def expand(self, audio):
ori_len = audio.shape[0]
tmp = resample(audio, r=0.5, type='sinc_best')
down_len = tmp.shape[0]
tmp = resample(tmp,
r=(ori_len + 1) / float(down_len),
type='sinc_best')
tmp = librosa.stft(audio, 1024)
phase = np.divide(tmp, np.abs(tmp))
spec_input = np.abs(librosa.stft(audio, 1024))[0:n_input, ::]
spec_input = spec_input[::, 0:spec_input.shape[1] // n_len * n_len]
spec_input = np.split(spec_input, spec_input.shape[1] // n_len, axis=1)
spec_input = np.asarray(spec_input)
spec_input = np.expand_dims(spec_input, axis=-1)
feed_dict = {self.input_op: np.log1p(spec_input) / 12.0}
debug = self.sess.run(self.debug_op, feed_dict=feed_dict)
np.save('debug.npy', debug)
S = self.sess.run(self.eva_op, feed_dict=feed_dict)
S[S >= 5e3] = 5e3
S[S <= 0] = 0
print('mean', np.mean(S))
print(np.sum(np.isinf(S)))
S = np.squeeze(np.concatenate(np.split(S, S.shape[0]), axis=2),
axis=(0, -1))
phase = phase[..., :S.shape[1]]
print(phase.shape)
print(S.shape)
print(np.sum(np.isinf(np.multiply(S, phase))))
X = librosa.istft(np.multiply(S, phase))
return X
def __stream_audio_realtime(filepath, rate=44100):
total_chunks = 0
format = pyaudio.paInt16
channels = 1 if sys.platform == 'darwin' else 2
record_cap = 10 # seconds
p = pyaudio.PyAudio()
stream = p.open(format=format, channels=channels, rate=rate, input=True, frames_per_buffer=ASR.chunk_size)
print "o\t recording\t\t(Ctrl+C to stop)"
try:
desired_rate = float(desired_sample_rate) / rate # desired_sample_rate is an INT. convert to FLOAT for division.
for i in range(0, rate/ASR.chunk_size*record_cap):
data = stream.read(ASR.chunk_size)
_raw_data = numpy.fromstring(data, dtype=numpy.int16)
_resampled_data = resample(_raw_data, desired_rate, "sinc_best").astype(numpy.int16).tostring()
total_chunks += len(_resampled_data)
stdout.write("\r bytes sent: \t%d" % total_chunks)
stdout.flush()
yield _resampled_data
stdout.write("\n\n")
except KeyboardInterrupt:
pass
finally:
print "x\t done recording"
stream.stop_stream()
stream.close()
p.terminate()
def resample(y, orig_sr, target_sr, res_type='sinc_fastest'):
"""Resample a signal from orig_sr to target_sr
Arguments:
y -- (ndarray) audio time series
orig_sr -- (int) original sample rate of y
target_sr -- (int) target sample rate
res_type -- (str) resample type (see below)
Returns y_hat:
y_hat -- (ndarray) y resampled from orig_sr to target_sr
Notes:
if scikits.samplerate is installed, resample will use res_type
otherwise, it will fall back on scip.signal.resample
"""
if orig_sr == target_sr:
return y
if _HAS_SAMPLERATE:
y_hat = samplerate.resample(y,
float(target_sr) / orig_sr,
res_type)
else:
n_samples = len(y) * target_sr / orig_sr
y_hat = scipy.signal.resample(y, n_samples, axis=-1)
return y_hat
def prepareData(self):
if self.newSampleRate is not None:
resamplingFactor = float(self.newSampleRate) / float(
self.sampleRate)
elif self.filterb is None:
print("skipping data preparation, nothing to do")
return
newChannels = [None for x in xrange(len(self.channels))]
for i, channel in itertools.izip(xrange(len(self.channels)),
self.channels):
print("channel {0}".format(i))
newData = channel
if self.filterb is not None:
print(" lowpass")
newData = signal.lfilter(self.filterb, self.filtera, channel)
if self.newSampleRate is not None:
print(" resampling")
newData = samplerate.resample(
numpy.array(channel), resamplingFactor,
'sinc_best' if self.filterb is None else 'sinc_medium')
newChannels[i] = newData
self.channels = newChannels
if self.newSampleRate is not None:
self.sampleRate = self.newSampleRate
def play(self):
if not self.sound:
self.sound = pygame.mixer.Sound(self.file)
self.freq = self.info.rate
if resampleOk:
