def pysptk_mfcc(self):
self.frame_length = 1024
self.hop_length = 80
self.pitch = pysptk.swipe(self.audio.astype(np.float64),
fs=self.sr,
hopsize=self.hop_length,
min=60,
max=240,
otype="pitch")
self.source_excitation = pysptk.excite(self.pitch, self.hop_length)
# Note that almost all of pysptk functions assume input array is C-contiguous and np.float4 element type
frames = librosa.util.frame(self.audio,
frame_length=self.frame_length,
hop_length=self.hop_length).astype(
np.float64).T
# Windowing
frames *= pysptk.blackman(self.frame_length)
assert frames.shape[1] == self.frame_length
# Order of mel-cepstrum
self.order = 25
self.alpha = 0.41
self.mc = pysptk.mcep(frames, self.order, self.alpha)
logH = pysptk.mgc2sp(self.mc, self.alpha, 0.0, self.frame_length).real
librosa.display.specshow(logH.T,
sr=self.sr,
hop_length=self.hop_length,
x_axis="time",
y_axis="linear")
def __call__(self, pkg, cached_file=None):
pkg = format_package(pkg)
wav = pkg['chunk']
wav = wav.data.numpy()
max_frames = wav.shape[0] // self.hop
if cached_file is not None:
# load pre-computed data
proso = torch.load(cached_file)
beg_i = pkg['chunk_beg_i'] // self.hop
end_i = pkg['chunk_end_i'] // self.hop
proso = proso[:, beg_i:end_i]
pkg[self.name] = proso
else:
# first compute logF0 and voiced/unvoiced flag
# f0 = pysptk.rapt(wav.astype(np.float32),
# fs=self.sr, hopsize=self.hop,
# min=self.f0_min, max=self.f0_max,
# otype='f0')
f0 = pysptk.swipe(wav.astype(np.float64),
fs=self.sr, hopsize=self.hop,
min=self.f0_min,
max=self.f0_max,
otype='f0')
# sound = pm.Sound(wav.astype(np.float32), self.sr)
# f0 = sound.to_pitch(self.hop / 16000).selected_array['frequency']
if len(f0) < max_frames:
pad = max_frames - len(f0)
f0 = np.concatenate((f0, f0[-pad:]), axis=0)
lf0 = np.log(f0 + 1e-10)
lf0, uv = interpolation(lf0, -1)
lf0 = torch.tensor(lf0.astype(np.float32)).unsqueeze(0)[:, :max_frames]
uv = torch.tensor(uv.astype(np.float32)).unsqueeze(0)[:, :max_frames]
if torch.sum(uv) == 0:
# if frame is completely unvoiced, make lf0 min val
lf0 = torch.ones(uv.size()) * np.log(self.f0_min)
# assert lf0.min() > 0, lf0.data.numpy()
# secondly obtain zcr
zcr = librosa.feature.zero_crossing_rate(y=wav,
frame_length=self.win,
hop_length=self.hop)
zcr = torch.tensor(zcr.astype(np.float32))
zcr = zcr[:, :max_frames]
# finally obtain energy
egy = librosa.feature.rmse(y=wav, frame_length=self.win,
hop_length=self.hop,
pad_mode='constant')
egy = torch.tensor(egy.astype(np.float32))
egy = egy[:, :max_frames]
proso = torch.cat((lf0, uv, egy, zcr), dim=0)
if self.der_order > 0 :
deltas=[proso]
for n in range(1,self.der_order+1):
deltas.append(librosa.feature.delta(proso.numpy(),order=n))
proso=torch.from_numpy(np.concatenate(deltas))
pkg[self.name] = proso
# Overwrite resolution to hop length
pkg['dec_resolution'] = self.hop
return pkg
def get_coefs(wav_file_path):
sample_rate, x = wavfile.read(wav_file_path)
# al.play(x.astype(float) / x.max(), fs=sample_rate)
frames = librosa.util.frame(
x,
frame_length=frame_length,
hop_length=hop_length).astype(np.float64).T
frames *= pysptk.blackman(frame_length)
f0 = pysptk.swipe(
x.astype(np.float64),
fs=sample_rate,
hopsize=hop_length,
min=50,
max=500)
# order = 40
# alpha = 0.41
mc = np.apply_along_axis(
pysptk.mcep,
1,
frames,
order,
alpha)
return sample_rate, f0, mc # sample_rate- ?, f0, mel-cepstrum coefs
def cal_prosody(self,wav):
# Input: wav: audio signal in numpy.array format
# Output: proso: Tensor, [max_frames, 4]
max_frames = wav.shape[0] // self.hop
f0 = pysptk.swipe(wav.astype(np.float64),
fs=self.sr, hopsize=self.hop,
min=self.f0_min,
max=self.f0_max,
otype='f0')
lf0 = np.log(f0 + 1e-10)
lf0, uv = self.interpolation(lf0, -1)
lf0 = torch.tensor(lf0.astype(np.float32)).unsqueeze(0)[:, :max_frames]# (1,num_frame)
uv = torch.tensor(uv.astype(np.float32)).unsqueeze(0)[:, :max_frames]
if torch.sum(uv) == 0:
# if frame is completely unvoiced, make lf0 min val
lf0 = torch.ones(uv.size()) * np.log(self.f0_min)
assert lf0.min() > 0, lf0.data.numpy()
# secondly obtain zcr
zcr = librosa.feature.zero_crossing_rate(y=wav,
frame_length=self.win,
hop_length=self.hop)
zcr = torch.tensor(zcr.astype(np.float32))
