def song(self,path):
y, sr = librosa.load(self.path, duration=120)
S_full, phase = librosa.magphase(librosa.stft(y))
S_filter = librosa.decompose.nn_filter(S_full,aggregate=np.median,metric='cosine',
width=int(librosa.time_to_frames(2, sr=sr)))
S_filter = np.minimum(S_full, S_filter)
margin_i, margin_v = 2, 10
power = 2
mask_i = librosa.util.softmask(S_filter,margin_i * (S_full - S_filter),power=power)
mask_v = librosa.util.softmask(S_full - S_filter,margin_v * S_filter,power=power)
S_foreground = mask_v * S_full
S_background = mask_i * S_full
music =librosa.griffinlim(S_background)
vocal =librosa.griffinlim(S_foreground)
scipy.io.wavfile.write('sound_results/song/music.wav',sr,music)
scipy.io.wavfile.write('sound_results/song/vocal.wav',sr,vocal)
utl.plotSounds([music, vocal], ["music", "vocal"], sr, "plot_results/song/song_separation_plot.png")
img = pg.QtGui.QGraphicsPixmapItem(pg.QtGui.QPixmap('plot_results/song/song_separation_plot.png'))
self.ui1.widget_song.addItem(img)
self.ui1.widget_song.invertY(True)
self.alarm("Check Plot & Sound Results Files")
def spectrogram_to_audio(data, data_recon, output_dir):
data_np = (data.squeeze(0).to(torch.device("cpu"))).detach().numpy()
data_griffin_lim = librosa.griffinlim(data_np)
data_recon_np = (data_recon.squeeze(0).to(
torch.device("cpu"))).detach().numpy()
data_recon_griffin_lim = librosa.griffinlim(data_recon_np)
source_aud_path = output_dir + '_input_' + '.wav'
target_aud_path = output_dir + '_output_' + '.wav'
librosa.output.write_wav(source_aud_path, data_griffin_lim, 16000)
librosa.output.write_wav(target_aud_path, data_recon_griffin_lim, 16000)
return source_aud_path, target_aud_path
def img_to_audio(image=None, out_wav=None, sr=48000, hl=None, wl=None):
if image == None or ".png" not in image:
print("Please Specify an image file! (e.g. my_image.png)")
return None
elif out_wav == None or ".wav" not in out_wav:
print("Please Specify an output file! (e.g. my_sound.wav)")
return None
if wl == "None":
wl = None
if hl == "None":
hl = None
basepath = path.dirname(__file__)
filepath = path.abspath(path.join(basepath, 'images', image))
print(f"Image location: {filepath}")
img = cv2.imread(filepath)
print(f"Read image of shape {img.shape}")
avg_img = np.mean(img, axis=2)
snd = librosa.griffinlim(avg_img, n_iter=64, hop_length=hl, win_length=wl)
print(f"Output audio with {len(snd)} samples and a sample rate of {sr} Hz")
librosa.output.write_wav(f"audio/{out_wav}", snd, sr)
def test_griffinlim(self):
# NOTE: This test is flaky without a fixed random seed
# See https://github.com/pytorch/audio/issues/382
torch.random.manual_seed(42)
tensor = torch.rand((1, 1000))
n_fft = 400
ws = 400
hop = 100
window = torch.hann_window(ws)
normalize = False
momentum = 0.99
n_iter = 8
length = 1000
rand_init = False
init = 'random' if rand_init else None
specgram = F.spectrogram(tensor, 0, window, n_fft, hop, ws, 2,
normalize).sqrt()
ta_out = F.griffinlim(specgram, window, n_fft, hop, ws, 1, normalize,
n_iter, momentum, length, rand_init)
lr_out = librosa.griffinlim(specgram.squeeze(0).numpy(),
n_iter=n_iter,
hop_length=hop,
momentum=momentum,
init=init,
length=length)
lr_out = torch.from_numpy(lr_out).unsqueeze(0)
self.assertTrue(torch.allclose(ta_out, lr_out, atol=5e-5))
def get_audio_from_stft_spectrogram_GL(self, stft_features):
return librosa.griffinlim(stft_features,
init=None,
win_length=self.window_length,
hop_length=self.overlap,
window=self.window,
center=True)
def test_griffinlim(self, momentum):
# FFT params
n_fft = 400
win_length = n_fft
hop_length = n_fft // 4
window = torch.hann_window(win_length, device=self.device)
power = 1
# GriffinLim params
n_iter = 8
waveform = get_whitenoise(device=self.device, dtype=self.dtype)
specgram = get_spectrogram(
waveform, n_fft=n_fft, hop_length=hop_length, power=power,
win_length=win_length, window=window)
result = F.griffinlim(
specgram,
window=window,
n_fft=n_fft,
hop_length=hop_length,
win_length=win_length,
power=power,
n_iter=n_iter,
momentum=momentum,
length=waveform.size(1),
rand_init=False)
expected = librosa.griffinlim(
specgram[0].cpu().numpy(),
n_iter=n_iter,
hop_length=hop_length,
momentum=momentum,
init=None,
length=waveform.size(1))[None, ...]