(freq,format,channels) = pygame.mixer.get_init()
if freq != self.freq:
print "reload sound ",self.file
self.freq = self.info.rate
self.sound = pygame.mixer.Sound(self.file)
if self.freq != freq:
snd_array = pygame.sndarray.array(self.sound)
samples = len(snd_array)/2
samples = int(samples*freq*1.0/(self.info.rate))
print "start resampling ",self.file," from ",self.info.rate," to ",freq," len ",len(snd_array)/2," visée ",samples
# if samples != len(snd_array):
# snd_array = np.resize(snd_array,(samples,2))
snd_array = resample(snd_array, freq*1.0/self.info.rate, "sinc_fastest").astype(snd_array.dtype)
# datal = signal.resample(snd_array[0::2],samples).astype(snd_array.dtype)
# datar = signal.resample(snd_array[1::2],samples).astype(snd_array.dtype)
# snd_array = np.resize(snd_array,(len(datal)*2,2))
# snd_array[0::2] = datal
# snd_array[1::2] = datar
# print "end resampling ",snd_array
self.sound = pygame.sndarray.make_sound(snd_array)
self.freq = freq
if self.volume:
self.sound.set_volume(self.volume)
self.channel = self.sound.play()
self.playTime = time.time()
if self.event:
self.channel.set_endevent(self.event)
def resample(source, source_sf, target_sf):
if source_sf != target_sf:
from scikits import samplerate
ratio = float(target_sf) / source_sf
return samplerate.resample(source, ratio, RESAMPLE_TYPE)
else:
return source
def __stream_audio_realtime(filepath, rate=44100):
total_chunks = 0
format = pyaudio.paInt16
channels = 1 if sys.platform == 'darwin' else 2
record_cap = 10 # seconds
p = pyaudio.PyAudio()
stream = p.open(format=format,
channels=channels,
rate=rate,
input=True,
frames_per_buffer=ASR.chunk_size)
print "o\t recording\t\t(Ctrl+C to stop)"
try:
desired_rate = float(
desired_sample_rate
) / rate # desired_sample_rate is an INT. convert to FLOAT for division.
for i in range(0, rate / ASR.chunk_size * record_cap):
data = stream.read(ASR.chunk_size)
_raw_data = numpy.fromstring(data, dtype=numpy.int16)
_resampled_data = resample(_raw_data, desired_rate,
"sinc_best").astype(
numpy.int16).tostring()
total_chunks += len(_resampled_data)
stdout.write("\r bytes sent: \t%d" % total_chunks)
stdout.flush()
yield _resampled_data
stdout.write("\n\n")
except KeyboardInterrupt:
pass
finally:
print "x\t done recording"
stream.stop_stream()
stream.close()
p.terminate()
def strfpak_resample(sound,slice_rate,resample_type):
'''
Resample a sound object using scikits.samplerate.resample.
Sound is upsampled to nearest multiple of slice_rate;
if sound.samplerate == slice_rate, no resampling is done.
Sound is also converted to float.
Currently, error is less than 10dB max, mostly very good.
'''
from scikits.samplerate import resample
from copy import deepcopy
# Calculate the resampling frequency for STRFPAK 5.3:
# round up to the nearest multiple of 'slice_rate'
input_freq = floor(sound.samplerate)
output_freq = ceil(input_freq/slice_rate)*slice_rate
# Copy so that we don't alter original object
sound2 = deepcopy(sound)
# Scale factor -- MATLAB makes .wav data be -1:1, scipy does integer
sound2.data = sound2.data/float(2**15)
# Resample if necessary
if output_freq > input_freq:
sound2.data = resample(sound2.data,output_freq/input_freq,
resample_type)
sound2.samplerate = output_freq
# Pad because matlab resampling works differently
sound2.data = sound2._zeropad(0,2)
return sound2
def record(self, signals, Fs):
'''
This simulates the recording of the signals by the microphones.
In particular, if the microphones and the room simulation
do not use the same sampling frequency, down/up-sampling
is done here.
Arguments:
----------
signals: An ndarray with as many lines as there are microphones.
Fs: the sampling frequency of the signals.