zcr = zcr[:, :max_frames]
# finally obtain energy
egy = librosa.feature.rms(y=wav, frame_length=self.win,
hop_length=self.hop,
pad_mode='constant')
egy = torch.tensor(egy.astype(np.float32))
egy = egy[:, :max_frames]
proso = torch.cat((lf0, uv, egy, zcr), dim=0).unsqueeze(0)#(1,4,num_frame)
return proso
def pysptk_featurize(audiofile):
labels = list()
features = list()
fs, x = wavfile.read(audiofile)
f0_swipe = pysptk.swipe(x.astype(np.float64),
fs=fs,
hopsize=80,
min=60,
max=200,
otype="f0")
features = features + stats(f0_swipe)
labels = stats_labels('f0_swipe', labels)
f0_rapt = pysptk.rapt(x.astype(np.float32),
fs=fs,
hopsize=80,
min=60,
max=200,
otype="f0")
features = features + stats(f0_rapt)
labels = stats_labels('f0_rapt', labels)
mgc = pysptk.mgcep(xw, 20, 0.0, 0.0)
features = features + stats(mgc)
labels = stats_labels('mel-spectrum envelope', labels)
return features, labels
def test_swipe_regression():
# Grund truth data is generated by:
#
# $ wav2raw pysptk/example_audio_data/arctic_a0007.wav
#
# $ x2x +sf ./pysptk/example_audio_data/arctic_a0007.raw | \
# pitch -a 1 -s 16 -p 80 -L 60 -H 240 -o 0 > \
# arctic_a007_p16_L60_H240_o0_swipe.pitch
#
# $ dmp +f arctic_a007_p16_L60_H240_o0_swuoe.pitch | awk '{print $2}' >\
# arctic_a007_p16_L60_H240_o0_swipe.txt
#
# $ pitch -h
# ...
#
# SPTK: version 3.10
# CVS Info: $Id: pitch.c,v 1.53 2016/12/25 05:00:19 uratec Exp $
ground_truth_path = join(
dirname(__file__), "data", "arctic_a007_p16_L60_H240_o0_swipe.txt"
)
with open(ground_truth_path) as f:
ground_truth = np.asarray([float(s) for s in [line for line in f.readlines()]])
ground_truth = ground_truth.astype(np.float32)
fs, x = wavfile.read(pysptk.util.example_audio_file())
assert fs == 16000
# Since SPTK might have memory corruption bug and the result might be
# non-deterministic, test it with multiple time...
for _ in range(5):
f0 = pysptk.swipe(
x.astype(np.float64), fs=fs, hopsize=80, min=60, max=240, otype=0
)
assert np.allclose(ground_truth, f0)
def source_excitation_generation(np_data, rate):
pitch = ps.swipe(np_data.astype(np.float64),
fs=rate,
hopsize=HOP_LENGTH,
min=60,
max=240,
otype="pitch")
source_excitation = ps.excite(pitch, HOP_LENGTH)
return source_excitation
def F0_swipe(
waveform,
hop_length=None,
sr=None,
hop_time=None,
f_min=60, # default in swipe
f_max=240, # default in swipe
threshold=0.5, # custom defualt (0.3 in swipe)
):
if hop_length is not None:
hopsize = hop_length
else:
hopsize = int(sr * hop_time)
if waveform.ndim == 1:
return torch.from_numpy(
swipe(
waveform.contiguous().double().numpy(),
fs=sr,
hopsize=hopsize,
min=f_min,
max=f_max,
threshold=threshold,
otype="f0",
)).float()
elif waveform.ndim == 2: # (B, N)
f0 = []
for audio in waveform:
f0.append(
torch.from_numpy(
swipe(
audio.contiguous().double().numpy(),
fs=sr,
hopsize=hopsize,
min=f_min,
max=f_max,
threshold=threshold,
otype="f0",
)).float())
return torch.stack(f0)
def get_synt_wav(wav_file_path):
# # Synthesis from mel-cepstrum
sample_rate, x = wavfile.read(wav_file_path)
# assert sample_rate == 16000
# al.play(x.astype(float) / x.max(), fs=sample_rate) # Audio(x, rate=sample_rate)
# all of pysptk functions assume input array is C-contiguous
# and np.float4 element type
frames = librosa.util.frame(
x,
frame_length=frame_length,
hop_length=hop_length).astype(np.float64).T
# Windowing
frames *= pysptk.blackman(frame_length)
# assert frames.shape[1] == frame_length
# F0 estimation
f0 = pysptk.swipe(
x.astype(np.float64),
fs=sample_rate,
hopsize=hop_length,
min=50,
max=500)
generator = excite.ExcitePulse(sample_rate, hop_length, False)
source_excitation = generator.gen(f0)
# apply function along with `time` axis (=1)
mc = np.apply_along_axis(
pysptk.mcep,
1,
frames,
order,
alpha)
# Convert mel-cesptrum to MLSADF coefficients
b = np.apply_along_axis(pysptk.mc2b, 1, mc, alpha)
synthesizer = pysptk.synthesis.Synthesizer(
pysptk.synthesis.MLSADF(
order=order, alpha=alpha),
hop_length)
x_synthesized = synthesizer.synthesis(source_excitation, b)
# Audio(x_synthesized, rate=sample_rate)
# al.play(x_synthesized.astype(float) / x_synthesized.max(), fs=sample_rate)
return x_synthesized
def pitch_detect(sndarray, fs, chunk_size):