self.assertEqual(result, torch.from_numpy(expected), atol=5e-5, rtol=1e-07)
def griffin_lim_(spc, n_fft, n_shift, win_length, window='hann', n_iters=100):
"""Convert linear spectrogram into waveform using Griffin-Lim.
Args:
spc (ndarray): Linear spectrogram (T, n_fft // 2 + 1).
n_fft (int): Number of FFT points.
n_shift (int): Shift size in points.
win_length (int): Window length in points.
window (str, optional): Window function type.
n_iters (int, optionl): Number of iterations of Griffin-Lim Algorithm.
Returns:
ndarray: Reconstructed waveform (N,).
"""
# assert the size of input linear spectrogram
assert spc.shape[1] == n_fft // 2 + 1
spc = np.abs(spc.T)
y = librosa.griffinlim(
S=spc,
n_iter=n_iters,
hop_length=n_shift,
win_length=win_length,
window=window
)
return y
def inverse_stft_griffin_lim(self, stft_mat, name):
filename = self.output_path + self.h_params.time_for_output + "_griffin_" + self.output_name + "_" + name + ".wav"
istft_mat = librosa.griffinlim(
abs(stft_mat),
hop_length=self.h_params.stft_hop_length,
win_length=self.h_params.stft_window_size)
sf.write(filename, istft_mat, self.h_params.down_sample_rate)
return istft_mat
def spec_wav(specgram, filename):
# Return the all-zero vector with the same shape of `a_content`
a = np.exp(specgram.cpu().detach().numpy()) - 1
p = 2 * np.pi * np.random.random_sample(specgram.shape) - np.pi
x = librosa.griffinlim(a)
p = np.angle(librosa.stft(x, 400))
librosa.output.write_wav(filename, x, sr=22050)
def audio_reconstruction_test(src_dir, dest_dir, ext=".png", size=None):
"""
Test different approaches to image to audio conversion
:param src_dir: image directory
:param dest_dir: audio directory
:param ext: image type
:param size: desired dimension for resizing
"""
paths = prep_utils.get_unprocessed_items(src_dir=src_dir,
dest_dir=dest_dir)
start_time = time.time()
for path in paths:
prep_utils.display_progress_eta(current_item=path,
total_items=paths,
start_time=start_time)
S = np.load(path)
cv2.imshow("image", S)
cv2.waitKey(0)
pd.DataFrame(S).to_csv(dest_dir + "S.csv", header=None, index=False)
S_scaled = prep_utils.increase_brightness(S)
cv2.imshow("image", S_scaled)
cv2.waitKey(0)
pd.DataFrame(S_scaled).to_csv(dest_dir + "S_scaled.csv",
header=None,
index=False)
out_path = dest_dir + "gray" + ext
cv2.imwrite(out_path, S_scaled)
S = cv2.imread(out_path, 0)
S = np.array(S, dtype=np.float32)
S_recovered = S
if size:
S_recovered = cv2.resize(S_recovered, (size, size),
interpolation=cv2.INTER_CUBIC)
pd.DataFrame(S_recovered).to_csv(dest_dir + "S_recovered.csv",
header=None,
index=False)
out_path = dest_dir + "resized" + ext
cv2.imwrite(out_path, S_recovered)
S_audio = np.genfromtxt(dest_dir + "S_recovered.csv", delimiter=',')
S_audio = np.array(S_audio, dtype=np.float32)
S_audio = cv2.resize(S_audio, (431, 1025),
interpolation=cv2.INTER_CUBIC)
