def test_save(tmpdir, audio):
p = str(tmpdir.join('test.wav'))
audio.save(p)
# cannot overwrite an existing file
with pytest.raises(ValueError) as err:
audio.save(p)
assert 'file already exist' in str(err)
audio2 = Audio.load(p)
assert audio == audio2
# test with float32 wav
signal = np.zeros((1000,), dtype=np.float32)
signal[10] = 1.0
signal[20] = -1.0
p = str(tmpdir.join('test2.wav'))
audio = Audio(signal, 1000)
audio.save(p)
meta = Audio.scan(p)
assert meta.nsamples == 1000
assert meta.nchannels == 1
audio2 = Audio.load(p)
assert audio2 == audio
assert audio2.data.min() == -1.0
assert audio2.data.max() == 1.0
def extract_features_shennong(audio_path, save_path):
audio = Audio.load(audio_path)
# 80-dim fbank with 1-dim energe
processor = FilterbankProcessor(sample_rate=audio.sample_rate,
num_bins=40,
use_energy=False) #80 fbank + 1 energy
fbank = processor.process(audio)
fbank = fbank.data #(fbank.data - fbank.data.mean()) / fbank.data.std()
# 3-dim pitch
processor = PitchProcessor(frame_shift=0.01, frame_length=0.025)
options = {
'sample_rate': audio.sample_rate,
'frame_shift': 0.01,
'frame_length': 0.025,
'min_f0': 20,
'max_f0': 500
}
processor = PitchProcessor(**options)
pitch = processor.process(audio)
postprocessor = PitchPostProcessor() # use default options
postpitch = postprocessor.process(pitch) # 3 dim
postpitch = postpitch.data #(postpitch.data - postpitch.data.mean()) / postpitch.data.std()
#features = postpitch
shape = min(fbank.shape[0], postpitch.shape[0])
#zero = np.zeros((,content[i].shape[1]),dtype=np.float32)
#content[i] = np.vstack((content[i],zero))
features = np.concatenate((fbank[:shape, :], postpitch[:shape, :]),
axis=-1)
# name = os.path.basename(audio_path).split('.')[0] + '.npy'
# np.save(os.path.join(save_path, name), features.data)
return features
def wavs_to_feats_df(wavs_list, feats):
assert feats in ['mfcc', 'bnf'], "Unknown feature parameter for wavs_to_feats_df function: {}".format(feats)
feats_list = []
for wav_file in wavs_list:
wav_data = Audio.load(wav_file).resample(8000)
assert wav_data.sample_rate == 8000, "Error. Could not resample file to 8000 Hz for MFCC/BNF feature extraction."
assert wav_data.nchannels == 1, "Unexpected non-mono file supplied: {}".format(filename)
if feats == 'mfcc':
mfcc_data = mfcc_processor.process(wav_data)
mfcc_data = delta_processor.process(mfcc_data)
feats_list.append(mfcc_data.data)
elif feats == 'bnf':
bnf_data = bnf_processor.process(wav_data)
feats_list.append(bnf_data.data)
feats_df = pd.DataFrame({
"filename" : [ os.path.splitext(os.path.basename(f))[0] for f in wavs_list ], # '.../filename.wav' => 'filename',
"features" : feats_list
})
return feats_df
def test_check_wavs_bad(wav_file, wav_file_8k, tmpdir, capsys):
def fun(utts):
c = pipeline._init_config(
pipeline.get_default_config('mfcc', with_cmvn=False))
u = pipeline._init_utterances(utts)
pipeline._Manager(c, u)
return u
# build a stereo file and make sure it is not supported by the
# pipeline
audio = Audio.load(wav_file)
stereo = Audio(np.asarray((audio.data, audio.data)).T,
sample_rate=audio.sample_rate)
assert stereo.nchannels == 2
wav_file_2 = str(tmpdir.join('stereo.wav'))
stereo.save(wav_file_2)
with pytest.raises(ValueError) as err:
fun([(wav_file_2, )])
assert 'all wav files are not mono' in str(err)
# ensure we catch differences in sample rates
capsys.readouterr() # clear buffer
w = [(wav_file, ), (wav_file_8k, )]
out = fun(w)
err = capsys.readouterr().err
assert 'several sample rates found in wav files' in err
assert sorted(out.keys()) == ['utt_1', 'utt_2']
# make sure timestamps are ordered
with pytest.raises(ValueError) as err:
fun([('1', wav_file, 1, 0)])
assert 'timestamps are not in increasing order for' in str(err)
def get_plp_dd(wav_fn, norm):
"""Return the MFCCs with deltas and delta-deltas for a audio file."""