"""
pitch_detect(sndarray,fs, chunk_size)
pitch_detect computes the fundamental frequency/pitches of blocks/ of Chunks
Parameters:sndarray - Discrete Data
fs -Sampling frequency
chunk_size
Returns f0
"""
new_sndarray = numpy.asarray(numpy.float64(sndarray))
f0 = pysptk.swipe(numpy.asarray(new_sndarray), fs, chunk_size, 65, 500,
0.001, 1)
return f0
def __call__(self, pkg, cached_file=None):
pkg = format_package(pkg)
wav = pkg['chunk']
wav = wav.data.numpy()
max_frames = wav.shape[0] // self.hop
if cached_file is not None:
# load pre-computed data
proso = torch.load(cached_file)
beg_i = pkg['chunk_beg_i'] // self.hop
end_i = pkg['chunk_end_i'] // self.hop
proso = proso[:, beg_i:end_i]
pkg['prosody'] = proso
else:
# first compute logF0 and voiced/unvoiced flag
f0 = pysptk.swipe(wav.astype(np.float64),
fs=self.sr,
hopsize=self.hop,
min=self.f0_min,
max=self.f0_max,
otype='f0')
lf0 = np.log(f0 + 1e-10)
lf0, uv = interpolation(lf0, -1)
lf0 = torch.tensor(lf0.astype(
np.float32)).unsqueeze(0)[:, :max_frames]
uv = torch.tensor(uv.astype(
np.float32)).unsqueeze(0)[:, :max_frames]
if torch.sum(uv) == 0:
# if frame is completely unvoiced, make lf0 min val
lf0 = torch.ones(uv.size()) * np.log(self.f0_min)
assert lf0.min() > 0, lf0.data.numpy()
# secondly obtain zcr
zcr = librosa.feature.zero_crossing_rate(y=wav,
frame_length=self.win,
hop_length=self.hop)
zcr = torch.tensor(zcr.astype(np.float32))
zcr = zcr[:, :max_frames]
# finally obtain energy
egy = librosa.feature.rmse(y=wav,
frame_length=self.win,
hop_length=self.hop,
pad_mode='constant')
egy = torch.tensor(egy.astype(np.float32))
egy = egy[:, :max_frames]
proso = torch.cat((lf0, uv, egy, zcr), dim=0)
pkg['prosody'] = proso
return pkg
def pysptk_features(x):
import pysptk
wav_max = 2**15-1
x = (x * wav_max).astype(np.float64)
frame_length = 512
hop_length = 160
frames = librosa.util.frame(x, frame_length=frame_length, hop_length=hop_length).T
frames *= pysptk.blackman(frame_length)
order = 25 # seems to be pretty standard, results in 26 values
alpha = 0.42 # this value is best for 16kHz sampling according to docs http://ftp.jaist.ac.jp/pub/pkgsrc/distfiles/SPTKref-3.9.pdf
mcep = pysptk.mcep(frames, order, alpha)
f0 = pysptk.swipe(x, fs=16000, hopsize=hop_length, min=60, max=240, otype="f0")
f0 = f0[1:1+mcep.shape[0]] # cut off ends to match mcep lengths
return np.concatenate([f0[:,np.newaxis], mcep], 1).astype(np.float32)
def process_audio(self, x):
pitch = pysptk.swipe(x, fs=self.sr, hopsize=self.hop_length, min=self.f0_floor, max=self.f0_ceil, otype="pitch")
f0, timeaxis = pyworld.dio(x, fs=self.sr, f0_floor=self.f0_floor, f0_ceil=self.f0_ceil, frame_period=self.frame_period)
f0 = pyworld.stonemask(x, f0, timeaxis, self.sr)
# x_frame = self.samples_to_frames(x)
if self.use_mel:
mel = librosa.feature.melspectrogram(x, sr=self.sr, n_fft=self.fft_length, hop_length=self.hop_length)
mfcc = librosa.feature.mfcc(S=mel, sr=self.sr, n_mfcc=self.n_mfcc)
if self.norm_mfcc:
mfcc = self.standardize_mfcc(mfcc)
return {'mel': mel.T, 'mfcc': mfcc.T, 'f0': f0, 'pitch': pitch}
else:
ap = pyworld.d4c(x, f0, timeaxis, self.sr, fft_size=self.fft_length) # Aperiodicity
sp = pyworld.cheaptrick(x, f0, timeaxis, self.sr,
fft_size=self.fft_length)
return {'sp': sp, 'ap': ap, 'f0': f0, 'pitch': pitch}
def get_MCEP(self, utterance):
utterance = librosa.util.normalize(utterance)
utterance = utterance + np.random.normal(
loc=0, scale=0.0000001, size=utterance.shape[0])
utterance = librosa.util.normalize(utterance)
utterance = utterance.astype(np.float64) # necessary for synthesizer
frames = librosa.util.frame(utterance,
frame_length=self.frame_length,
hop_length=self.hop_length).astype(
np.float64).T
# Windowing
frames *= pysptk.blackman(self.frame_length)
assert frames.shape[1] == self.frame_length
# Pitch
pitch = pysptk.swipe(utterance.astype(np.float64),
fs=self.sr,
hopsize=self.hop_length,
min=60,
max=240,
otype="pitch")
mcep = pysptk.mcep(frames, self.order, self.alpha)
return mcep, pitch
f.write("order,time\n")
for order in (0, 4, 9, 14, 24, 49):
start = time.time()
# Note that almost all of pysptk functions assume input array is C-contiguous and np.float64 element type
frames = librosa.util.frame(x,
