_idx]))
self.filter_size = torch.Size(filters.shape)
self.register_buffer('window', torch.hann_window(window_size))
def __init__(self, flows, n_group, n_early_every, n_early_size, sr,
window_size, hop_size, n_mels, memory_efficient, **kwargs):
super().__init__()
self.flows = flows
self.n_group = n_group
self.n_early_every = n_early_every
self.n_early_size = n_early_size
self.win_size = window_size
self.hop_size = hop_size
self.n_mels = n_mels
self.sr = sr
self.upsample_factor = hop_size // n_group
sub_win_size = window_size // n_group
# self.upsampler = nn.ConvTranspose1d(n_mels, n_mels, sub_win_size, self.upsample_factor,
# padding=sub_win_size // 2, bias=False)
self.invconv1x1 = nn.ModuleList()
self.WNs = nn.ModuleList()
# Set up layers with the right sizes based on how many dimensions
# have been output already
n_remaining_channels = n_group
self.z_split_sizes = []
for k in range(flows):
if k % self.n_early_every == 0 and k:
n_remaining_channels -= n_early_size
self.z_split_sizes.append(n_early_size)
self.invconv1x1.append(
InvertibleConv1x1(n_remaining_channels,
memory_efficient=memory_efficient))
self.WNs.append(
AffineCouplingBlock(WN,
memory_efficient=memory_efficient,
in_channels=n_remaining_channels // 2,
aux_channels=n_mels,
**kwargs))
self.z_split_sizes.append(n_remaining_channels)
filters = mel(sr, window_size, n_mels, fmax=8000)
self.filter_idx = np.nonzero(filters)
self.register_buffer('filter_value',
torch.Tensor(filters[self.filter_idx]))
self.filter_size = torch.Size(filters.shape)
self.register_buffer('window', torch.hann_window(window_size))
def __init__(self,
sr=22050,
n_fft=2048,
n_mels=128,
hop_length=512,
window='hann',
center=True,
pad_mode='reflect',
htk=False,
fmin=0.0,
fmax=None,
norm=1,
trainable_mel=False,
trainable_STFT=False):
super(MelSpectrogram, self).__init__()
self.stride = hop_length
self.center = center
self.pad_mode = pad_mode
self.n_fft = n_fft
# Create filter windows for stft
start = time()
wsin, wcos, self.bins2freq, _ = create_fourier_kernels(n_fft,
freq_bins=None,
window=window,
freq_scale='no',
sr=sr)
self.wsin = torch.tensor(wsin, dtype=torch.float)
self.wcos = torch.tensor(wcos, dtype=torch.float)
print("STFT filter created, time used = {:.4f} seconds".format(time() -
start))
# Creating kenral for mel spectrogram
start = time()
mel_basis = mel(sr, n_fft, n_mels, fmin, fmax, htk=htk, norm=norm)
self.mel_basis = torch.tensor(mel_basis)
print("Mel filter created, time used = {:.4f} seconds".format(time() -
start))
if trainable_mel == True:
self.mel_basis = torch.nn.Parameter(self.mel_basis)
if trainable_STFT == True:
self.wsin = torch.nn.Parameter(self.wsin)
self.wcos = torch.nn.Parameter(self.wcos)
class AudioProcessor:
"""Process audio data."""
sample_rate = 16000
top_db = 15
ref_db = 20
max_db = 100
fft_len = 1024
hop_len = 256
mel_basis = mel(sample_rate, fft_len, fmin=90, fmax=7600, n_mels=80).T
min_level = np.exp(-100 / 20 * np.log(10))
@classmethod
def butter_highpass(cls, cutoff=30, order=5):
"""Create butter highpass filter."""
normal_cutoff = cutoff / (0.5 * cls.sample_rate)
return butter(order, normal_cutoff, btype='high', analog=False)
@classmethod
def short_time_fourier_transform(cls, wav):
"""Apply short time Fourier transform."""
d_matrix = stft(wav, n_fft=cls.fft_len, hop_length=cls.hop_len)
return np.abs(d_matrix)
@classmethod
def file2spectrogram(cls, file_path):
"""Load audio file and create spectrogram."""
wav = load(file_path, sr=cls.sample_rate)[0]
wav = trim(wav, top_db=cls.top_db)[0]
wav = filtfilt(*cls.butter_highpass(), wav)
wav = wav * 0.96
d_mag = cls.short_time_fourier_transform(wav)
d_mel = np.dot(d_mag.T, cls.mel_basis)
db_val = 20 * np.log10(np.maximum(cls.min_level, d_mel))
db_scaled = db_val - cls.ref_db
db_normalized = (db_scaled + cls.max_db) / cls.max_db
return np.clip(db_normalized, 0, 1).astype(np.float32)
def transform2mel(spectrogram,samplerate,fft_window_size,n_mel_bands = 80,freq_min = 0,freq_max = None):
'''Transform to Mel
convert a spectrogram to a Mel scale spectrogram by grouping original frequency bins
to Mel frequency bands (using Mel filter from Librosa)
Parameters
spectrogram: input spectrogram
samplerate: samplerate of audio signal
fft_window_size: number of time window / frequency bins in the FFT analysis
n_mel_bands: number of desired Mel bands, typically 20, 40, 80 (max. 128 which is default when 'None' is provided)
freq_min: minimum frequency (Mel filters will be applied >= this frequency, but still return n_meld_bands number of bands)
freq_max: cut-off frequency (Mel filters will be applied <= this frequency, but still return n_meld_bands number of bands)
Returns:
mel_spectrogram: Mel spectrogram: np.array of shape(n_mel_bands,frames) maintaining the number of frames in the original spectrogram
'''
from librosa.filters import mel
# Syntax: librosa.filters.mel(sr, n_fft, n_mels=128, fmin=0.0, fmax=None, htk=False)
mel_basis = mel(samplerate,fft_window_size, n_mels=n_mel_bands,fmin=freq_min,fmax=freq_max)
freq_bin_max = mel_basis.shape[1] # will be fft_window_size / 2 + 1
# IMPLEMENTATION WITH FOR LOOP
# initialize Mel Spectrogram matrix
#n_mel_bands = mel_basis.shape[0] # get the number of bands from result in case 'None' was specified as parameter
#mel_spectrogram = np.empty((n_mel_bands, frames))
#for i in range(frames): # stepping through the wave segment, building spectrum for each window
# mel_spectrogram[:,i] = np.dot(mel_basis,spectrogram[0:freq_bin_max,i])
# IMPLEMENTATION WITH DOT PRODUCT (15% faster)
# multiply the mel filter of each band with the spectogram frame (dot product executes it on all frames)
# filter will be adapted in a way so that frequencies beyond freq_max will be discarded
mel_spectrogram = np.dot(mel_basis,spectrogram[0:freq_bin_max,:])
return (mel_spectrogram)