y = librosa.griffinlim(S_audio)
out = dest_dir + "s.wav"
# Save reconstructed data
scipy.io.wavfile.write(out, 22050, y)
break
def inv_linear_spectrogram(linear_spectrogram):
if hparams.signal_normalization:
D = _denormalize(linear_spectrogram)
else:
D = linear_spectrogram
S = _db_to_amp(D + hparams.ref_level_db) # Convert back to linear
return librosa.griffinlim(S**1.55)
def get_spectrogram(wav):
y, sr = librosa.load(librosa.ex('trumpet'))
# Get the magnitude spectrogram
S = np.abs(librosa.stft(y))
# Invert using Griffin-Lim
y_inv = librosa.griffinlim(S)
# Invert without estimating phase
y_istft = librosa.istft(S)
return S
def inv_spectrogram(spectrogram):
'''Converts spectrogram to waveform using librosa'''
S = _db_to_amp(_denormalize(spectrogram) +
hparams.spec_ref_level_db) # Convert back to linear
#S = librosa.db_to_amplitude(_denormalize(spectrogram) + hparams.spec_ref_level_db)
D = librosa.griffinlim(S**hparams.power,
hop_length=hparams.hop_size,
win_length=hparams.fft_wsize)
return inv_preemphasis(D)
def inv_mel_spectrogram(mel_spectrogram):
if hparams.signal_normalization:
D = _denormalize(mel_spectrogram)
else:
D = mel_spectrogram
S = _mel_to_linear(
_db_to_amp(D + hparams.ref_level_db)) # Convertir de vuelta a lineal
return librosa.griffinlim(S**1.55)
def inverse_spectrogram(s, mel=False):
"""Convert log-magnitude spectrogram to waveform."""
S = db_to_amplitude(s)
wf = ms_to_frames(hp.stft_window_ms)
hf = ms_to_frames(hp.stft_shift_ms)
if mel: S = librosa.feature.inverse.mel_to_stft(S, power=1, sr=hp.sample_rate, n_fft=hp.num_fft)
y = librosa.griffinlim(S ** hp.griffin_lim_power, n_iter=hp.griffin_lim_iters, hop_length=hf, win_length=wf)
if hp.use_preemphasis: y = deemphasis(y)
y /= max(y)
return y
def griffin_lim(level,audio,args=1):
starting_w=80.0
ending_w = 1.0
levels = 100.0
noise_level = starting_w + level *((ending_w-starting_w)/100)
noise_level=round(noise_level,4)
S = np.abs(librosa.stft(audio))
y_inv = librosa.griffinlim(S,n_iter=int(noise_level))
return y_inv
def test_griffinlim_multi(y_multi):
y, sr = y_multi
# Compute the stft
D = librosa.stft(y)
# Run a couple of iterations of griffin-lim
yout = librosa.griffinlim(np.abs(D), n_iter=2, length=y.shape[-1])
# Check the lengths
assert np.allclose(y.shape, yout.shape)
def _spec_to_wav(spec, sr=sample_rate, engine='librosa'):
''' using Griffin-Lim algorithm '''
if engine == 'librosa':
return librosa.griffinlim(spec,
hop_length=stft_params['hop_length'],
win_length=stft_params['win_length'])
elif engine == 'torch':
return tf.GriffinLim(**stft_params, power=power)(spec)
raise ValueError(engine)
def griffin_lim(
spc: np.ndarray,
n_fft: int,
n_shift: int,
win_length: int = None,
window: Optional[str] = "hann",
n_iter: Optional[int] = 32,
) -> np.ndarray:
"""Convert linear spectrogram into waveform using Griffin-Lim.
