def mashability(song1, song2):
"""
Returns how well song1 transitions into song2 using cosine matrix similarity
and FFT semitone bin approximation matrices
"""
# If the tempo differs by more than thirty then we should never make that transition
if abs(song1.bpm - song2.bpm) > 30:
return 1
sample_length = MIX_LENGTH # beats per sample
beats1 = song1.AudioFile.analysis.beats[song1.mix_out : song1.mix_out + sample_length]
beats2 = song2.AudioFile.analysis.beats[song1.mix_in : song1.mix_in + sample_length]
data1 = audio.getpieces(song1.AudioFile, beats1)
data2 = audio.getpieces(song2.AudioFile, beats2)
data1.encode("temp1.mp3")
data2.encode("temp2.mp3")
y1, sr1 = librosa.load("temp1.mp3")
y2, sr2 = librosa.load("temp2.mp3")
S1 = np.abs(librosa.stft(y1, n_fft=4096))
chroma1 = librosa.feature.chroma_stft(S=S1, sr=sr1)
S2 = np.abs(librosa.stft(y2, n_fft=4096))
chroma2 = librosa.feature.chroma_stft(S=S2, sr=sr2)
# im = librosa.display.specshow(chroma1,x_axis = "time",y_axis = "chroma")
# im2 = librosa.display.specshow(chroma2,x_axis = "time",y_axis = "chroma")
# plt.show()
orthogonal_arr = []
for i in range(min(chroma1.shape[1], chroma2.shape[1])):
orthogonal_arr.append(dst.cosine(chroma1[:, i], chroma2[:, i]))
return sum(orthogonal_arr) / len(orthogonal_arr)
def reverse_channel(a, b, n_fft=2**13, win_length=2**12, hop_length=2**10):
'''
Estimates the channel distortion in b relative to a and reverses it
:parameters:
- a : np.ndarray
Some signal
- b : np.ndarray
Some other signal with channel distortion relative to a
- n_fft : int
Number of samples in each FFT computation, default 2**13
- win_length : int
Number of samples in each window, default 2**12
- hop_length : int
Number of samples between successive FFT computations, default 2**10
:returns:
- b_filtered : np.ndarray
The signal b, filtered to reduce channel distortion
'''
# Compute spectrograms
a_spec = librosa.stft(a, n_fft=n_fft, win_length=win_length, hop_length=hop_length)
b_spec = librosa.stft(b, n_fft=n_fft, win_length=win_length, hop_length=hop_length)
# Compute the best filter
H = best_filter_coefficients(a_spec, b_spec)
# Apply it in the frequency domain (ignoring aliasing! Yikes)
b_spec_filtered = H*b_spec
# Get back to time domain
b_filtered = librosa.istft(b_spec_filtered, win_length=win_length, hop_length=hop_length)
return b_filtered
def test_stft_bad_window():
y = np.zeros(22050 * 5)
n_fft = 2048
window = np.ones(n_fft // 2)
librosa.stft(y, n_fft=n_fft, window=window)
def getSpectra(self, n_fft = 4096, hop_length = 1024):
if self.spectra == None:
stft = librosa.stft(self.signal, n_fft, hop_length)
self.spectra = Spectra(stft, self.sr, n_fft, hop_length)
elif self.spectra.n_fft != n_fft or self.spectra.hop_length != hop_length:
stft = librosa.stft(self.signal, n_fft, hop_length)
return Spectra(stft, self.sr, n_fft, hop_length)
return self.spectra
def hpss(y):
D = librosa.stft(y)
H, P = librosa.decompose.hpss(D, kernel_size=KERNEL_SIZE, power=HPSS_P)
D_harm = np.abs(librosa.stft(librosa.istft(H), n_fft=N_FFT, hop_length=HOP))
D_perc = np.abs(librosa.stft(librosa.istft(P), n_fft=N_FFT, hop_length=HOP))
return D_harm, D_perc
def __test_stft(center, pad_mode):
D1 = librosa.stft(y, center=center, pad_mode='reflect')
D2 = librosa.stft(y, center=center, pad_mode=pad_mode)
assert D1.shape == D2.shape
if center and pad_mode != 'reflect':
assert not np.allclose(D1, D2)
else:
assert np.allclose(D1, D2)
def SaveSpectrogram(y_mix, y_inst,y_vocal, filename, orig_sr=44100) :
y_mix = librosa.core.resample(y_mix,orig_sr,SR)
y_vocal = librosa.core.resample(y_vocal,orig_sr,SR)
y_inst = librosa.core.resample(y_inst,orig_sr,SR)
S_mix = np.abs(librosa.stft(y_mix,n_fft=window_size,hop_length=hop_length)).astype(np.float32)
S_inst = np.abs(librosa.stft(y_inst,n_fft=window_size,hop_length=hop_length)).astype(np.float32)
S_vocal = np.abs(librosa.stft(y_vocal,n_fft=window_size,hop_length=hop_length)).astype(np.float32)
norm = S_mix.max()
S_mix /= norm
S_inst /= norm
S_vocal /= norm
np.savez(os.path.join('./Spectrogram',filename+'.npz'),mix=S_mix,inst=S_inst ,vocal=S_vocal)
def wavs_to_specs(wavs_mono, wavs_src1, wavs_src2, n_fft = 1024, hop_length = None):
stfts_mono = list()
stfts_src1 = list()
stfts_src2 = list()
for wav_mono, wav_src1, wav_src2 in zip(wavs_mono, wavs_src1, wavs_src2):
stft_mono = librosa.stft(wav_mono, n_fft = n_fft, hop_length = hop_length)
stft_src1 = librosa.stft(wav_src1, n_fft = n_fft, hop_length = hop_length)
stft_src2 = librosa.stft(wav_src2, n_fft = n_fft, hop_length = hop_length)
stfts_mono.append(stft_mono)
stfts_src1.append(stft_src1)
stfts_src2.append(stft_src2)
return stfts_mono, stfts_src1, stfts_src2
def midi_to_cqt(midi, sf2_path=None, fs=22050, hop=512):
'''
Feature extraction routine for midi data, converts to a drum-free, percussion-suppressed CQT.
Input:
midi - pretty_midi.PrettyMIDI object
sf2_path - path to .sf2 file to pass to pretty_midi.fluidsynth
fs - sampling rate to synthesize audio at, default 22050
hop - hop length for cqt, default 512
Output:
midi_gram - Simulated CQT of the midi data
'''
# Synthesize the MIDI using the supplied sf2 path
midi_audio = midi.fluidsynth(fs=fs, sf2_path=sf2_path)
# Use the harmonic part of the signal
H, P = librosa.decompose.hpss(librosa.stft(midi_audio))
midi_audio_harmonic = librosa.istft(H)
# Compute log frequency spectrogram of audio synthesized from MIDI
midi_gram = np.abs(librosa.cqt(y=midi_audio_harmonic,
sr=fs,
hop_length=hop,
fmin=librosa.midi_to_hz(36),
n_bins=60,
tuning=0.0))**2
return midi_gram
def amplitude_for_file(audio_path):
y, sr = librosa.load(audio_path)
# from http://bmcfee.github.io/librosa/librosa.html#librosa.core.logamplitude
# Get a power spectrogram from a waveform y
S = np.abs(librosa.stft(y)) ** 2
log_S = librosa.logamplitude(S)
return log_S
def audio_to_cqt_and_onset_strength(audio, fs=22050, hop=512):
'''
Feature extraction for audio data.
Gets a power CQT of harmonic component and onset strength signal of percussive.