frame_length=frame_length,
hop_length=hop_length).astype(np.float64).T
# Windowing
frames *= pysptk.blackman(frame_length)
assert frames.shape[1] == frame_length
pitch = pysptk.swipe(x.astype(np.float64),
fs=sr,
hopsize=hop_length,
min=60,
max=240,
otype="pitch")
source_excitation = pysptk.excite(pitch, hop_length)
# Order of mel-cepstrum
mc = pysptk.mcep(frames, order, alpha)
logH = pysptk.mgc2sp(mc, alpha, 0.0, frame_length).real
print(mc.shape)
#plt.plot(mc)
#plotname="x_syn_coefs_" + str(order) + ".png"
#plt.savefig(plotname)
# Convert mel-cesptrum to MLSADF coefficients
b = pysptk.mc2b(mc, alpha)
def __test(x, fs, hopsize, otype):
pysptk.swipe(x, fs, hopsize, otype=otype)
def __test(x, fs, hopsize, otype):
f0 = pysptk.swipe(x, fs, hopsize, otype=otype)
assert np.all(np.isfinite(f0))
if otype == 1:
assert np.all(f0 >= 0)
OUT_WAVE_FILE = "out.wav" # 分析再合成した音声
# 音声の読み込み
fs, x = wavfile.read(IN_WAVE_FILE)
x = x.astype(np.float64)
# 音声の切り出しと窓掛け
frames = librosa.util.frame(x,
frame_length=FRAME_LENGTH,
hop_length=HOP_LENGTH).astype(np.float64).T
frames *= pysptk.blackman(FRAME_LENGTH) # 窓掛け（ブラックマン窓）
# ピッチ抽出
pitch = pysptk.swipe(x,
fs=fs,
hopsize=HOP_LENGTH,
min=MIN_F0,
max=MAX_F0,
otype="pitch")
# 励振源信号(声帯音源)の生成
source_excitation = pysptk.excite(pitch, HOP_LENGTH)
# 線形予測分析による線形予測符号化(LPC)係数の抽出
lpc = pysptk.lpc(frames, ORDER)
lpc[:, 0] = np.log(lpc[:, 0])
# LPC係数をPARCOR係数に変換
parcor = pysptk.lpc2par(lpc)
# 全極フィルタの作成
synthesizer = Synthesizer(AllPoleLatticeDF(order=ORDER), HOP_LENGTH)
def get_target_feats(utterance, utterance_wav, alignments):
'''Generates target features for an utterance (in this case duration,
initial phone fundamental frequency, final phone fundamental frequency, and
energy'''
#phone_start = int(alignments[0] * fs)
first_phone_start = alignments[0][0]
first_phone_end = alignments[0][1]
#print("START: " + str(phone_start))
#phone_end = int(alignments[1] * fs)
#print("END: " + str(phone_end))
last_phone_start = alignments[-1][0]
last_phone_end = alignments[-1][1]
#print(phone_start)
#print(phone_end)
#print(utterance_wav)
#print(len(utterance_wav))
duration = last_phone_end - first_phone_start
first_phone_samples = utterance_wav[first_phone_start:first_phone_end]
last_phone_samples = utterance_wav[last_phone_start:last_phone_end]
all_phone_samples = utterance_wav[first_phone_start:last_phone_end]
#phone_test = utterance_wav[phone_start]
#try:
'''
print(first_phone_start)
print(first_phone_end)
print(last_phone_start)
print(last_phone_end)
'''
try:
f_0_init = np.mean(pysptk.swipe(first_phone_samples.astype(np.float64),
fs=fs,
hopsize=100,
otype='f0'))
#print(f_0_init)
f_0_end = np.mean(pysptk.swipe(last_phone_samples.astype(np.float64),
fs=fs,
hopsize=100,
otype='f0'))
except:
print("Outof bounds!!")
print(utterance)
print(first_phone_start)
print(first_phone_end)
print(last_phone_start)
print(last_phone_end)
print(alignments)
raise Exception("Out of bounds")
return (0, 0, 0, 0)
#print(f_0_end)
#except IndexError:
# For "Index Error: Out of bounds on buffer access (axis 0)
#mfcc = pysptk.mfcc(samples)
#pitch = pysptk.swipe(phone_samples.astype(np.float64), fs=fs, hopsize=100, otype='pitch')
#excitation = pysptk.excite(pitch)
#excitation_mu = np.mean(excitation)
#excitation_std = np.std(excitation)
#print()
energy = np.sum(np.square(all_phone_samples)) / duration
return duration, f_0_init, f_0_end, energy
def eval_pitch(ted_audio_path, user_audio_path, png_save_path):
sr, x = wavfile.read(ted_audio_path) # ted 목소리
assert sr == 16000
x = x.astype(np.float64)
frame_length = 1024
hop_length = 80
f_you = pysptk.swipe(x.astype(np.float64),
fs=sr,
hopsize=hop_length,
min=60,
max=240,
otype="f0")
sr1, x1 = wavfile.read(user_audio_path)
assert sr1 == 16000
x1 = x1.astype(np.float64)
frame_length = 1024
hop_length = 80
# F0 estimation # 각 주파수에서 기본 주파수 뽑기 ,def strength 와 같은 과정.
f_ted = pysptk.swipe(x1.astype(np.float64),
fs=sr1,
hopsize=hop_length,
min=60,
max=240,
otype="f0")
plt.figure(figsize=(20, 5))
##############
width = int(len(f_ted) / 22) # width 조정 해주기
width1 = int(len(f_you) / 22)
if np.where(f_you >= 60)[0][0] > np.where(
f_ted >= 60)[0][0]: #테드가 왼쪽에서 더 빠르게 시작하면?