audio = Audio.load(wav_fn)
processor = PlpProcessor(sample_rate=audio.sample_rate, window_type="hamming",frame_length=0.025, frame_shift=0.01,
low_freq=0, vtln_low=60, vtln_high=7200, high_freq=audio.sample_rate/2)
plp_static = processor.process(audio, vtln_warp=1.0)
d_processor = DeltaPostProcessor(order=2)
plp_deltas = d_processor.process(plp_static)
features = np.float64(plp_deltas._to_dict()["data"])
if norm == "cmvn":
features = (features - np.mean(features, axis=0)) / np.std(features, axis=0)
return features
def transform_all_wavs(folder_wav, type, folder_out): # will output [timexdim}
processor = BottleneckProcessor(weights=type)
count = 0
for file in os.listdir(folder_wav):
if count % 500 == 0:
print(count)
count += 1
if not file.endswith('.wav'):
continue
audio = Audio.load(os.path.join(folder_wav, file))
features = processor.process(audio)
#print(features.shape)
#print(features)
np.savetxt(fname = os.path.join(folder_out,file[:-4] + '.csv'), X=features._data)
def test_compare_kaldi(wav_file):
a1 = Audio.load(wav_file).data
with tempfile.NamedTemporaryFile('w+') as tfile:
tfile.write('test {}\n'.format(wav_file))
tfile.seek(0)
with SequentialWaveReader('scp,t:' + tfile.name) as reader:
for key, wave in reader:
a2 = wave.data().numpy()
assert a1.max() == a2.max()
assert a1.min() == a2.min()
assert len(a1) == len(a2.flatten()) == 22713
assert a1.dtype == np.int16 and a2.dtype == np.float32
assert a1.shape == (22713,) and a2.shape == (1, 22713)
assert pytest.approx(a1, a2)
def main():
parser = argparse.ArgumentParser(
description=__doc__,
formatter_class=argparse.RawDescriptionHelpFormatter,
)
parser.add_argument('data_dir', help='input directory with wavs')
parser.add_argument(
'output_dir',
default='/tmp',
nargs='?',
help='output directory (created files are deleted at exit)')
args = parser.parse_args()
# load audio data and compute total duration
audio_data = {
os.path.basename(f): Audio.load(f)
for f in list_files_with_extension(args.data_dir, '.wav')
}
total_duration = datetime.timedelta(
seconds=int(sum(a.duration for a in audio_data.values())))
print('found {} wav files, total duration of {}'.format(
len(audio_data), str(total_duration)))
# compute the features (default MFCC)
print('computing MFCC features...')
t1 = datetime.datetime.now()
processor = MfccProcessor()
features = FeaturesCollection(
**{k: processor.process(v)
for k, v in audio_data.items()})
t2 = datetime.datetime.now()
print('took {}'.format(t2 - t1))
# save the features in all the supported formats
data = {
'duration': total_duration,
'data': {
ext: analyze_serializer(features, ext, args.output_dir)
for ext in supported_extensions().keys()
}
}
print_results(data)
def get_features(sound_file, chosen_processor):
# computes the feature coefficients of a sound file
# :param sound_file : sound file in format .wav
# :type amount: .wav file
# :returns: feature coefficients per frame of 25ms every 10ms can be 'filterbank'
# 'plp', 'rasteplp' or 'bottleneck'
# :rtype: a numpy array
audio = Audio.load(sound_file)
processors = {
'filterbank': FilterbankProcessor(sample_rate=audio.sample_rate),
'plp': PlpProcessor(sample_rate=audio.sample_rate),
'rastaplp': RastaPlpProcessor(sample_rate=audio.sample_rate),
'bottleneck': BottleneckProcessor(weights='BabelMulti')
}
features = chosen_processor.process(audio)
features = pd.DataFrame(features)
return (features)
def get_mfcc_vtln(wav_fn, f, norm, lang):
"""Return the MFCCs with deltas and delta-deltas for a audio file."""