Input:
midi - pretty_midi.PrettyMIDI object
fs - sampling rate to synthesize audio at, default 22050
hop - hop length for cqt, default 512, onset strength hop will be 1/4 of this
Output:
audio_gram - CQT of audio data
audio_onset_strength - onset strength signal
'''
# Use harmonic part for gram, percussive part for onsets
H, P = librosa.decompose.hpss(librosa.stft(audio))
audio_harmonic = librosa.istft(H)
audio_percussive = librosa.istft(P)
# Compute log-frequency spectrogram of original audio
audio_gram = np.abs(librosa.cqt(y=audio_harmonic,
sr=fs,
hop_length=hop,
fmin=librosa.midi_to_hz(36),
n_bins=60))**2
# Beat track the audio file at 4x the hop rate
audio_onset_strength = librosa.onset.onset_strength(audio_percussive, hop_length=hop/4, sr=fs)
return audio_gram, audio_onset_strength
def test_piptrack():
def __test(S, freq):
pitches, mags = librosa.piptrack(S=S, fmin=100)
idx = (mags > 0)
assert len(idx) > 0
recovered_pitches = pitches[idx]
# We should be within one cent of the target
assert np.all(np.abs(np.log2(recovered_pitches) - np.log2(freq)) <= 1e-2)
sr = 22050
duration = 3.0
for freq in [110, 220, 440, 880]:
# Generate a sine tone
y = np.sin(2 * np.pi * freq * np.linspace(0, duration, num=duration*sr))
for n_fft in [1024, 2048, 4096]:
# Using left-aligned frames eliminates reflection artifacts at the boundaries
S = np.abs(librosa.stft(y, n_fft=n_fft, center=False))
yield __test, S, freq
def parse_audio(path, audio_conf, windows, normalize=False):
'''
Input:
path : string 导入音频的路径
audio_conf : dict 求频谱的音频参数
windows : dict 加窗类型
Output:
spect : FloatTensor 每帧的频谱
'''
y = load_audio(path)
n_fft = int(audio_conf['sample_rate']*audio_conf["window_size"])
win_length = n_fft
hop_length = int(audio_conf['sample_rate']*audio_conf['window_stride'])
window = windows[audio_conf['window']]
D = librosa.stft(y, n_fft=n_fft, hop_length=hop_length,
win_length=win_length, window=window)
spect, phase = librosa.magphase(D)
spect = torch.FloatTensor(spect)
spect = spect.log1p()
if normalize:
mean = spect.mean()
std = spect.std()
spect.add_(-mean)
spect.div_(std)
return spect.transpose(0,1)
def parse_audio(self, audio_path):
if self.augment:
y = load_randomly_augmented_audio(audio_path, self.sample_rate)
else:
y = load_audio(audio_path)
if self.noiseInjector:
add_noise = np.random.binomial(1, self.noise_prob)
if add_noise:
y = self.noiseInjector.inject_noise(y)
n_fft = int(self.sample_rate * self.window_size)
win_length = n_fft
hop_length = int(self.sample_rate * self.window_stride)
# STFT
D = librosa.stft(y, n_fft=n_fft, hop_length=hop_length,
win_length=win_length, window=self.window)
spect, phase = librosa.magphase(D)
# S = log(S+1)
spect = np.log1p(spect)
spect = torch.FloatTensor(spect)
if self.normalize:
mean = spect.mean()
std = spect.std()
spect.add_(-mean)
spect.div_(std)
return spect
def test_piptrack_properties():
def __test(S, n_fft, hop_length, fmin, fmax, threshold):
pitches, mags = librosa.core.piptrack(S=S,
n_fft=n_fft,
hop_length=hop_length,
fmin=fmin,
fmax=fmax,
threshold=threshold)
# Shape tests
eq_(S.shape, pitches.shape)
eq_(S.shape, mags.shape)
# Make sure all magnitudes are positive
assert np.all(mags >= 0)
# Check the frequency estimates for bins with non-zero magnitude
idx = (mags > 0)
assert np.all(pitches[idx] >= fmin)
assert np.all(pitches[idx] <= fmax)
# And everywhere else, pitch should be 0
assert np.all(pitches[~idx] == 0)
y, sr = librosa.load('data/test1_22050.wav')
for n_fft in [2048, 4096]:
for hop_length in [None, n_fft // 4, n_fft // 2]:
S = np.abs(librosa.stft(y, n_fft=n_fft, hop_length=hop_length))
for fmin in [0, 100]:
for fmax in [4000, 8000, sr // 2]:
for threshold in [0.1, 0.2, 0.5]:
yield __test, S, n_fft, hop_length, fmin, fmax, threshold
def stretch_demo(input_file, output_file, speed):
'''Phase-vocoder time stretch demo function.
:parameters:
- input_file : str
path to input audio
- output_file : str
path to save output (wav)
- speed : float > 0
speed up by this factor
'''
N_FFT = 2048
HOP_LENGTH = N_FFT /4
# 1. Load the wav file, resample
print 'Loading ', input_file
y, sr = librosa.load(input_file)
# 2. generate STFT @ 2048 samples
print 'Computing short-time fourier transform... '
D = librosa.stft(y, n_fft=N_FFT, hop_length=HOP_LENGTH)
print 'Playing back at %3.f%% speed' % (speed * 100)
D_stretch = librosa.phase_vocoder(D, speed, hop_length=HOP_LENGTH)
y_stretch = librosa.istft(D_stretch, hop_length=HOP_LENGTH)
print 'Saving stretched audio to: ', output_file
librosa.output.write_wav(output_file, y_stretch, sr)
def compute_beat_mls(filename, beat_times, mel_bands=num_mel_bands, fft_size=1024, hop_size=512):
"""
Compute average Mel log spectrogram per beat given previously
extracted beat times.
:param filename: path to audio file
:param beat_times: list of beat times in seconds
:param mel_bands: number of Mel bands
:param fft_size: FFT size
:param hop_size: hop size for FFT processing
:return: beat Mel spectrogram (mel_bands x frames)
"""
y, sr = librosa.load(filename, sr=22050, mono=True)
spec = np.abs(librosa.stft(y=y, n_fft=fft_size, hop_length=hop_size, win_length=fft_size,
window=scipy.signal.hamming))
mel_fb = librosa.filters.mel(sr=22050, n_fft=fft_size, n_mels=mel_bands, fmin=50, fmax=10000, htk=True)
s = np.sum(mel_fb, axis=1)
mel_fb = np.divide(mel_fb, s[:, np.newaxis])
mel_spec = np.dot(mel_fb, spec)
mel_spec = np.log10(1. + 1000. * mel_spec)
beat_frames = np.round(beat_times * (22050. / hop_size))
beat_melspec = np.max(mel_spec[:, beat_frames[0]:beat_frames[1]], axis=1)
for k in xrange(1, beat_frames.shape[0]-1):
beat_melspec = np.column_stack((beat_melspec,
np.max(mel_spec[:, beat_frames[k]:beat_frames[k+1]], axis=1)))
return beat_melspec
def __test_consistency(frame_length, hop_length, center):
y, sr = librosa.load(__EXAMPLE_FILE, sr=None)
# Ensure audio is divisible into frame size.
y = librosa.util.fix_length(y, y.size - y.size % frame_length)
assert y.size % frame_length == 0
# STFT magnitudes with a constant windowing function and no centering.
S = librosa.magphase(librosa.stft(y,
n_fft=frame_length,
hop_length=hop_length,
window=np.ones,
center=center))[0]
# Try both RMS methods.
rms1 = librosa.feature.rms(S=S, frame_length=frame_length,
hop_length=hop_length)
rms2 = librosa.feature.rms(y=y, frame_length=frame_length,
hop_length=hop_length, center=center)
assert rms1.shape == rms2.shape
# Normalize envelopes.
rms1 /= rms1.max()
rms2 /= rms2.max()
# Ensure results are similar.
np.testing.assert_allclose(rms1, rms2, rtol=5e-2)
def get_spectrograms(sound_file):
'''Extracts melspectrogram and log magnitude from given `sound_file`.
Args:
sound_file: A string. Full path of a sound file.
Returns:
Transposed S: A 2d array. A transposed melspectrogram with shape of (T, n_mels)
Transposed magnitude: A 2d array.Has shape of (T, 1+hp.n_fft//2)
'''
# Loading sound file
y, sr = librosa.load(sound_file, sr=hp.sr) # or set sr to hp.sr.
# stft. D: (1+n_fft//2, T)
D = librosa.stft(y=y,
n_fft=hp.n_fft,
hop_length=hp.hop_length,
win_length=hp.win_length)
# magnitude spectrogram
magnitude = np.abs(D) #(1+n_fft/2, T)
# power spectrogram
power = magnitude**2 #(1+n_fft/2, T)
# mel spectrogram
S = librosa.feature.melspectrogram(S=power, n_mels=hp.n_mels) #(n_mels, T)
return np.transpose(S.astype(np.float32)), np.transpose(magnitude.astype(np.float32)) # (T, n_mels), (T, 1+n_fft/2)
def getChromagram(self, n_fft = 4096, hop_length = 1024):
if self.chromagram == None:
if self.spectra == None:
stft = librosa.stft(self.signal, n_fft, hop_length)
self.spectra = Spectra(stft, self.sr, n_fft, hop_length)
self.chromagram = librosa.feature.chromagram(S=self.spectra.getMagnitude())
return self.chromagram
def mfcc_clustering(file_name, n_clusters):
"""
From Prem
:return:
"""
clusterer = KMeans(n_clusters=n_clusters)
print(file_name)
mix, sr = librosa.load(file_name)
mix_stft = librosa.stft(mix)
comps, acts = find_template(mix_stft, sr, 100, 101, 0, mix_stft.shape[1])
cluster_comps = librosa.feature.mfcc(S=comps)[1:14]
save_mfcc_img(file_name[:-4] + "_mfcc.png", np.flipud(cluster_comps))
clusterer.fit_transform(cluster_comps.T)
labels = clusterer.labels_
# print(labels)
sources = []
for cluster_index in range(n_clusters):
indices = np.where(labels == cluster_index)[0]
template, residual = extract_template(comps[:, indices], mix_stft)
t = librosa.istft(template)
sources.append(t)
return np.array(sources)
def find_peaks(self, n_fft=2048, tau=10, kappa=10):
''' Extracts a fingerprint from the loaded audio data. Stores value
internally as a integer hash code.
'''
# compute stft
if self._audio_data is None:
raise ValueError("No audio data loaded.")