diff = np.where(f_you >= 60)[0][0] - np.where(f_ted >= 60)[0][0]
zero_ = np.zeros(diff)
new_f0 = np.r_[zero_, f_ted]
cal0 = copy.copy(new_f0)
cal0[np.where(cal0 < 75)] = 0
new_f0[np.where(new_f0 < 75)] = 0
value = []
loc = []
c3 = 0
i = 0
new_f0 = list(new_f0)
while i < len(range(len(new_f0))):
ten = new_f0[i:i + width]
if ten == []:
break
a = max(ten)
b = ten.index(a)
value.append(a)
loc.append(i + b)
if i + width > len(new_f0):
break
else:
c3 = c3 + 1
i = width * c3
base = np.empty(len(new_f0))
base.fill(np.nan)
for i in range(len(loc)):
location = loc[i]
base[location] = value[i]
df_blue = pd.DataFrame(base)
plt.figure(figsize=(20, 5))
df_blue.interpolate(method='polynomial',
order=2,
linewidth=2,
inplace=True)
bbb = list(df_blue[0])
ccc = list(map(lambda x: 0 if x < 0 else x, bbb))
df_blue[0] = ccc
df_blue.fillna(0, inplace=True)
for g in range(len(df_blue)):
if df_blue[0][g] > max(f_ted) * 1.2:
df_blue[0][g] = max(f_ted) * 1.2
#합친게 you0
you0 = f_you
you0[np.where(you0 < 75)] = 0
value1 = []
loc1 = []
c1 = 0
i1 = 0
you0 = list(you0)
while i < len(range(len(you0))):
ten1 = you0[i1:i1 + width1]
if ten1 == []:
break
a1 = max(ten1)
b1 = ten1.index(a1)
value1.append(a1)
loc1.append(i1 + b1)
if i1 + width1 > len(you0):
break
else:
c1 = c1 + 1
i1 = width1 * c1
base1 = np.empty(len(you0))
base1.fill(np.nan)
for i in range(len(loc1)):
location1 = loc1[i]
base1[location1] = value1[i]
df_red = pd.DataFrame(base1)
df_red.interpolate(method='polynomial',
order=2,
linewidth=2,
inplace=True)
bbb1 = list(df_red[0])
ccc1 = list(map(lambda x: 0 if x < 0 else x, bbb1))
df_red[0] = ccc1
df_red.fillna(0, inplace=True)
for h in range(len(df_red)):
if df_red[0][h] > max(f_you) * 1.2:
df_red[0][h] = max(f_you) * 1.2
df_red[0] = df_red[0] * max(f_ted) / max(f_you)
area = []
diff_areas = []
if len(df_red[0]) > len(df_blue[0]):
for i in range(diff, len(df_blue[0])):
area.append(df_blue[0][i])
diff_areas.append(abs(df_red[0][i] - df_blue[0][i]))
result = 1 - (sum(diff_areas / sum(area)))
else:
for i in range(diff, len(df_red[0])):
area.append(df_blue[0][i])
diff_areas.append(abs(df_red[0][i] - df_blue[0][i]))
result = 1 - (sum(diff_areas) / sum(area))
ranks = []
for i in range(1, len(df_blue[0]) - 1, 1):
if df_blue[0][i] > 60:
if df_blue[0][i] > df_blue[0][
i - 1] and df_blue[0][i] > df_blue[0][i + 1]:
ranks.append(df_blue[0][i] / max(df_blue[0]))
ranks1 = []
for i in range(1, len(df_red[0]) - 1, 1):
if df_red[0][i] > 60:
if df_red[0][i] > df_red[0][
i - 1] and df_red[0][i] > df_red[0][i + 1]:
ranks1.append(df_red[0][i] / max(df_red[0]))
diffrent = []
if len(ranks) > len(ranks1):
for i in range(len(ranks1)):
diffrent.append(abs(ranks1[i] - ranks[i]))
else:
for i in range(len(ranks)):
diffrent.append(abs(ranks1[i] - ranks[i]))
points4 = 1 - (sum(diffrent) / sum(ranks))
else: # 테드가 더늦겟시작
diff = np.where(f_ted >= 60)[0][0] - np.where(f_you >= 60)[0][0]
zero_ted = np.zeros(diff)
new_ted = np.r_[zero_ted, f_you]
new_ted[np.where(new_ted < 75)] = 0
value1 = []
loc1 = []
c1 = 0
i1 = 0
new_ted = list(new_ted)
while i1 < len(range(len(new_ted))):
ten1 = new_ted[i1:i1 + width1]
if ten1 == []:
break
a1 = max(ten1)
b1 = ten1.index(a1)
value1.append(a1)
loc1.append(i1 + b1)
if i1 + width1 > len(new_ted):
break
else:
c1 = c1 + 1
i1 = width1 * c1
base = np.empty(len(new_ted))
base.fill(np.nan)
for h in range(len(loc1)):
location1 = loc1[h]
base[location1] = value1[h]
df_blue = pd.DataFrame(base)
df_blue.interpolate(method='polynomial',
order=2,
linewidth=2,
inplace=True)
bbb = list(df_blue[0])
ccc = list(map(lambda x: 0 if x < 0 else x, bbb))
df_blue[0] = ccc
df_blue.fillna(0, inplace=True)
for g in range(len(df_blue)):
if df_blue[0][g] > max(f_you) * 1.2:
df_blue[0][g] = max(f_you) * 1.2
f_ted1 = f_ted
cal2 = copy.copy(f_ted1)
cal2[np.where(f_ted1 < 75)] = 0
f_ted1[np.where(f_ted1 < 75)] = 0
value2 = []
loc2 = []
c2 = 0
i2 = 0
f_ted1 = list(f_ted1)
while i2 < len(range(len(f_ted1))):
ten2 = f_ted1[i2:i2 + width]
if ten2 == []:
break
a2 = max(ten2)
b2 = ten2.index(a2)
value2.append(a2)
loc2.append(i2 + b2)