Args:
spc: Linear spectrogram (T, n_fft // 2 + 1).
n_fft: The number of FFT points.
n_shift: Shift size in points.
win_length: Window length in points.
window: Window function type.
n_iter: The number of iterations.
Returns:
Reconstructed waveform (N,).
"""
# assert the size of input linear spectrogram
assert spc.shape[1] == n_fft // 2 + 1
if V(librosa.__version__) >= V("0.7.0"):
# use librosa's fast Grriffin-Lim algorithm
spc = np.abs(spc.T)
y = librosa.griffinlim(
S=spc,
n_iter=n_iter,
hop_length=n_shift,
win_length=win_length,
window=window,
center=True if spc.shape[1] > 1 else False,
)
else:
# use slower version of Grriffin-Lim algorithm
logging.warning(
"librosa version is old. use slow version of Grriffin-Lim algorithm."
"if you want to use fast Griffin-Lim, please update librosa via "
"`source ./path.sh && pip install librosa==0.7.0`."
)
cspc = np.abs(spc).astype(np.complex).T
angles = np.exp(2j * np.pi * np.random.rand(*cspc.shape))
y = librosa.istft(cspc * angles, n_shift, win_length, window=window)
for i in range(n_iter):
angles = np.exp(
1j
* np.angle(librosa.stft(y, n_fft, n_shift, win_length, window=window))
)
y = librosa.istft(cspc * angles, n_shift, win_length, window=window)
return y
def post_audio(self, y):
y = np.array(y)
y = np.exp(y)
y = librosa.feature.inverse.mel_to_stft(y,
sr=self.sr,
n_fft=self.n_fft,
power=self.power)
y = librosa.griffinlim(y,
hop_length=self.hop_length,
win_length=self.win_length)
return y
def griffinlim_librosa(spectrogram, fs, hparams):
hop_length = int(hparams.hop_length_ms / 1000 * fs)
win_length = int(hparams.win_length_ms / 1000 * fs)
return inv_preemphasis(
librosa.griffinlim(
spectrogram,
n_iter=hparams.griffin_lim_iters,
hop_length=hop_length,
win_length=win_length,
),
hparams,
)
def extract_audio(
Z, feature,
params): # if normalized Z: unnormalize first, then pass to func.
# convert to audio
if feature == "Stft":
# undo log-magnitude scaling
S = librosa.db_to_amplitude(Z)
# upsample
S = _upsample_fft(S, params["fft_sample_rate"],
params["stft_window_length"])
yhat = librosa.griffinlim(S, hop_length=params["stft_hop_length"])
elif feature == "Mel":
# undo log-power scaling
S = librosa.db_to_power(Z)
yhat = librosa.feature.inverse.mel_to_audio(
S,
sr=params["fft_sample_rate"],
n_fft=params["stft_window_length"],
hop_length=params["stft_hop_length"],
)
elif feature == "Cqt":
# undo log-amplitude scaling
S = librosa.db_to_amplitude(Z)
yhat = librosa.griffinlim_cqt(
S,
sr=params["fft_sample_rate"],
hop_length=params["stft_hop_length"],
fmin=librosa.note_to_hz(params["cqt_min_frequency"]),
)
elif feature == "Mfcc":
yhat = librosa.feature.inverse.mfcc_to_audio(
Z,
n_mels=params["frequency_bins"],
sr=params["fft_sample_rate"],
n_fft=params["stft_window_length"],
hop_length=params["stft_hop_length"],
)
else:
print("Error: feature invalid")
# throw/raise something
return -1
return yhat, params["fft_sample_rate"]
def mel_to_sound(dst_sound_file_path,
mel_spectrogram,
sample_rate,
method='griffin_lim'):
'''
converts mel spectrogram (librosa object) to sound file with defined sample rate and 1 method:
griffin_lim
https://paperswithcode.com/method/griffin-lim-algorithm
'''
inverted_features = librosa.feature.inverse.mel_to_stft(mel_spectrogram)
audio_signal = librosa.griffinlim(inverted_features)
scipy.io.wavfile.write(dst_sound_file_path, audio_signal, sample_rate)
def griffinlim_sp(spectrogram, n_fft, win_length, hop_length):
if win_length is None:
win_length = n_fft
if hop_length is None:
hop_length = win_length // 4
s = librosa.griffinlim(
spectrogram,
n_iter=50,
hop_length=hop_length,
win_length=win_length,
momentum=0.5,
)
return s / np.max(np.abs(s))
def amp_sp_to_raw(amp_sp: np.array,
fs: int,
hop_size_ms: int = 5,
preemphasis: float = 0.00):
"""
Transform the amplitude spectrum into the waveform with
Griffin-Lim. The amplitude spectrum has to have the pitch
information. Using amplitude spectrum which was extracted with
pitch aligned windows (as WORLD does it) will not work.