'''
if signals.shape[0] != self.M:
raise NameError('The signals array should have as many lines as there are microphones.')
if signals.ndim != 2:
raise NameError('The signals should be a 2D array.')
if Fs != self.Fs:
from scikits.samplerate import resample
Fs_ratio = self.Fs/float(Fs)
newL = int(Fs_ratio*signals.shape[1])-1
self.signals = np.zeros((self.M, newL))
# scikits.samplerate resample function considers columns as channels (hence the transpose)
for m in range(self.M):
self.signals[m] = resample(signals[m], Fs_ratio, 'sinc_best')
else:
self.signals = signals
def convert_wav(File, ofile):
import scikits.audiolab as audiolab
from scikits.samplerate import resample
# lastest scikits.audiolab include sound record lib, based on python-alsaaudio
# if you want make the down sample rate using scipy.signal
#import scipy.signal
#using audiolab to read wav file
Signal, fs = audiolab.wavread(File)[:2]
#changing the original sample rate to 16000fs fast mode
Signal = resample(Signal, fr/float(fs), 'sinc_best')
#changing sample rate from audio file using scipy this is a bit slow
#Signal=scipy.signal.resample(Signal,int(round(len(Getsignal)*fr)/float(fs)),window=None)
# file Format type
fmt = audiolab.Format('flac', 'pcm16')
nchannels = 1
# convert into the file .flac
ofile = audiolab.Sndfile(FileNameTmp, 'w', fmt, nchannels, fr)
#writing in the file
ofile.write_frames(Signal)
#
return ofile
def wav_convert(data, SR, tar_freq):
# If the input signal is stereo, make it mono.
if ndim(data) == 2:
# Mix stereo signal into a mono signal
buff01 = 0.49 * (data[:, 0] + data[:, 1])
wave_ts = array(buff01)
else:
wave_ts = array(data[:])
wave_ts = array(wave_ts)
# Set a sampling rate
up_SR = 44100
# Compute a ratio to feed into resample function
ratio = float(float(tar_freq)/float(up_SR))
# Resample the file.
wave_ts = resample(wave_ts, ratio , 'linear')
# Transpose the data list.
wave_ts = transpose(wave_ts)
# Return wave_ts signal
return wave_ts
def wav_convert(data, SR, tar_freq):
# If the input signal is stereo, make it mono.
if ndim(data) == 2:
buff01 = 0.49 * (data[:, 0] + data[:, 1])
wave_ts = array(buff01)
else:
wave_ts = array(data[:])
wave_ts = array(wave_ts)
# print "Shape of wave_ts", shape(wave_ts)
up_SR = 44100
ratio = float(float(tar_freq)/float(up_SR))
print tar_freq
print float(float(tar_freq)/float(up_SR))
wave_ts = resample(wave_ts, ratio , 'linear')
# SR_div = int(floor(up_SR/tar_freq))
# wave_ts = sig.decimate(wave_ts, SR_div)
# Transpose the data list.
wave_ts = transpose(wave_ts)
print 'length wave_ts', shape(wave_ts)
return wave_ts
def analyze(sig, samplerate=8000, resample_to=8000):
e_min_scale = 0.3
e_min_distance = 1
e_a_scale = 0.5
e_W_ms = 2000
min_len = 30
if samplerate != resample_to:
print("resampling")
sig = srate.resample(sig, resample_to/samplerate, 'sinc_best')
frames,frame_size = ad.frames(sig, samplerate, 64)
#ac_peaks = [ad.ac_peaks(frame) for frame in frames]
#energy = ad.energies(frames, ...)
#energy = log_energy
print("getting normalized spectra")
normalized_spectrum = ad.normalized_spectrum(frames, samplerate)
print("spectral entropy")
spectral_entropy = np.fromiter(ad.entropy(frame) for frame in normalized_spectrum)
entopy_t = np.percentile(spectral_entropy, 80)
se_segments = ad.segments_to_seconds(ad.entropy_segment_indexes(spectral_entropy, entropy_t))
print("energy based computations")
energy, smooth, en_a, en_t, en_lmin = ad.energy_thresholds(sig, noise_dist=e_min_distance, a_scale=e_a_scale, min_scale=e_min_scale, W_ms=e_W_ms)
energy_segments = ad.get_voice_segments(smooth, en_t, ad.get_segment_indexes(smooth, en_t, min_len=10))
#energy_t = ad.thresholds(frames)
#entropy_t = ad.entropy_t(frames)
#energy_indexes = ad.get_segment_indexes(x, t, min_len=30)
#entropy_indexes = ad.get_entropy_indexes(x, t, min_len=30)
#combination_indexes = ad.get_combined_indexes(x, t, min_len=30)
return spectral_entropy, se_t, se_segments, energy, en_t, energy_segments
def listen(stream, queue):
try:
while not exit:
stream.start_stream()
print('Listening...')
for i in range(0,size):
data = stream.read(frame)
ar = numpy.fromstring(data, dtype=numpy.int16)
data2 = resample(ar, (16000./48000.), 'linear')
q.put(data2)
# samps = numpy.fromstring(data, dtype=numpy.int16)
# print (samps, q.qsize())
rms = audioop.rms(data, 2)
print rms
stream.stop_stream()
if exit:
sys.exit()
q.join()
except IOError:
print('ERROR!!!!')