if i2 + width > len(f_ted1):
break
else:
c2 = c2 + 1
i2 = width * c2
base2 = np.empty(len(f_ted1))
base2.fill(np.nan)
for i in range(len(loc2)):
location2 = loc2[i]
base2[location2] = value2[i]
df_red = pd.DataFrame(base2)
df_red.interpolate(method='polynomial',
order=2,
linewidth=2,
inplace=True)
bbb2 = list(df_red[0])
ccc2 = list(map(lambda x: 0 if x < 0 else x, bbb2))
df_red[0] = ccc2
df_red.fillna(0, inplace=True)
for g in range(len(df_red)):
if df_red[0][g] > max(f_ted) * 1.2:
df_red[0][g] = max(f_ted) * 1.2
df_blue[0] = df_blue[0] * max(f_ted) / max(f_you)
area = []
diff_areas = []
if len(df_red[0]) > len(df_blue[0]):
for i in range(diff, len(df_blue[0])):
area.append(df_red[0][i])
diff_areas.append(abs(df_red[0][i] - df_blue[0][i]))
result = 1 - (sum(diff_areas / sum(area)))
else:
for i in range(diff, len(df_red[0])):
area.append(df_red[0][i])
diff_areas.append(abs(df_red[0][i] - df_blue[0][i]))
result = 1 - (sum(diff_areas) / sum(area))
ranks = []
for i in range(1, len(df_blue[0]) - 1, 1):
if df_blue[0][i] > 60:
if df_blue[0][i] > df_blue[0][
i - 1] and df_blue[0][i] > df_blue[0][i + 1]:
ranks.append(df_blue[0][i] / max(df_blue[0]))
ranks1 = []
for i in range(1, len(df_red[0]) - 1, 1):
if df_red[0][i] > 60:
if df_red[0][i] > df_red[0][
i - 1] and df_red[0][i] > df_red[0][i + 1]:
ranks1.append(df_red[0][i] / max(df_red[0]))
diffrent = []
if len(ranks) > len(ranks1):
for i in range(len(ranks1)):
diffrent.append(abs(ranks1[i] - ranks[i]))
else:
for i in range(len(ranks)):
diffrent.append(abs(ranks1[i] - ranks[i]))
points4 = 1 - (sum(diffrent) / sum(ranks1))
result = int(result * 100)
result1 = int(points4 * 100)
pitch_result_rate = max(result, result1)
global pitch_result
if pitch_result_rate >= 85:
pitch_result = 'Excellent'
elif pitch_result_rate >= 65:
pitch_result = 'Good'
else:
pitch_result = 'Bad'
line1, = plt.plot(df_blue, color='navy', linewidth=5)
line2, = plt.plot(df_red, color='crimson', linewidth=5)
plt.title('Pitch Result', fontsize=50)
plt.legend(handles=(line1, line2), labels=('Ted', 'You'), fontsize=20)
plt.ylabel('Pitch', fontsize=20)
plt.tick_params(axis='x',
which='both',
bottom=False,
top=False,
labelbottom=False)
plt.savefig(png_save_path + 'pitch_result.png')
return pitch_result_rate, pitch_result
def __test(x, fs, hopsize, otype):
f0 = pysptk.swipe(x, fs, hopsize, otype=otype)
assert np.all(np.isfinite(f0))
def __call__(self, x):
pitch = pysptk.swipe(x, fs=self.sr, hopsize=self.hop_length,
min=self.f0_floor, max=self.f0_ceil, otype="pitch")
f0 = pysptk.swipe(x, fs=self.sr, hopsize=self.hop_length,
min=self.f0_floor, max=self.f0_ceil, otype="f0")
return pitch, f0
def pitch_detect(sndarray,fs, chunk_size):
new_sndarray = numpy.asarray(numpy.float64(sndarray))
f0 = pysptk.swipe(numpy.asarray(new_sndarray), fs, chunk_size, 65,500,0.001,1)
# Fundamental Frequency
return f0
duration = 0
file = ""
if (i <= 9):
file = dir_path + "chunks/chunk-0" + str(i) + ".wav"
else:
file = dir_path + "chunks/chunk-" + str(i) + ".wav"
duration = get_duration(file)
fs, x = wavfile.read(file)
assert fs == 16000
f0_swipe = pysptk.swipe(x.astype(np.float64),
fs=fs,
hopsize=80,
min=60,
otype="f0")
a = []
f = []
X_Frequecies_Vector = []
for w in f0_swipe:
if w != 0:
f.append(w)
if len(f) >= 30:
f = random.sample(f, 30)
else:
f += [0] * (30 - len(f))
freq_matrix.append(f)
#a = np.var(f)
def processingVideo(): for i in range(66,67): dir_path = "/home/eduardo/data_base_www/2PhaT6AbH3Q/" #move_files() num_emphasys = [] n_chunks = len(glob.glob(dir_path +"chunks/chunk*")) freq_matrix = [[0 for i in range(2)] for j in range(n_chunks)] for i in range(0, n_chunks): duration = 0 file ="" if(i <=9): file = dir_path+"chunks/chunk-0"+str(i)+".wav" else: file = dir_path+"chunks/chunk-"+str(i)+".wav" duration = get_duration(file) fs, x = wavfile.read(file) assert fs == 16000 f0_swipe = pysptk.swipe(x.astype(np.float64), fs = fs, hopsize = 20, min=60) a = [] f = [] X_Frequecies_Vector = [] for w in f0_swipe: if w != 0: f.append(w) #if not f: # a = 0 #else: # a = stats.mode(f)[0][0] a = np.mean(f) #print(len(f)) c, Pxx_den = signal.welch(x, fs, nperseg=1024) #if(Pxx_den.any()): # v = 0 ## v = stats.mode(Pxx_den)[0][0] v = np.mean(Pxx_den) if(~np.isnan(v) and ~np.isnan(a)): #num_emphasys.append ( v ) l = [] l.append(a) l.append(v) freq_matrix[i] = l weightA = [0 for i in range(n_chunks)] weightT = [0 for i in range(n_chunks)] transcript_matrix, annotation_matrix, avg_depth = cs.computeMatrix(dir_path, n_chunks) best_model_silhouettte = -1000 iterations_without_improvment = 0 max = 32 model2 = cluster.SpectralClustering(max, affinity='precomputed', n_init=10000, n_jobs=-1) cluster_labels = model2.fit_predict(transcript_matrix) for i in range(len(cluster_labels)-1): if(cluster_labels[i] != cluster_labels[i+1]): weightT[i+1] = np.sqrt(pow(freq_matrix[i+1][0],2) + pow(freq_matrix[i+1][1],2)) for j in range(len(annotation_matrix) -1): if(not set(annotation_matrix[j]).intersection(annotation_matrix[j+1])): weightA[j+1] = float(np.sqrt(pow(freq_matrix[j+1][0],2) + pow(freq_matrix[j+1][1],2)) /abs(avg_depth[j] - avg_depth[j+1])) #rankingT = sorted(range(len(weightT)), key=lambda x: weightT[x])[-70:] rankingA = sorted(range(len(weightA)), key=lambda x: weightA[x])[-70:] #ranking = list(set(rankingT).intersection(rankingA)) #merged = sorted(list(set(ranking).union(rankingA))) #matrixT = np.array(transcript_matrix) #matrixA = np.array(annotation_matrix) #ranking = getranking(n_chunks, freq_matrix, matrixT, matrixA, 25) evaluate_method.evaluate(dir_path, sorted(rankingA), "aas")
def get_pitch_pysptk(wav):
sample_rate, samples = wavfile.read(wav)
f0_swipe = pysptk.swipe(samples.astype(np.float64), fs=sample_rate, hopsize=80, min=60, max=200, otype="f0")
return f0_swipe
def read_audio_n_process(file, label, base_path, sampling_rate,
sample_size_in_seconds, overlap, normalise, method):
"""
This method is called by the preprocess data method
:param file:
:param label:
:param base_path:
:param sampling_rate:
:param sample_size_in_seconds:
:param overlap:
:param normalise:
:return:
"""
data, out_labels = [], []
filepath = base_path + file
if os.path.exists(filepath):
audio, sr = librosa.load(filepath, sr=sampling_rate)
# mask = envelope(audio, sr, 0.0005)
# audio = audio[mask]
sr = sampling_rate
# audio = remove_silent_parts(filepath, sr=sampling_rate)
chunks = cut_audio(audio,
sampling_rate=sr,
sample_size_in_seconds=sample_size_in_seconds,
overlap=overlap)
for chunk in chunks:
if method == 'fbank':
zero_crossing = librosa.feature.zero_crossing_rate(chunk)
f0 = pysptk.swipe(chunk.astype(np.float64),
fs=sr,
hopsize=510,
min=60,
max=240,
otype="f0").reshape(1, -1)
pitch = pysptk.swipe(chunk.astype(np.float64),
fs=sr,
hopsize=510,
min=60,
max=240,
otype="pitch").reshape(1, -1)
f0_pitch_multiplier = 1
features = mel_filters(chunk, sr, normalise)
f0 = np.reshape(f0[:, :features.shape[1] *
f0_pitch_multiplier],
newshape=(f0_pitch_multiplier, -1))
pitch = np.reshape(pitch[:, :features.shape[1] *
f0_pitch_multiplier],
newshape=(f0_pitch_multiplier, -1))
# shimmer_jitter = get_shimmer_jitter_from_opensmile(chunk, time.time(), sr)
# shimmer_jitter = np.tile(shimmer_jitter, math.ceil(features.shape[-1] / len(shimmer_jitter)))[
# :features.shape[
# -1]] # Repeating the values to match the features length of filterbanks
# shimmer_jitter = np.reshape(shimmer_jitter, newshape=(1, -1))
features = np.concatenate((features, zero_crossing, f0, pitch),
axis=0) # shimmer_jitter
elif method == 'mfcc':
features = mfcc_features(chunk, sr, normalise)
elif method == 'gaf':
features = gaf(chunk)
elif method == 'raw':
features = chunk
else:
raise Exception(
'Specify a method to use for pre processing raw audio signal. Available options - {fbank, mfcc, gaf, raw}'
)
data.append(features)
out_labels.append(float(label))
return data, out_labels, chunks
else:
print('File not found ', filepath)
return [], [], []
def get_f0(waveform,
sample_rate,
hop_length_seconds=0.01,
method='swipe',
f0_min=60,
f0_max=300):
"""Compute the F0 contour using PYSPTK: https://github.com/r9y9/pysptk/.