"""
raw = librosa.griffinlim(amp_sp.T * np.sqrt(amp_sp.shape[1]),
hop_length=int(fs * hop_size_ms / 1000.))
return AudioProcessing.depreemphasis(raw, preemphasis)
def synthesize(self, mel):
mel = self._denormalize_from_VC(mel)
lin_out = librosa.feature.inverse.mel_to_stft(mel,
sr=Config.audio_sr,
n_fft=Config.n_fft,
fmin=Config.fmin,
fmax=Config.fmax)
waveform = librosa.griffinlim(lin_out,
win_length=Config.win_length,
hop_length=Config.hop_length)
return waveform
def spec2audio(cls,
spec: np.ndarray,
n_iter: int = 60,
enhancement_factor: float = 1.5) -> np.ndarray:
"""Converts magnitude spectrogram into a waveform using Griffin-Lim
algorithm and istft"""
spec = cls.denormalize(spec)
spec = cls.db2lin(spec + AudioProcessParam.ref_level_db)
spec = np.power(spec, enhancement_factor) # enhance
waveform = librosa.griffinlim(spec,
n_iter=n_iter,
hop_length=cls.param.hop_length)
# TODO: de-emphasize the waveform?
return cls.de_emphasize(waveform)
def griffinlim(*targets):
try:
print("Griffinlim booted")
while True:
S = (yield)
# Invert using Griffin-Lim
y_inv = librosa.griffinlim(S)
for t in targets:
print(t)
t.send(y_inv)
except GeneratorExit:
print("Griffinlim shutdown")
for t in targets:
t.close()
def load_vocal_audio(self, y, sr):
S_full, phase = librosa.magphase(librosa.stft(y))
S_filter = librosa.decompose.nn_filter(S_full,
aggregate=np.median,
metric='cosine',
width=int(librosa.time_to_frames(2, sr=sr)))
S_filter = np.minimum(S_full, S_filter)
margin_i, margin_v = 2, 10
power = 2
mask_v = librosa.util.softmask(S_full - S_filter,
margin_v * S_filter,
power=power)
S_foreground = mask_v * S_full
output_data = librosa.griffinlim(S_foreground)
return output_data, sr
def spec_to_waveform(spec, min_level_db, ref_level_db, power, n_iter,
win_length, hop_length, preemphasis):
"""Convert output linear spec to waveform using griffin-lim vocoder.
Args:
spec (ndarray): the output linear spectrogram, shape(C, T), where C means n_fft, T means frames.
"""
denoramlized = np.clip(spec, 0, 1) * (-min_level_db) + min_level_db
lin_scaled = np.exp((denoramlized + ref_level_db) / 20 * np.log(10))
wav = librosa.griffinlim(lin_scaled**power,
n_iter=n_iter,
hop_length=hop_length,
win_length=win_length)
if preemphasis > 0:
wav = signal.lfilter([1.], [1., -preemphasis], wav)
return wav