S = librosa.stft(self._audio_data, n_fft)
# find peaks
n_frames = S.shape[0]
n_bins = S.shape[1]
peaks = []
for n in range(n_frames):
for k in range(n_bins):
p = S[n][k]
is_peak = True
# search neigborhood
for i in range(tau):
for j in range(kappa):
n_ = n + i - tau//2
k_ = k + j - kappa//2
if 0 <= n_ < n_frames and 0 < k_ < n_bins and (n != n_ and k != n_):
p_ = S[n_][k_]
if abs(p_) < abs(p):
is_peak = False
if is_peak:
peaks.append((n, k))
return peaks
def percussive(y):
'''Extract the percussive component of an audio time series'''
D = librosa.stft(y)
P = librosa.decompose.hpss(D)[1]
return librosa.istft(P)
def extract_features(audio_file: Path,
seconds: int = params.nsynth_max_seconds,
window_size: int = params.librosa_spec_windows,
hop_length: int = params.librosa_hop_length,
calc_chroma_stft=True,
calc_mfcc_stft=True,
calc_mfcc=True):
audio_data, sr = librosa.load(audio_file, sr=None)
if seconds:
audio_data = audio_data[:seconds * sr]
stft = np.abs(librosa.stft(
audio_data,
hop_length=hop_length)) if calc_chroma_stft or calc_mfcc_stft else None
chroma_stft = librosa.feature.chroma_stft(
S=stft, sr=sr, n_chroma=window_size,
hop_length=hop_length) if calc_chroma_stft else None
mfcc_stft = librosa.feature.mfcc(
audio_data, S=stft, sr=sr, n_mfcc=window_size,
hop_length=hop_length) if calc_mfcc_stft else None
mfcc = librosa.feature.mfcc(
audio_data, sr=sr, n_mfcc=window_size,
hop_length=hop_length) if calc_mfcc else None
return (chroma_stft, mfcc_stft, mfcc)
def makeSpectragrams(filename):
f, sr = librosa.load(filename)
print "first"
melSpectra = librosa.feature.melspectrogram(f)
cqtSpectra = librosa.cqt(f)
stftSpectra = librosa.stft(f)
print "stuff"
librosa.display.specshow(melSpectra)
# plt.specgram(melSpectra)
imageName = filename, "MelSpectragram.png"
title = "Mel Spectrogram \nof " + filename[26:]
plt.title(title)
plt.ion()
# plt.savefig(imageName)
plt.show()
librosa.display.specshow(cqtSpectra)
title = "Constant Q Spectrogram \nof " + filename[26:]
plt.title(title)
# plt.spectrogram(cqtSpectra)
plt.show()
librosa.display.specshow(stftSpectra)
title = "STFT Spectrogram \nof " + filename[26:]
plt.title(title)
# plt.spectrogram(cqtSpectra)
plt.show()
return True
def get_feature(fname):
#b,_ = librosa.load(fname, res_type = 'kaiser_fast')
b,_ = librosa.load(fname, res_type = 'kaiser_fast')
try:
mfcc = np.mean(librosa.feature.mfcc(y = b,n_mfcc=60).T,axis=0)
mels = np.mean(librosa.feature.melspectrogram(b, sr = SAMPLE_RATE).T,axis = 0)
stft = np.abs(librosa.stft(b))
chroma = np.mean(librosa.feature.chroma_stft(S=stft, sr = SAMPLE_RATE).T,axis = 0)
contrast=np.mean(librosa.feature.spectral_contrast(S=stft, sr=SAMPLE_RATE).T,axis=0)
tonnetz=np.mean(librosa.feature.tonnetz(librosa.effects.harmonic(b), sr = SAMPLE_RATE).T,axis = 0)
ft2 = librosa.feature.zero_crossing_rate(b)[0]
ft3 = librosa.feature.spectral_rolloff(b)[0]
ft4 = librosa.feature.spectral_centroid(b)[0]
ft5 = librosa.feature.spectral_contrast(b)[0]
ft6 = librosa.feature.spectral_bandwidth(b)[0]
ft2_trunc = np.hstack([np.mean(ft2),np.std(ft2), skew(ft2), np.max(ft2), np.min(ft2)])
ft3_trunc = np.hstack([np.mean(ft3),np.std(ft3), skew(ft3), np.max(ft3), np.min(ft3)])
ft4_trunc = np.hstack([np.mean(ft4),np.std(ft4), skew(ft4), np.max(ft4), np.min(ft4)])
ft5_trunc = np.hstack([np.mean(ft5),np.std(ft5), skew(ft5), np.max(ft5), np.min(ft5)])
ft6_trunc = np.hstack([np.mean(ft6),np.std(ft6), skew(ft6), np.max(ft6), np.min(ft6)])
return pd.Series(np.hstack((mfcc,mels,chroma,contrast,tonnetz,ft2_trunc, ft3_trunc, ft4_trunc, ft5_trunc, ft6_trunc)))
#d = np.hstack([mfcc,mels,chroma,contrast,tonnetz,ft2_trunc,ft3_trunc,ft4_trunc,ft5_trunc,ft6_trunc])
#features = np.empty((0,238))
#d = np.vstack([features,d])
except:
print('bad file')
return pd.Series([0]*238)
def parse_audio(path, audio_conf, windows, normalize=True):
'''
Input:
path : string 导入音频的路径
audio_conf : dcit 求频谱的音频参数
windows : dict 加窗类型
Output:
spect : ndarray 每帧的频谱
'''
y = load_audio(path)
n_fft = int(audio_conf['sample_rate']*audio_conf["window_size"])
win_length = n_fft
hop_length = int(audio_conf['sample_rate']*audio_conf['window_stride'])
window = windows[audio_conf['window']]
D = librosa.stft(y, n_fft=n_fft, hop_length=hop_length,
win_length=win_length, window=window)
spect, phase = librosa.magphase(D)
spect = np.log1p(spect)
if normalize:
mean = spect.mean()
std = spect.std()
spect = np.add(spect, -mean)
spect = np.divide(spect, std)
return spect.transpose()
def melspectrogram(y, sr=22050, n_fft=256, hop_length=128, **kwargs):
"""Compute a mel spectrogram from a time series
Arguments:
y -- (ndarray) audio time-series
sr -- (int) sampling rate of y | default: 22050
n_fft -- (int) number of FFT components | default: 256
hop_length -- (int) frames to hop | default: 128
**kwargs -- Mel filterbank parameters
See melfb() documentation for details.
Returns S:
S -- (ndarray) Mel spectrogram
"""
# Compute the STFT
powspec = np.abs(librosa.stft(y,
n_fft = n_fft,
hann_w = n_fft,
hop_length = hop_length))**2
# Build a Mel filter
mel_basis = melfb(sr, n_fft, **kwargs)
# Remove everything past the nyquist frequency
mel_basis = mel_basis[:, :(n_fft/ 2 + 1)]
return np.dot(mel_basis, powspec)
def compute_cqt(filename):
a, sr = librosa.load(filename, sr=SR)
spectrum = librosa.stft(a)
harm_spec, _ = librosa.decompose.hpss(spectrum)
harm = librosa.istft(harm_spec)
cqt = np.abs(librosa.cqt(harm, sr=sr, hop_length=HOP, real=False))
return cqt
def test_real_hpss():
# Load an audio signal
y, sr = librosa.load('data/test1_22050.wav')
D = np.abs(librosa.stft(y))
def __hpss_test(window, power, mask, margin):
H, P = librosa.decompose.hpss(D, kernel_size=window, power=power, mask=mask, margin=margin)
if margin == 1.0 or margin == (1.0, 1.0):
if mask:
assert np.allclose(H + P, np.ones_like(D))
else:
assert np.allclose(H + P, D)
else:
if mask:
assert not np.any(H.astype(bool) & P.astype(bool))
else:
assert np.all(H + P <= D)
for window in [31, (5, 5)]:
for power in [1, 2, 10]:
for mask in [False, True]:
for margin in [1.0, 3.0, (1.0, 1.0), (9.0, 10.0)]:
yield __hpss_test, window, power, mask, margin
def local_audio_3d(self, filepath, mode="waveform"):
""" Converts a local audio file into a 3D model.