pass
stream.stop_stream()
stream.close()
pyaud.terminate()
print "----------------------------------------------------------------------------------------------------------------"
def calculate_attributes(self):
source = self.source
freq = self.frequency
sampling_rate = float(source.sampling_rate)
fft_sampling_rate = sampling_rate/float(source.fft_step_size)
window_length = float(source.fft_window_size)/sampling_rate
# FIXME real time should be passed in as an extra field
self.start = window_length
if self.first_frame > 0:
self.start += (self.first_frame - 1)/fft_sampling_rate
self.length = window_length
if len(freq) > 1:
# if 'length' not in self.__dict__:
# print self.__dict__
# print self.__dict__['length']
self.length += (len(freq) - 1)/fft_sampling_rate
# print self.length
self.end = self.start + self.length
for k in ('frequency','amplitude'):
a = getattr(self,k)
setattr(self, k+'_min', min(a))
setattr(self, k+'_max', max(a))
setattr(self, k+'_mean', sum(a)/len(a))
freq_window = 2 # seconds
freq_fft_size = 128
resampled_freq = resample(freq, freq_window*freq_fft_size/fft_sampling_rate, 'sinc_fastest') # FIXME truncate array?
self.freq_fft = abs(rfft(resampled_freq,n=freq_fft_size,overwrite_x=True))[1:]
def generate_scale(self):
"""
Given the initial note, middle C, create the rest of the musical scale by
resampling.
Returns: Dictionary of musical scale with the key being the name of the note
and the value being the corresponding sound object.
"""
pygame.mixer.init()
wav = util.get_app_path() + "res/piano-c.wav"
sound = pygame.mixer.Sound(wav)
pygame.mixer.set_num_channels(32)
sndarray = pygame.sndarray.array(sound)
ratio_dict = {'low_c': 1, 'c_sharp': .944, 'd': .891, 'd_sharp':.841, 'e':.794,
'f':.749, 'f_sharp': .707, 'g': .667, 'g_sharp': .63, 'a': .594,
'a_sharp': .561, 'b':.53, 'high_c':.5}
# Generate the Sound objects from the dictionary.
scale = {}
for key,value in ratio_dict.iteritems():
smp = resample(sndarray, value,"sinc_fastest").astype(sndarray.dtype)
# Use the key, currently a string, as a variable
scale[key] = pygame.sndarray.make_sound(smp)
self.scale_dict=scale
def calculate_attributes(self):
source = self.source
freq = self.frequency
sampling_rate = float(source.sampling_rate)
fft_sampling_rate = sampling_rate / float(source.fft_step_size)
window_length = float(source.fft_window_size) / sampling_rate
# FIXME real time should be passed in as an extra field
self.start = window_length
if self.first_frame > 0:
self.start += (self.first_frame - 1) / fft_sampling_rate
self.length = window_length
if len(freq) > 1:
# if 'length' not in self.__dict__:
# print self.__dict__
# print self.__dict__['length']
self.length += (len(freq) - 1) / fft_sampling_rate
# print self.length
self.end = self.start + self.length
for k in ('frequency', 'amplitude'):
a = getattr(self, k)
setattr(self, k + '_min', min(a))
setattr(self, k + '_max', max(a))
setattr(self, k + '_mean', sum(a) / len(a))
freq_window = 2 # seconds
freq_fft_size = 128
resampled_freq = resample(freq, freq_window * freq_fft_size /
fft_sampling_rate,
'sinc_fastest') # FIXME truncate array?
self.freq_fft = abs(
rfft(resampled_freq, n=freq_fft_size, overwrite_x=True))[1:]
def train_codebook(basedirectory,
spectral,
desired_fs,
clfs,
n_samples):
"""Train the codebooks.
Arguments:
:param basedirectory: root directory of the audio corpus
:param spectral:
Spectral feature extraction.
Object should be picklable and implement the
\c Spectral abc; i.e. provide a \c transform method.
:param clfs:
list of clusterers. valid clusterers have a \c fit method
and a \c predict method. optionally, for soft vq, also implement
a \c predict_proba method.
:param n_samples:
number of spectral frames to sample from the audio corpus.