ref = os.path.basename(f).replace(".wav", "")
if not os.path.isfile("warps_{}.pkl".format(lang)):
if os.path.isfile('warps_{}.txt'.format(lang)):
factors = {}
with open('warps_{}.txt'.format(lang), mode='r',
encoding='utf-8') as opfile:
wop = opfile.read().split('\n')
for line in wop:
if len(line) > 1:
l_sp = line.split()
factors[l_sp[0]] = float(l_sp[1])
print(factors)
with open('warps_{}.pkl'.format(lang), mode='wb') as opfile:
pickle.dump(factors, opfile)
else:
print('no warp factors found')
exit()
with open("warps_{}.pkl".format(lang), mode="rb") as op:
factors = pickle.load(op)
warp = float(factors[ref])
audio = Audio.load(wav_fn)
processor = MfccProcessor(sample_rate=audio.sample_rate,
window_type="hamming",
frame_length=0.025,
frame_shift=0.01,
cepstral_lifter=26.0,
low_freq=0,
vtln_low=60,
vtln_high=7200,
high_freq=audio.sample_rate / 2)
d_processor = DeltaPostProcessor(order=2)
mfcc_static = processor.process(audio, vtln_warp=warp)
mfcc_deltas = d_processor.process(mfcc_static)
features = np.float64(mfcc_deltas._to_dict()["data"])
if norm == "cmvn":
features = (features - np.mean(features, axis=0)) / np.std(features,
axis=0)
return features
def main():
parser = argparse.ArgumentParser(description=__doc__)
parser.add_argument('wav', help='wav file to compute features on')
# load the wav file
wav_file = parser.parse_args().wav
audio = Audio.load(wav_file)
# initialize features processors
processors = {
'spectrogram': SpectrogramProcessor(sample_rate=audio.sample_rate),
'filterbank': FilterbankProcessor(sample_rate=audio.sample_rate),
'mfcc': MfccProcessor(sample_rate=audio.sample_rate),
'plp': PlpProcessor(sample_rate=audio.sample_rate),
'rastaplp': RastaPlpProcessor(sample_rate=audio.sample_rate),
'bottleneck': BottleneckProcessor(weights='BabelMulti')}
# compute the features for all processors
features = {k: v.process(audio) for k, v in processors.items()}
# plot the audio signal and the resulting features
fig, axes = plt.subplots(
nrows=len(processors)+1,
gridspec_kw={'top': 0.95, 'bottom': 0.05, 'hspace': 0},
subplot_kw={'xticks': [], 'yticks': []})
time = np.arange(0.0, audio.nsamples) / audio.sample_rate
axes[0].plot(time, audio.astype(np.float32).data)
axes[0].set_xlim(0.0, audio.duration)
axes[0].text(
0.02, 0.8, 'audio',
bbox={'boxstyle': 'round', 'alpha': 0.5, 'color': 'white'},
transform=axes[0].transAxes)
for n, (k, v) in enumerate(features.items(), start=1):
axes[n].imshow(v.data.T, aspect='auto')
axes[n].text(
0.02, 0.8, k,
bbox={'boxstyle': 'round', 'alpha': 0.5, 'color': 'white'},
transform=axes[n].transAxes)
plt.show()
def get_audio(self, utterance):
"""Returns the audio data for that `utterance`"""
utt = self.utterances[utterance]
audio = Audio.load(utt.file)
if utt.tstart is not None:
assert utt.tstop > utt.tstart
audio = audio.segment([(utt.tstart, utt.tstop)])[0]
if self.features == 'bottleneck':
# resample here the signal (this avoid bugs if one part of
# the pipeline on 8k and the other on 16k), then update
# the metadata for the wav to be used by the rest of the
# pipeline
self.log.debug(
'resampling audio from %dHz@%db to %dHz@%db',
audio.sample_rate, audio.dtype.itemsize * 8, 8000, 16)
audio = audio.resample(8000).astype(np.int16)
self._wavs_metadata[self.utterances[utterance].file] = (
Audio._metawav(
audio.nchannels, audio.sample_rate,
audio.nsamples, audio.duration))
return audio
def test_load_badfile():
with pytest.raises(ValueError) as err:
Audio.load('/spam/spam/with/eggs')
assert 'file not found' in str(err)
def audio_8k(wav_file_8k):
return Audio.load(wav_file_8k)
def test_load_notwav():
with pytest.raises(ValueError) as err:
Audio.load(__file__)
assert 'is it a wav?' in str(err)
k = shortest_path_position[0][0]
l = shortest_path_position[1][0]
# divide the shortest distance by the length of the path
average_distance = (distance_matrix[vector_1.shape[0]-1][vector_2.shape[0]-1]) \
/ path_length
return average_distance
all_features = {}
# get bottleneck features of all .wav files (stimuli)
for root, dirs, files in os.walk(WAV_FOLDER):
for wav_file in files:
if wav_file.endswith(".wav"):
audio = Audio.load(root + wav_file)
processor = BottleneckProcessor(weights='BabelMulti')
features = processor.process(audio)
vectors = features.data
utterance = wav_file.split('.')[0]
all_features[utterance] = vectors
for row in distance_list.itertuples():
row_index = getattr(row, 'Index')
trip_id = getattr(row, 'tripletid')
bottle_oth = all_features[trip_id + "_OTH"]
bottle_tgt = all_features[trip_id + "_TGT"]
bottle_x = all_features[trip_id + "_X"]
eucl_oth_x = \
calculate_distances_dtw(bottle_oth,\
def audio(wav_file):
return Audio.load(wav_file)