Args:
filepath: string containing the path to the audio file
mode: musical parameter to be used to create the 3D model
options: waveform, stft
"""
print "Loading audio file " + filepath
waveform, sr = librosa.load(filepath)
if mode == "waveform": # time domain analysis
# Downsample waveform and store positive values only
if len(waveform) > 1000:
m = len(str(len(waveform)))
downsample_factor = (10 ** (m-1-3) * int(str(len(waveform))[0])) # 1k (i.e. 10^3) magnitude
else:
downsample_factor = 1
half_waveform = [waveform[i] for i in xrange(len(waveform)) if waveform[i]>0 and i%downsample_factor==0]
# Reshape and rescale waveform
processed_waveform = self._movingaverage(half_waveform, self.ma_window_size)
# processed_waveform = self._limit_spikes(half_waveform, np.mean(half_waveform), 5)
processed_waveform = self._rescale_list(processed_waveform, 0, self.height_Y)
processed_waveform = self.make_waveform_square(processed_waveform, self.n_waveform_bars) # make waveform "square"
# Convert 2D waveform into 3D
print "Creating 3D model"
model_3d = self.make_waveform_3d(processed_waveform, self.height_Z)
else: # frequency domain analysis
self.mask_val /= 100
# Get STFT magnitude
print "Analyzing frequency components"
stft = librosa.stft(waveform, n_fft=256)
stft, phase = librosa.magphase(stft)
# Downsample and rescale STFT
if len(stft[0]) > 1000:
m = len(str(len(stft[0])))
downsample_factor = (10 ** (m-1-3)) * int(str(len(stft[0]))[0]) # 1k (i.e. 10^3) magnitude
else:
downsample_factor = 1
new_stft = []
for curr_fft in stft:
min_loudness_value = max(curr_fft)
ds_fft = [curr_fft[j] + min_loudness_value for j in xrange(len(curr_fft)) if j%downsample_factor==0]
ds_fft = self._rescale_list(ds_fft, self.min_absolute_value, self.height_Z)
new_stft.append(ds_fft)
print "Creating 3D model"
model_3d = np.array(new_stft)
print "Exporting the 3D file"
if self.OUTPUT_FOLDER[-1] != '/':
self.OUTPUT_FOLDER.append('/')
output_filename = self.OUTPUT_FOLDER + filepath.split('/')[-1][:-4] + "_" + mode + ".stl"
numpy2stl(model_3d,
output_filename,
scale=self.scale,
mask_val=self.mask_val,
solid=True)
def plot_signal(idx, data):
if len(idx) == 0:
raise PreventUpdate
figs = make_subplots(rows=2,
cols=1,
subplot_titles=('Waveform', 'Spectrogram'))
try:
filename = data[idx[0]]['audio_filepath']
audio, fs = librosa.load(filename, sr=None)
time_stride = 0.01
hop_length = int(fs * time_stride)
n_fft = 512
# linear scale spectrogram
s = librosa.stft(y=audio, n_fft=n_fft, hop_length=hop_length)
s_db = librosa.power_to_db(np.abs(s)**2, ref=np.max, top_db=100)
figs.add_trace(
go.Scatter(
x=np.arange(audio.shape[0]) / fs,
y=audio,
line={'color': 'green'},
name='Waveform',
hovertemplate=
'Time: %{x:.2f} s<br>Amplitude: %{y:.2f}<br><extra></extra>',
),
row=1,
col=1,
)
figs.add_trace(
go.Heatmap(
z=s_db,
colorscale=[
[0, 'rgb(30,62,62)'],
[0.5, 'rgb(30,128,128)'],
[1, 'rgb(30,255,30)'],
],
colorbar=dict(yanchor='middle',
lenmode='fraction',
y=0.2,
len=0.5,
ticksuffix=' dB'),
dx=time_stride,
dy=fs / n_fft / 1000,
name='Spectrogram',
hovertemplate=
'Time: %{x:.2f} s<br>Frequency: %{y:.2f} kHz<br>Magnitude: %{z:.2f} dB<extra></extra>',
),
row=2,
col=1,
)
figs.update_layout({
'margin': dict(l=0, r=0, t=20, b=0, pad=0),
'height': 500
})
figs.update_xaxes(title_text='Time, s', row=1, col=1)
figs.update_yaxes(title_text='Amplitude', row=1, col=1)
figs.update_xaxes(title_text='Time, s', row=2, col=1)
figs.update_yaxes(title_text='Frequency, kHz', row=2, col=1)
except Exception:
pass
return figs
"""
Created on Fri Mar 8 08:41:08 2019
@author: MR toad
"""
import librosa
import os
import numpy as np
import librosa.display
import matplotlib.pyplot as plt
#from PIL import Image
file_path = 'E:/speech/'
mfcc_path = 'E:/mfcc/'
pic_path = 'E:/pic'
file_name_list = os.listdir(file_path)
for file_name in file_name_list:
y, sr = librosa.load(file_path + file_name)
mfcc_feature = librosa.feature.mfcc(y=y, sr=sr)
np.save(mfcc_path + file_name.split('.')[0] + ".npy", mfcc_feature)
plt.figure(figsize=(12, 8))
D = librosa.amplitude_to_db(librosa.stft(y), ref=np.max)
plt.subplot(4, 2, 1)
librosa.display.specshow(D, y_axis='linear')
plt.colorbar(format='%+2.0f dB')
plt.title('Linear-frequency power spectrogram')
plt.savefig(file_name.split('.')[0] + ".png", dpi=300)
def _stft(self, x):
return librosa.stft(x, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length)
def GenerateLibrosaFeatures(wav_path):
y, sr = librosa.load(wav_path)
stft = librosa.stft(y, n_fft=BUFFER_LENGTH, hop_length=HOP_LENGTH)
D = np.abs(stft)**2
S = np.log(librosa.feature.melspectrogram(S=D, n_mels=MEL_COUNT))
return S
for chromagram_i in range(12):
data[i,50+chromagram_i] = chromagram_mean[chromagram_i]
data[i,62+chromagram_i] = chromagram_var[chromagram_i]
#plt.figure(figsize=(16, 6))
#librosa.display.specshow(chromagram, x_axis='time', y_axis='chroma', hop_length=hop_length, cmap='coolwarm')
#----------------------
#Fourier Transform(傅里叶变换)
# Default FFT window size
n_fft = 2048 # FFT window size
hop_length = 512 # number audio of frames between STFT columns (looks like a good default)
# Short-time Fourier transform (STFT)
D = np.abs(librosa.stft(audio_file, n_fft = n_fft, hop_length = hop_length)) #振幅D=（f，t）
frequent_weights = D.sum(axis=1) #各频率权重时间和（行为频率列为时间）
frequent_list = librosa.fft_frequencies(sr=sr, n_fft = n_fft)/sr
#print(np.shape(frequent_list))
#print(np.shape(frequent_weight))
#print(frequent_list)
#print(frequent_weight)
chroma_stft_mean = np.average(frequent_list,weights=frequent_weights)
chroma_stft_var = sum(((frequent_list-chroma_stft_mean)*frequent_weights)**2)/sum(frequent_weights)
#print(chroma_stft_mean)
#print(chroma_stft_var)
data[i,74] = chroma_stft_mean
data[i,75] = chroma_stft_var
print(data)
def get_spec(wav, n_fft=1024, window="hamming", hop_length=256):
return librosa.stft(wav, window=window, n_fft=n_fft, hop_length=hop_length)
def stft_fn(y):
return librosa.stft(y=y,
n_fft=int(frame_size),
hop_length=hop_size,
center=False).T
def specgram_lbrs(audiopath: str, plotpath: str=None, name: str=None,
cmap: str='gray_r', algorithm='default', y_axis=None,
**kwargs):
"""
Generates a spectrogram of an audio file using librosa.display.specshow
function.
The output will be in png format.
:param audiopath: string
Path of the audio file.
:param plotpath: string
Path to plot the spectrogram. Default to the current working directory.
:param name: string
Name of the output image. Default to audio name.
:param cmap: string
Automatic colormap detection
See matplotlib.pyplot.pcolormesh.
:type algorithm: str or callable
:param algorithm: Algorithm to use to compute the spectrogram.
Available algorithms: 'default', 'mel', 'log'.
The 'default' mode uses the librosa.stft function to compute the
spectrogram data, the 'mel' uses the librosa.feature.melspectrogram
function and 'log' uses librosa.cqt function.
Expected return type of the algorithm: np.ndarray [shape=(Any, t)]
:param y_axis: None or str.
Range for the y-axes. This parameter is passed to the
librosa.display.specshow function.
:param kwargs:
Additional kwargs are passed on to the defined algorithm function.
"""
if plotpath is not None and not os.path.isdir(plotpath):
os.makedirs(plotpath)
if algorithm not in ['mel', 'log', 'default'] or not callable:
raise ValueError('Unrecognized scale or not a callable object.')