Args:
waveform (np.array, [T, ]): waveform over which to compute f0
sample_rate (int > 0): number of samples per second in waveform
hop_length (int): hop size argument in pysptk.swipe. Corresponds to hopsize
in the window sliding of the computation of f0.
method (str): is one of 'swipe' or 'rapt'. Define which method to use for f0
calculation. See https://github.com/r9y9/pysptk
Returns:
dict: Dictionary containing keys:
"contour" (np.array, [1, t1]): f0 contour of waveform. Contains unvoiced
frames.
"values" (np.array, [1, t2]): nonzero f0 values waveform. Note that this
discards all unvoiced frames. Use to compute mean, std, and other statistics.
"mean" (float): mean of the f0 contour.
"std" (float): standard deviation of the f0 contour.
"""
assert method in (
'swipe',
'rapt'), "The method argument should be one of 'swipe' or 'rapt'."
hop_length = numseconds_to_numsamples(hop_length_seconds, sample_rate)
if method == 'swipe':
f0_contour = swipe(
waveform.astype(np.float64),
fs=sample_rate,
hopsize=hop_length,
min=f0_min,
max=f0_max,
otype="f0",
)[np.newaxis, :]
elif method == 'rapt':
# For this estimation, waveform needs to be in the int PCM format.
f0_contour = rapt(
np.round(waveform * 32767).astype(np.float32),
fs=sample_rate,
hopsize=hop_length,
min=f0_min,
max=f0_max,
otype="f0",
)[np.newaxis, :]
# Remove unvoiced frames.
f0_values = f0_contour[:, np.where(f0_contour[0, :] != 0)][0]
f0_mean = np.mean(f0_values[0])
f0_std = np.std(f0_values[0])
return {
"contour": f0_contour,
"values": f0_values,
"mean": f0_mean,
"std": f0_std,
}
def __call__(self, tensor):
"""
Args:
tensor (Tensor): Tensor of audio of size (samples x 1)
"""
# pysptk and interpolate are a MUST in this transform
import pysptk
from ahoproc_tools.interpolate import interpolation
t_npy = tensor.cpu().squeeze(1).numpy()
#print('t_npy shape: ', t_npy.shape)
seqlen = t_npy.shape[0]
T = seqlen // self.hop_length
# compute LF0 and UV
f0 = pysptk.swipe(t_npy.astype(np.float64),
fs=self.sr,
hopsize=self.hop_length,
min=60,
max=240,
otype="f0")[:T]
lf0 = np.log(f0 + 1e-10)
lf0, uv = interpolation(lf0, -1)
if np.any(lf0 == np.log(1e-10)):
# all lf0 goes to minf0 as a PAD symbol
lf0 = np.ones(lf0.shape) * np.log(60)
# all frames are unvoiced
uv = np.zeros(uv.shape)
ret = {
'lf0': torch.FloatTensor(lf0).view(-1, 1),
'uv': torch.FloatTensor(uv.astype(np.float32)).view(-1, 1)
}
tot_frames = T
# MelSpectrum and MFCCs
mel = self.mel(tensor).transpose(0, 1).squeeze(2)
# do compression?
if self.dynamic_norm_spec:
mel = torch.log1p(mel * 10000) / torch.log(torch.FloatTensor([10]))
ret['mel_spec'] = mel[:tot_frames]
mfcc = librosa.feature.mfcc(y=t_npy,
sr=self.sr,
n_fft=self.n_fft,
hop_length=self.hop_length,
n_mfcc=self.mfcc_order).T
mfcc = mfcc[:tot_frames]
ret['mfcc'] = torch.FloatTensor(mfcc)
# Spectrogram abs magnitude [dB]
spec = librosa.stft(t_npy,
n_fft=self.n_fft,
hop_length=self.hop_length,
win_length=self.win_length,
window=self.window)
spec_db = librosa.amplitude_to_db(spec).T
spec_ang = np.angle(spec).T
spec_db = spec_db[:tot_frames]
spec_ang = spec_ang[:tot_frames]
ret['mag'] = torch.FloatTensor(spec_db)
ret['pha'] = torch.FloatTensor(spec_ang)
# ZCR, E and lF0
egy = librosa.feature.rmse(y=t_npy,
frame_length=self.win_length,
hop_length=self.hop_length,
pad_mode='constant').T
egy = egy[:tot_frames]
zcr = librosa.feature.zero_crossing_rate(y=t_npy,
frame_length=self.win_length,
hop_length=self.hop_length).T
zcr = zcr[:tot_frames]
ret['egy'] = torch.FloatTensor(egy)
ret['zcr'] = torch.FloatTensor(zcr)
ntensor = tensor.clone()
if hasattr(self, 'chopper'):
do_chop = random.random() > 0.5
if do_chop:
ntensor = self.chopper(ntensor, self.sr)
if hasattr(self, 'additive'):
do_add = random.random() > 0.5
if do_add:
ntensor = self.additive(ntensor.numpy(), self.sr)
if hasattr(self, 'clipping'):
do_clip = random.random() > 0.5
if do_clip:
ntensor = self.clipping(ntensor.numpy())
ret['wav'] = ntensor.view((-1, 1))
ret['cwav'] = tensor.view((-1, 1))
return ret