# Load audio and convert it to mono
y, sr = librosa.load(audiopath)
y = librosa.core.to_mono(y)
# Apply algorithm to obtain an array of spectrogram data
if algorithm == 'default':
spec_data = librosa.stft(y, **kwargs)
# Convert the data spectrogram to decibel units
spec_data = librosa.power_to_db(librosa.magphase(spec_data, power=2)[0],
ref=np.max)
elif algorithm == 'mel':
kwargs.setdefault('n_mels', 128)
kwargs.setdefault('fmax', 8000)
spec_data = librosa.feature.melspectrogram(y=y, sr=sr, **kwargs)
# Convert the data spectrogram to decibel units
spec_data = librosa.power_to_db(spec_data, ref=np.max)
elif algorithm == 'log':
spec_data = librosa.cqt(y, sr, **kwargs)
# Convert the data spectrogram to decibel units
spec_data = librosa.power_to_db(librosa.magphase(spec_data, power=2)[0],
ref=np.max)
else:
spec_data = algorithm(y=y, sr=sr, **kwargs)
# Plot spectrogram
fig = plt.figure(figsize=FIG_SIZE)
librosa.display.specshow(spec_data, sr=sr, cmap=cmap, y_axis=y_axis)
plt.axis('off')
fig.subplots_adjust(left=0, right=1, bottom=0, top=1)
if name is None:
name = audiopath.split('/')[-1]
if plotpath is not None:
plt.savefig(plotpath + '/' + name + '.png')
else:
plt.savefig(name + '.png')
plt.close()
def to_spec(wav, len_frame=ModelConfig.L_FRAME, len_hop=ModelConfig.L_HOP):
return librosa.stft(wav, n_fft=len_frame, hop_length=len_hop)
def features(X, sample_rate):
stft = np.abs(librosa.stft(X))
# fmin 和 fmax 对应于人类语音的最小最大基本频率
pitches, magnitudes = librosa.piptrack(X,
sr=sample_rate,
S=stft,
fmin=70,
fmax=400)
pitch = []
for i in range(magnitudes.shape[1]):
index = magnitudes[:, 1].argmax()
pitch.append(pitches[index, i])
pitch_tuning_offset = librosa.pitch_tuning(pitches)
pitchmean = np.mean(pitch)
pitchstd = np.std(pitch)
pitchmax = np.max(pitch)
pitchmin = np.min(pitch)
# 频谱质心
cent = librosa.feature.spectral_centroid(y=X, sr=sample_rate)
cent = cent / np.sum(cent)
meancent = np.mean(cent)
stdcent = np.std(cent)
maxcent = np.max(cent)
# 谱平面
flatness = np.mean(librosa.feature.spectral_flatness(y=X))
# 使用系数为50的MFCC特征
mfccs = np.mean(librosa.feature.mfcc(y=X, sr=sample_rate, n_mfcc=50).T,
axis=0)
mfccsstd = np.std(librosa.feature.mfcc(y=X, sr=sample_rate, n_mfcc=50).T,
axis=0)
mfccmax = np.max(librosa.feature.mfcc(y=X, sr=sample_rate, n_mfcc=50).T,
axis=0)
# 色谱图
chroma = np.mean(librosa.feature.chroma_stft(S=stft, sr=sample_rate).T,
axis=0)
# 梅尔频率
mel = np.mean(librosa.feature.melspectrogram(X, sr=sample_rate).T, axis=0)
# ottava对比
contrast = np.mean(librosa.feature.spectral_contrast(S=stft,
sr=sample_rate).T,
axis=0)
# 过零率
zerocr = np.mean(librosa.feature.zero_crossing_rate(X))
S, phase = librosa.magphase(stft)
meanMagnitude = np.mean(S)
stdMagnitude = np.std(S)
maxMagnitude = np.max(S)
# 均方根能量
rmse = librosa.feature.rmse(S=S)[0]
meanrms = np.mean(rmse)
stdrms = np.std(rmse)
maxrms = np.max(rmse)
ext_features = np.array([
flatness, zerocr, meanMagnitude, maxMagnitude, meancent, stdcent,
maxcent, stdMagnitude, pitchmean, pitchmax, pitchstd,
pitch_tuning_offset, meanrms, maxrms, stdrms
])
ext_features = np.concatenate(
(ext_features, mfccs, mfccsstd, mfccmax, chroma, mel, contrast))
return ext_features
plt.xlabel("Frequency")
plt.ylabel("Magnitude")
plt.title("Power spectrum")
# STFT -> spectrogram
hop_length = 512 # in num. of samples
n_fft = 2048 # window in num. of samples
# calculate duration hop length and window in seconds
hop_length_duration = float(hop_length) / sample_rate
n_fft_duration = float(n_fft) / sample_rate
print("STFT hop length duration is: {}s".format(hop_length_duration))
print("STFT window duration is: {}s".format(n_fft_duration))
# perform stft
stft = librosa.stft(signal, n_fft=n_fft, hop_length=hop_length)
# calculate abs values on complex numbers to get magnitude
spectrogram = np.abs(stft)
# display spectrogram
plt.figure(figsize=FIG_SIZE)
librosa.display.specshow(spectrogram, sr=sample_rate, hop_length=hop_length)
plt.xlabel("Time")
plt.ylabel("Frequency")
plt.colorbar()
plt.title("Spectrogram")
# apply logarithm to cast amplitude to Decibels
log_spectrogram = librosa.amplitude_to_db(spectrogram)
plt.figure(figsize=FIG_SIZE)
for j in range(
len(info1)): #Get the files in the folder inside each id
#print(j)
path1 = path + '/' + info1[j] # Path of the info1 folder
info2 = os.listdir(path1) # Get the audio files
for k in range(len(info2)): # For each audio files
path2 = path1 + '/' + info2[k]
sig, rate = librosa.load(path2, duration=3.0)
#rate, sig = wav.read(path2) # Read .wav signal
sig = scipy.signal.medfilt(sig, kernel_size=None) # Filtering
mfcc1 = librosa.feature.mfcc(
sig, rate, n_mfcc=13) #, hop_length=800, n_fft=1600)
#print(mfcc1.shape)
librosa_stft = np.abs(librosa.stft(sig))
cent = librosa.feature.spectral_centroid(y=sig, sr=rate)
#print(cent.shape)
flat = librosa.feature.spectral_flatness(y=sig)
#print(flat.shape)
rolloff = librosa.feature.spectral_rolloff(y=sig,
sr=rate,
roll_percent=0.99)
#print(rolloff.shape)
zcr = librosa.feature.zero_crossing_rate(sig)
#print(zcr.shape)
Tot_feat = np.vstack(
(mfcc1, librosa_stft, cent, flat, rolloff, zcr))
#Tot_feat = speechpy.processing.cmvn(Tot_feat,variance_normalization=True)
def separate_melody_accompaniment(x,
Fs,
N,
H,
traj,
n_harmonics=10,
tol_cent=50):
"""F0-based melody-accompaniement separation
Notebook: C8/C8S2_MelodyExtractSep.ipynb
Args:
x: Audio signal
Fs: Sampling frequency
N: Window size in samples
H: Hopsize in samples
traj: F0 traj (time in seconds in 1st column, frequency in Hz in 2nd column)
n_harmonics: Number of harmonics
tol_cent: Tolerance in cents
Returns:
x_mel: Reconstructed audio signal for melody
x_acc: Reconstructed audio signal for accompaniement
"""
# Compute STFT
X = librosa.stft(x,
n_fft=N,
hop_length=H,
win_length=N,
pad_mode='constant')
Fs_feature = Fs / H
T_coef = np.arange(X.shape[1]) / Fs_feature
freq_res = Fs / N
F_coef = np.arange(X.shape[0]) * freq_res
# Adjust trajectory
traj_X_values = interp1d(traj[:, 0],
traj[:, 1],
kind='nearest',
fill_value='extrapolate')(T_coef)
traj_X = np.hstack((T_coef[:, None], traj_X_values[:, None, ]))
# Compute binary masks
mask_mel = convert_trajectory_to_mask_cent(traj_X,
F_coef,
n_harmonics=n_harmonics,
tol_cent=tol_cent)
mask_acc = np.ones(mask_mel.shape) - mask_mel
# Compute masked STFTs
X_mel = X * mask_mel
X_acc = X * mask_acc
# Reconstruct signals
x_mel = librosa.istft(X_mel,
hop_length=H,
win_length=N,
window='hann',
center=True,
length=x.size)
x_acc = librosa.istft(X_acc,
hop_length=H,
win_length=N,
window='hann',
center=True,
length=x.size)
return x_mel, x_acc
def extract_IMU_sound_video(video_IMU_data):
features = np.empty((0, Data_window_watch, 12))
labels = []
features_sound = np.empty((0, 193))
features_video = np.empty((0, Data_window_video, 64, 64, 3))
for i in range(len(video_IMU_data)):
print(i)
#Saving features after every 100 samples
if (i + 1) % 100 == 0:
features_video = features_video.astype('uint8')
data = [features, labels, features_sound, features_video]
with open('Data_train_' + str(i) + 'ser.pkl', 'wb') as f:
pickle.dump(data, f)
features = np.empty((0, Data_window_watch, 12))
labels = []
features_sound = np.empty((0, 193))
features_video = np.empty((0, Data_window_video, 64, 64, 3))
#print(video_IMU_data[i][0])
V_name = video_IMU_data[i][0]
V_type = video_IMU_data[i][1]
V_stime = video_IMU_data[i][2]
V_etime = video_IMU_data[i][3]
#print(V_name,V_etime-V_stime)
duration = V_etime - V_stime
#Sound Features
Sound_X = video_IMU_data[i][6]
Sound_sample_rate_expected = video_IMU_data[i][5]
#print('Leng of sound:',len(Sound_X))
sound_len = len(Sound_X)
sound_sampling = (sound_len *
1000) / duration #Duration is in milli seconds
#Sound windows below
sound_win = []
start = 0
end = start + sound_sampling * TimeWindow
end = int(end)
while end <= sound_len:
winx = Sound_X[start:end]
stft = np.abs(librosa.stft(winx))
mfccs = np.mean(librosa.feature.mfcc(y=winx,
sr=Sound_sample_rate_expected,
n_mfcc=40).T,
axis=0)
chroma = np.mean(librosa.feature.chroma_stft(
S=stft, sr=Sound_sample_rate_expected).T,
axis=0)
mel = np.mean(librosa.feature.melspectrogram(
winx, sr=Sound_sample_rate_expected).T,
axis=0)
contrast = np.mean(librosa.feature.spectral_contrast(
S=stft, sr=Sound_sample_rate_expected).T,
axis=0)
tonnetz = np.mean(librosa.feature.tonnetz(
y=librosa.effects.harmonic(winx),
sr=Sound_sample_rate_expected).T,
axis=0)
#print (start,end,len(winx))
ext_features = np.hstack([mfccs, chroma, mel, contrast, tonnetz])
sound_win.append(ext_features)
start = start + sound_sampling * TimeWindow_slide
start = int(start)
end = start + sound_sampling * TimeWindow
end = int(end)
#Video Features and Windows
video_len = video_IMU_data[i][7]
#print(video_len,duration)
video_sampling = (video_len * 1000) / duration
video_raw = []
start = 0
end = start + video_sampling * TimeWindow
end = int(end)
#print('Video Sampling Rate:',video_sampling)
#print('Start is:',start,': End is:',end)
# cap = cv2.VideoCapture(V_name)
# video_continue=True
# while video_continue:
# video_continue, img = cap.read()
# #video_raw.append(cv2.resize(img, (img_size, img_size)))
# video_raw.append(cv2.resize(img, (img_size, img_size)))
# # video_raw.append(img)
cap = cv2.VideoCapture(V_name)
while True:
ret, img = cap.read()
if not ret:
break
video_raw.append(
cv2.resize(img, (img_size, img_size)).astype('uint8'))
Windows_video = []
while end <= video_len:
#print('Start is:',start,': End is:',end)
win_video = video_raw[start:end]
win_video = win_video[:Data_window_video]
Windows_video.append(win_video)
start = start + video_sampling * TimeWindow_slide
start = int(start)
end = start + video_sampling * TimeWindow
end = int(end)
#Dict of sensor data
V_dict = dict()
# Loop over the data items
for j in range(len(video_IMU_data[i][4])):
#print(video_IMU_data[i][4][j])
S_type = video_IMU_data[i][4][j][0]
timestamp = video_IMU_data[i][4][j][2]
x = video_IMU_data[i][4][j][3]
y = video_IMU_data[i][4][j][4]
z = video_IMU_data[i][4][j][5]
if S_type in V_dict:
V_dict[S_type].append([timestamp, x, y, z])
else:
V_dict[S_type] = []
V_dict[S_type].append([timestamp, x, y, z])
#print(S_type,timestamp,x,y,z)
# 3 = Acc Right Wrist
# 4 = GYRO Right Wrist
# 11 = Acc left Wrist
# 12 = GYRO left Wrist
raw3 = []
raw4 = []
raw11 = []
raw12 = []
for sid in V_dict:
if sid == 'raw3':
#print(sid,len(V_dict[sid]))
# Timewindow 2 Sec, Sampling 25HZ.
total_samples = len(V_dict[sid])
sampling = (total_samples * 1000) / (duration)
#print('Sampling',sampling)
start = 0
end = start + sampling * TimeWindow
end = int(end)
while end <= total_samples:
datawind = V_dict[sid][start:end]
#Fixing Datawind_leng Here:
datawind = datawind[:Data_window_watch]
raw3.append(datawind)
#print('window Samples:',len(datawind))
#winlen1.append(len(datawind))
#print(start,end)
start = start + (sampling) * (TimeWindow_slide)
start = int(start)
end = start + sampling * TimeWindow
end = int(end)
if sid == 'raw4':
#print(sid,len(V_dict[sid]))
# Timewindow 2 Sec, Sampling 25HZ.
total_samples = len(V_dict[sid])
sampling = (total_samples * 1000) / (duration)
#print('Sampling',sampling)
start = 0
end = start + sampling * TimeWindow
end = int(end)
while end <= total_samples:
datawind = V_dict[sid][start:end]
#print('window Samples:',len(datawind))
#winlen2.append(len(datawind))
datawind = datawind[:Data_window_watch]
raw4.append(datawind)
#print(start,end)
start = start + (sampling) * (TimeWindow_slide)
start = int(start)
end = start + sampling * TimeWindow
end = int(end)
if sid == 'raw11':
#print(sid,len(V_dict[sid]))
# Timewindow 2 Sec, Sampling 25HZ.
total_samples = len(V_dict[sid])
sampling = (total_samples * 1000) / (duration)
#print('Sampling',sampling)
start = 0
end = start + sampling * TimeWindow
end = int(end)
while end <= total_samples:
datawind = V_dict[sid][start:end]
#print('window Samples:',len(datawind))
#winlen3.append(len(datawind))
datawind = datawind[:Data_window_watch]
raw11.append(datawind)
#print(start,end)
start = start + (sampling) * (TimeWindow_slide)
start = int(start)
end = start + sampling * TimeWindow
end = int(end)
if sid == 'raw12':
#print(sid,len(V_dict[sid]))
# Timewindow 2 Sec, Sampling 25HZ.
total_samples = len(V_dict[sid])
sampling = (total_samples * 1000) / (duration)
#print('Sampling',sampling)
start = 0
end = start + sampling * TimeWindow
end = int(end)
while end <= total_samples:
datawind = V_dict[sid][start:end]
datawind = datawind[:Data_window_watch]
raw12.append(datawind)
#print('window Samples:',len(datawind))
#winlen4.append(len(datawind))
#print(start,end)
start = start + (sampling) * (TimeWindow_slide)
start = int(start)
end = start + sampling * TimeWindow
end = int(end)
#print(sid,len(V_dict[sid]))
#print(len(raw3),len(raw4),len(raw11),len(raw12))
raw3 = np.array(raw3)
raw4 = np.array(raw4)
raw11 = np.array(raw11)
raw12 = np.array(raw12)
sound_win = np.array(sound_win)
Windows_video = np.array(Windows_video)
# We abandon the timestamp
#print(raw3.shape,raw4.shape,raw11.shape,raw12.shape)
raw3_windows = raw3.shape[0]
raw4_windows = raw4.shape[0]
raw11_windows = raw11.shape[0]
raw12_windows = raw12.shape[0]
#print('sensor wind:',raw12_windows)
sound_win_windows = sound_win.shape[0]
video_win_windows = Windows_video.shape[0]
#print('Sound wind:',sound_win_windows)
#print('Video wind:',video_win_windows)
try:
#Sometimes Watch have less features, and we don't know the Reason
min_features_watch = np.array(
[raw3.shape[1], raw4.shape[1], raw11.shape[1],
raw12.shape[1]]).min()
if min_features_watch < Data_window_watch:
print('Skipping:', raw3.shape, raw4.shape, raw11.shape,
raw12.shape)
continue
min_features_Video = Windows_video.shape[1]
if min_features_Video < Data_window_video:
print('Skipping:', Windows_video.shape)
continue
except:
print('Error Skipping:', raw3.shape, raw4.shape, raw11.shape,
raw12.shape)
continue
#print(sound_win_windows)
min_windows = np.array([
raw3_windows, raw4_windows, raw11_windows, raw12_windows,
sound_win_windows, video_win_windows
]).min()
if min_windows > 0:
output = np.concatenate(
(raw3[:min_windows, :, 1:4], raw4[:min_windows, :, 1:4],
raw11[:min_windows, :, 1:4], raw12[:min_windows, :, 1:4]),
axis=2)
#print(output.shape)
features = np.vstack([features, output])
# print(sound_win[:min_windows].shape)
features_sound = np.vstack(
[features_sound, sound_win[:min_windows]])
#print(Windows_video[:min_windows].shape, features_video.shape)
features_video = np.vstack(
[features_video, Windows_video[:min_windows]])
#output.shape[0],V_type
for k in range(output.shape[0]):
labels.append(V_type)
# print(raw4.shape)
# print(output.shape)
# print(features.shape, features_sound.shape, Windows_video[:min_windows].shape )
# break
# 12-num of windows, 47-num of samples in window 12= 3*4 num of sensor reading types
features_video = features_video.astype('uint8')
data = [features, labels, features_sound, features_video]
with open('Data_train_' + str(i) + '.pkl', 'wb') as f:
pickle.dump(data, f)
print('Saved: ' + 'Data_train_' + str(i) + '.pkl')
print(features_video.shape)
return features, labels, features_sound, features_video
def stft(y):
return librosa.stft(
y=y,
n_fft=hp.n_fft, hop_length=hp.hop_length, win_length=hp.win_length)
energy.append(np.mean(e))
ent = 0.0
m = np.mean(e)
for j in range(0,len(e[0])):
q = np.absolute(e[0][j] - m)
ent = ent + (q * np.log10(q))
entropy_of_energy.append(ent)
f_list_1 = []
f_list_1.append(zero_crossings)
f_list_1.append(energy)
f_list_1.append(entropy_of_energy)
f_np_1 = np.array(f_list_1)
f_np_1 = np.transpose(f_np_1)[:-1]
kmeans = KMeans(n_clusters=2, random_state=0).fit(f_np_1)
result=kmeans.predict(f_np_1)
D = li.amplitude_to_db(np.abs(li.stft(y)), ref=np.max)
plt.subplot(3,1,1)
plt.title("Audio Analog Signal")
plt.plot(y[1950:2000])
plt.subplot(3,1,2)
plt.title("Spectogram")
librosa.display.specshow(D, y_axis='linear')
plt.colorbar(format='%+2.0f dB')
plt.subplot(3,1,3)
plt.title("Audio Digital Signal")
plt.plot(result, marker='d', color='blue', drawstyle='steps')
plt.show()
stream.stop_stream()
stream.close()
audio.terminate()
def LoadAudio(file_path):
y, sr = load(file_path, sr=SR)
stft = librosa.stft(y, n_fft=window_size, hop_length=hop_length)
mag, phase = librosa.magphase(stft)
return mag.astype(np.float32), phase
x = np.reshape(x, (-1, N_CHANNELS))
if np.array_equal(x.T, a_content):
print("equal")
else:
print("not equal")
print("content = ", a_content)
print("x = ", x)
diff = a_content - x.T
print("diff = ", diff)
a = np.zeros_like(a_content)
a[:N_CHANNELS, :] = np.exp(x.T) - 1
p = 2 * np.pi * np.random.random_sample(a.shape) - np.pi
for i in range(500):
S = a * np.exp(1j * p)
x = librosa.istft(S)
p = np.angle(librosa.stft(x, N_FFT))
OUTPUT_FILENAME = 'outputs/' + CONTENT_FILENAME[:
-4] + '_' + STYLE_FILENAME[:-4] + '_ctw-' + str(
CONTENT_WEIGHT
) + '_stw-' + str(
STYLE_WEIGHT
) + '_iter-' + str(
ITERATIONS) + '.wav'
librosa.output.write_wav(OUTPUT_FILENAME, x, fs)
print(OUTPUT_FILENAME)
print("done")
my_dpi = 120
for index, row in speakers_filtered.iterrows():
dir_ = root + '/' + row['SUBSET'] + '/' + str(row['ID']) + '/'
print('working on df row {}, spaker {}'.format(index, row['CODE']))
if not os.path.exists(dir_):
print('dir {} not exists, skipping'.format(dir_))
continue
files_iter = Path(dir_).glob('**/*.flac')
files_ = [str(f) for f in files_iter]
for f in files_:
ay, sr = librosa.load(f)
duration = ay.shape[0] / sr
start = 0
while start + 5 < duration:
slice_ = ay[start * sr:(start + 5) * sr]
start = start + 5 - 1
x = librosa.stft(slice_)
xdb = librosa.amplitude_to_db(abs(x))
plt.figure(figsize=(227 / my_dpi, 227 / my_dpi), dpi=my_dpi)
plt.axis('off')
librosa.display.specshow(xdb, sr=sr, x_axis='time', y_axis='log')
plt.savefig(root + '/train-gram/' + str(row['CODE']) + '/' +
uuid.uuid4().hex + '.png',
dpi=my_dpi)
plt.close()
print('work done on index {}, speaker {}'.format(index, row['CODE']))
import numpy as np
import librosa
import librosa.display
import matplotlib.pyplot as plt
# Get the file path to the included audio example
filename = 'Train/blues/blues.00000.au'
# Load the example clip
y, sr = librosa.load(filename)
# Compute spectral centroid
sc = librosa.feature.spectral_centroid(y=y, sr=sr)
# Compute spectrogram
S, phase = librosa.magphase(librosa.stft(y=y))
librosa.feature.spectral_centroid(S=S)
# Plot the result
plt.figure()
plt.subplot(2, 1, 1)
plt.semilogy(sc.T, label='Spectral centroid')
plt.ylabel('Hz')
plt.xticks([])
plt.xlim([0, sc.shape[-1]])
plt.legend()
plt.subplot(2, 1, 2)
librosa.display.specshow(librosa.amplitude_to_db(S, ref=np.max),
y_axis='log',
x_axis='time')
plt.title('log Power spectrogram')
def read_audio_spectum(filename):
x, fs = librosa.load(filename)
S = librosa.stft(x, N_FFT)
p = np.angle(S)
S = np.log1p(np.abs(S[:, :430]))
return S, fs
def getStft(y, n_fft=2048, hop_length=512):
return librosa.feature.rmse(S=librosa.stft(y, n_fft=n_fft, hop_length=hop_length))[0]
def display_sample_info(file_path, label=''):
"""Generate various representations a given audio file.
E.g. Mel, MFCC and power spectrogram's.
Args:
file_path (str): Path to the audio file.
label (str): Optional label to display for the given audio file.
Returns:
Nothing.
"""
if not os.path.isfile(file_path):
raise ValueError('{} does not exist.'.format(file_path))
# By default, all audio is mixed to mono and resampled to 22050 Hz at load time.
y, sr = librosa.load(file_path, sr=None, mono=True)
# At 16000 Hz, 512 samples ~= 32ms. At 16000 Hz, 200 samples = 12ms. 16 samples = 1ms @ 16kHz.
hop_length = 200 # Number of samples between successive frames e.g. columns if a spectrogram.
f_max = sr / 2. # Maximum frequency (Nyquist rate).
f_min = 64. # Minimum frequency.
n_fft = 1024 # Number of samples in a frame.
n_mels = 80 # Number of Mel bins to generate.
n_mfcc = 13 # Number of Mel cepstral coefficients to extract.
win_length = 333 # Window length.
# Create info string.
num_samples = y.shape[0]
duration = librosa.get_duration(y=y, sr=sr)
info_str_format = 'Label: {}\nPath: {}\nDuration={:.3f}s with {:,d} Samples\n' \
'Sampling Rate={:,d} Hz\nMin, Max=[{:.2f}, {:.2f}]'
info_str = info_str_format.format(label, file_path, duration, num_samples,
sr, np.min(y), np.max(y))
print(info_str)
# Escape some LaTeX special characters
info_str_tex = info_str.replace('_', '\\_')
plt.figure(figsize=(10, 7))
plt.subplot(3, 1, 1)
display.waveplot(y, sr=sr)
plt.title('Monophonic')
# Plot waveforms.
y_harm, y_perc = librosa.effects.hpss(y)
plt.subplot(3, 1, 2)
display.waveplot(y_harm, sr=sr, alpha=0.33)
display.waveplot(y_perc, sr=sr, color='r', alpha=0.40)
plt.title('Harmonic and Percussive')
# Add file information.
plt.subplot(3, 1, 3)
plt.axis('off')
plt.text(0.0, 1.0, info_str_tex, color='black', verticalalignment='top')
plt.tight_layout()
# Calculating MEL spectrogram and MFCC.
db_pow = np.abs(
librosa.stft(y=y,
n_fft=n_fft,
hop_length=hop_length,
win_length=win_length))**2
s_mel = librosa.feature.melspectrogram(S=db_pow,
sr=sr,
hop_length=hop_length,
fmax=f_max,
fmin=f_min,
n_mels=n_mels)
s_mel = librosa.power_to_db(s_mel, ref=np.max)
s_mfcc = librosa.feature.mfcc(S=s_mel, sr=sr, n_mfcc=n_mfcc)
# STFT (Short-time Fourier Transform)
# https://librosa.github.io/librosa/generated/librosa.core.stft.html
plt.figure(figsize=(12, 10))
db = librosa.amplitude_to_db(librosa.magphase(librosa.stft(y))[0],
ref=np.max)
plt.subplot(3, 2, 1)
display.specshow(db,
sr=sr,
x_axis='time',
y_axis='linear',
hop_length=hop_length)
plt.colorbar(format='%+2.0f dB')
plt.title('Linear-frequency power spectrogram')
plt.subplot(3, 2, 2)
display.specshow(db,
sr=sr,
x_axis='time',
y_axis='log',
hop_length=hop_length)
plt.colorbar(format='%+2.0f dB')
plt.title('Log-frequency power spectrogram')
plt.subplot(3, 2, 3)
display.specshow(s_mfcc,
sr=sr,
x_axis='time',
y_axis='linear',
hop_length=hop_length)
plt.colorbar(format='%+2.0f dB')
plt.title('MFCC spectrogram')
# # CQT (Constant-T Transform)
# # https://librosa.github.io/librosa/generated/librosa.core.cqt.html
cqt = librosa.amplitude_to_db(librosa.magphase(librosa.cqt(y, sr=sr))[0],
ref=np.max)
# plt.subplot(3, 2, 3)
# display.specshow(cqt, sr=sr, x_axis='time', y_axis='cqt_note', hop_length=hop_length)
# plt.colorbar(format='%+2.0f dB')
# plt.title('Constant-Q power spectrogram (note)')
plt.subplot(3, 2, 4)
display.specshow(cqt,
sr=sr,
x_axis='time',
y_axis='cqt_hz',
hop_length=hop_length)
plt.colorbar(format='%+2.0f dB')
plt.title('Constant-Q power spectrogram (Hz)')
plt.subplot(3, 2, 5)
display.specshow(db,
sr=sr,
x_axis='time',
y_axis='log',
hop_length=hop_length)
plt.colorbar(format='%+2.0f dB')
plt.title('Log power spectrogram')
plt.subplot(3, 2, 6)
display.specshow(s_mel, x_axis='time', y_axis='mel', hop_length=hop_length)
plt.colorbar(format='%+2.0f dB')
plt.title('Mel spectrogram')
# TODO Import project used features (python_speech_features).
# norm_features = 'none'
# mfcc = load_sample(file_path, feature_type='mfcc', feature_normalization=norm_features)[0]
# mfcc = np.swapaxes(mfcc, 0, 1)
#
# mel = load_sample(file_path, feature_type='mel', feature_normalization=norm_features)[0]
# mel = np.swapaxes(mel, 0, 1)
(__sr, __y) = wavfile.read(file_path)
num_features = 26
win_len = WIN_LENGTH
win_step = WIN_STEP
__mel = psf.logfbank(signal=__y,
samplerate=__sr,
winlen=win_len,
winstep=win_step,
nfilt=num_features,
nfft=n_fft,
lowfreq=f_min,
highfreq=f_max,
preemph=0.97)
__mfcc = psf.mfcc(signal=__y,
samplerate=__sr,
winlen=win_len,
winstep=win_step,
numcep=num_features // 2,
nfilt=num_features,
nfft=n_fft,
lowfreq=f_min,
highfreq=f_max,
preemph=0.97,
ceplifter=22,
appendEnergy=False)
__mfcc = __mfcc.astype(np.float32)
__mel = __mel.astype(np.float32)
__mfcc = np.swapaxes(__mfcc, 0, 1)
__mel = np.swapaxes(__mel, 0, 1)
plt.figure(figsize=(5.2, 1.6))
display.waveplot(y, sr=sr)
fig = plt.figure(figsize=(10, 4))
plt.subplot(2, 1, 2)
display.specshow(__mfcc,
sr=__sr,
x_axis='time',
y_axis='mel',
hop_length=win_step * __sr)
# plt.set_cmap('magma')
# plt.xticks(rotation=295)
plt.xlabel('Time (s)')
plt.ylabel('Frequency (Hz)')
plt.xlim(xmin=0)
plt.ylim(0, 8000)
plt.colorbar(format='%+2.0f')
plt.title('MFCC', visible=False)
plt.subplot(2, 1, 1)
display.specshow(__mel,
sr=__sr,
x_axis='time',
y_axis='mel',
hop_length=win_step * __sr)
# plt.set_cmap('magma')
# plt.xticks(rotation=295)
plt.xlabel('Time (s)')
plt.ylabel('Frequency (Hz)')
plt.xlim(xmin=0)
plt.ylim(0, 8000)
plt.colorbar(format='%+2.0f', label='Power (dB)')
plt.title('Mel Spectrogram', visible=False)
plt.tight_layout()
fig.savefig('/tmp/mel-mfcc-plot-we-did-it.pdf', bbox_inches='tight')
plt.show()
def hpss_wav(y):
H, P = librosa.decompose.hpss(librosa.stft(y))
return librosa.istft(H), librosa.istft(P)
def gen_audio_features(item, config):
"""Generate audio features and transformations
Args:
item (Dict): dictionary containing the attributes to encode.
config (Dict): configuration dictionary.
Returns:
(bool): keep this sample or not.
mel (ndarray): mel matrix in np.float32.
energy (ndarray): energy audio profile.
f0 (ndarray): fundamental frequency.
item (Dict): dictionary containing the updated attributes.
"""
# get info from sample.
audio = item["audio"]
utt_id = item["utt_id"]
rate = item["rate"]
# check audio properties
assert len(audio.shape) == 1, f"{utt_id} seems to be multi-channel signal."
assert np.abs(
audio).max() <= 1.0, f"{utt_id} is different from 16 bit PCM."
# check sample rate
if rate != config["sampling_rate"]:
audio = librosa.resample(audio, rate, config["sampling_rate"])
logging.info(
f"{utt_id} sampling rate is {rate}, not {config['sampling_rate']}, we resample it."
)
# trim silence
if config["trim_silence"]:
if "trim_mfa" in config and config["trim_mfa"]:
_, item["text_ids"], audio = ph_based_trim(
config,
utt_id,
item["text_ids"],
item["raw_text"],
audio,
config["hop_size"],
)
if (
audio.__len__() < 1
): # very short files can get trimmed fully if mfa didnt extract any tokens LibriTTS maybe take only longer files?
logging.warning(
f"File have only silence or MFA didnt extract any token {utt_id}"
)
return False, None, None, None, item
else:
audio, _ = librosa.effects.trim(
audio,
top_db=config["trim_threshold_in_db"],
frame_length=config["trim_frame_size"],
hop_length=config["trim_hop_size"],
)
# resample audio if necessary
if "sampling_rate_for_feats" in config:
audio = librosa.resample(audio, rate,
config["sampling_rate_for_feats"])
sampling_rate = config["sampling_rate_for_feats"]
assert (
config["hop_size"] * config["sampling_rate_for_feats"] % rate == 0
), "'hop_size' must be 'int' value. Please check if 'sampling_rate_for_feats' is correct."
hop_size = config["hop_size"] * config[
"sampling_rate_for_feats"] // rate
else:
sampling_rate = config["sampling_rate"]
hop_size = config["hop_size"]
# get spectrogram
D = librosa.stft(
audio,
n_fft=config["fft_size"],
hop_length=hop_size,
win_length=config["win_length"],
window=config["window"],
pad_mode="reflect",
)
S, _ = librosa.magphase(D) # (#bins, #frames)
# get mel basis
fmin = 0 if config["fmin"] is None else config["fmin"]
fmax = sampling_rate // 2 if config["fmax"] is None else config["fmax"]
mel_basis = librosa.filters.mel(
sr=sampling_rate,
n_fft=config["fft_size"],
n_mels=config["num_mels"],
fmin=fmin,
fmax=fmax,
)
mel = np.log10(np.maximum(np.dot(mel_basis, S),
1e-10)).T # (#frames, #bins)
mel_eos = np.zeros(shape=[1, np.shape(mel)[1]
]) # (1, #bins) # represent mel for eos_token.
mel = np.concatenate([mel, mel_eos], axis=0) # (#frames + 1, #bins)
# check audio and feature length
audio_eos = np.zeros(
shape=[hop_size]) # (hop_size) # represent audio for eos_token.
audio = np.concatenate([audio, audio_eos], axis=-1)
audio = np.pad(audio, (0, config["fft_size"]), mode="edge")
audio = audio[:len(mel) * hop_size]
assert len(mel) * hop_size == len(
audio), f"{len(mel) * hope_size}, {len(audio)}"
# extract raw pitch
_f0, t = pw.dio(
audio.astype(np.double),
fs=sampling_rate,
f0_ceil=fmax,
frame_period=1000 * hop_size / sampling_rate,
)
f0 = pw.stonemask(audio.astype(np.double), _f0, t, sampling_rate)
if len(f0) >= len(mel):
f0 = f0[:len(mel)]
else:
f0 = np.pad(f0, (0, len(mel) - len(f0)))
# extract energy
energy = np.sqrt(np.sum(S**2, axis=0))
energy = np.concatenate([energy, [0]],
axis=-1) # # represent energy for eos_token.
assert len(mel) == len(f0) == len(
energy), f"{len(mel)}, {len(f0)}, {len(energy)}"
# apply global gain
if config["global_gain_scale"] > 0.0:
audio *= config["global_gain_scale"]
if np.abs(audio).max() >= 1.0:
logging.warn(
f"{utt_id} causes clipping. It is better to reconsider global gain scale value."
)
item["audio"] = audio
item["mel"] = mel
item["f0"] = remove_outlier(f0)
item["energy"] = remove_outlier(energy)
return True, mel, energy, f0, item
t=np.linspace(0,N/fe,N);
s = 0.2*np.cos(2*np.pi*200*t) + 2*np.cos(2*np.pi*400*t);
tf=np.linspace(0,fe/N,N);
plt.subplot(1,2,1);
plt.plot(t[:200],s[:200]);
plt.title('280Hz et 500Hz,fe=8000Hz')
plt.subplot(1,2,2);
plt.plot(np.abs(np.fft.fft(s)));
plt.title('280Hz et 500Hz,fe=8000Hz')
"""
#x, fe = librosa.load('ressources/mesange-tete-noire.wav')
x, fe = librosa.load('ressources/PIANO.wav')
plt.figure(figsize=(14, 5))
librosa.display.waveplot(x, sr=fe)
plt.title('')
plt.show()
fe /= 2
n = len(x)
t = np.linspace(0, n / fe, n, endpoint=False)
s = 0.75 * np.cos(2 * np.pi * 440 * t)
plt.plot(t, x)
plt.plot(np.abs(np.fft.fft(s)))
Sdb = librosa.amplitude_to_db(abs(s))
S = np.abs(librosa.stft(s))
Sdb = librosa.amplitude_to_db(abs(S))
#librosa.display.specshow(Sdb, sr=fe, x_axis='time', y_axis='hz')
#librosa.display.specshow(Sdb, sr=fe, x_axis='time', y_axis='hz')
sd.play(x, fe)
status = sd.wait()
def calculate_melsp(x, n_fft=1024, hop_length=128):
stft = np.abs(librosa.stft(x, n_fft=n_fft, hop_length=hop_length))**2
log_stft = librosa.power_to_db(stft)
melsp = librosa.feature.melspectrogram(S=log_stft, n_mels=128)
return melsp
def decompose_audio(y):
harmonic, percussive = librosa.decompose.hpss(librosa.stft(y), margin=2)
harmonic = librosa.istft(harmonic)
percussive = librosa.istft(percussive)
return harmonic, percussive
import ntpath
root = os.path.dirname(os.path.realpath(__file__))
path_name = r'test'
direc_name = os.path.join(root, path_name)
train_path = r'test/audio'
csv_file = os.path.join(direc_name, 'features_test.csv')
folders = os.listdir(path_name)
with open(csv_file, "w", newline='') as output:
audio_class_folder = os.path.join(root, train_path)
files = os.listdir(audio_class_folder)
for file in files:
print(file)
X, samp_rate = librosa.load(os.path.join(audio_class_folder, file))
stft = np.array(np.abs(librosa.stft(X)))
mfcc = np.array(
np.mean(librosa.feature.mfcc(y=X, sr=samp_rate, n_mfcc=40).T,
axis=0))
chroma = np.array(
np.mean(librosa.feature.chroma_stft(S=stft, sr=samp_rate).T,
axis=0))
contrast = np.array(
np.mean(librosa.feature.spectral_contrast(S=stft, sr=samp_rate).T,
axis=0))
features = np.append(mfcc, chroma)
features = np.append(features, contrast)
features_full = features.tolist()
writer = csv.writer(output, delimiter=',')
writer.writerow(features_full)
print('Yay')