:returns:
a list of Codebook objects, of same length as the output of spectral_func
"""
wavs = list(rglob(basedirectory, '*.wav'))
np.random.shuffle(wavs)
inds = None
idx = 0
X = None
for i, wav in enumerate(wavs):
if i % 10 == 0 and i > 0:
print 'samples: {3}/{4}; loading file: {0} ({1}/{2})'.format(
wavs[i],
i+1,
len(wavs),
X.shape[0],
n_samples
)
sig, fs, _ = audiolab.wavread(wav)
start, stop = trim_silence(sig, fs)
specs = spectral.transform(samplerate.resample(sig[start:stop],
desired_fs/fs,
'sinc_best'))
if inds is None:
inds = [0] + list(np.cumsum([spec.shape[1] for spec in specs]))
spec = np.hstack(specs)
if idx + spec.shape[0] >= n_samples:
spec = spec[:n_samples - idx, :]
if X is None:
X = spec
else:
X = np.vstack((X, spec))
idx += spec.shape[0]
if idx >= n_samples:
break
cdbs = [Codebook(clf) for clf in clfs]
for i, cdb in enumerate(cdbs):
cdb.train(X[:, inds[i]:inds[i+1]])
return cdbs
def resample_noise_file(sr_audio, sr_noise, data_noise):
start1 = time.time()
if sr_noise != 0:
resampling_rate = float(sr_audio) / float(sr_noise)
data_noise = resample(data_noise, resampling_rate, "sinc_best").astype(np.int16)
end1 = time.time()
# print("0:", end1-start1)
return data_noise
def resample(self, samplerate, resample_type="sinc_best"):
"""
Returns a resampled version of the sound.
"""
if not have_scikits_samplerate:
raise ImportError("Need scikits.samplerate package for resampling")
y = array(resample(self, float(samplerate / self.samplerate), resample_type), dtype=float64)
return Sound(y, samplerate=samplerate)
def TestGBPDN2():
# ________________________________________
print 'Test: generalized basis pursuit decomposition'
fs = 8000.
btmp,fstmp,fmt = audiolab.wavread('glockenspiel.wav')
b = samplerate.resample(btmp, fs/fstmp,'sinc_best')
L = len(b)
A = GaborBlock(L,1024)
B = GaborBlock(A.M,64)
C = DictionaryUnion(A,B)
b = np.hstack((b,np.zeros(C.M-L))) # pad to block boundary
spow = 10*np.log10(b.conj().dot(b))
# additive white noise
snr = 15
nvar = np.sum(np.abs(b)**2)/(10**(snr/10)) # 1e-2
n = np.sqrt(nvar)*np.random.randn(C.M)/np.sqrt(C.M)
tonemap = np.reshape(range(A.N),(A.N/A.fftLen,A.fftLen)).transpose()
transmap = np.reshape(range(B.N),(B.N/B.fftLen,B.fftLen)).transpose()
f,fgrad = BP_factory()
#f,fgrad = TT_factory(tonemap,transmap)
xe = GBPDN_momentum(C,b+n,f,fgrad,maxerr=nvar,maxits=200,stoptol=1e-3,muinit=1e-1,momentum=0.9,smoothinit=1e-5,anneal=0.96)
ye = np.real(C.dot(xe))
r = b-ye;
rpow = 10*np.log10(r.conj().dot(r))
print 'SNR = %f' % (spow-rpow)
ynoise = np.array(samplerate.resample(b+n,fstmp/fs,'sinc_best'),dtype='float64')
ydenoise = np.array(samplerate.resample(ye,fstmp/fs,'sinc_best'),dtype='float64')
yetone = np.real(A.dot(xe[:tonemap.size]))
yetone = np.array(samplerate.resample(yetone,fstmp/fs,'sinc_best'),dtype='float64')
yetrans = np.real(B.dot(xe[tonemap.size:]))
yetrans = np.array(samplerate.resample(yetrans,fstmp/fs,'sinc_best'),dtype='float64')
print 'Error (should be <= %f): %f' % (nvar,np.sum((r)**2))
print '----------------------------------------'
def resample(x, fs_old, fs_new, axis=0, algorithm='scipy'):
"""Resample signal
If available resampling is done using scikit samplerate. Otherwise,
scipy's FFT-based resample function will be used.
Converters available in scikits.samplerate:
- sinc_medium
- linear
- sinc_fastest
- zero_order_hold
- sinc_best
"""
if fs_old == fs_new:
return x
else:
ratio = float(fs_new) / fs_old
if use_scikits_resample and algorithm != 'scipy':
if algorithm == 'scikits':
algo = 'sinc_medium'
else:
algo = algorithm
if axis == 0:
tmp = sks.resample(x, ratio, algo)
else:
tmp = sks.resample(x.T, ratio, algo)
if tmp.dtype != x.dtype:
tmp = tmp.astype(x.dtype, casting='safe')
return tmp
else:
if axis == 0:
n_samples_new = int(np.round(x.shape[0] * ratio))
return signal.resample(x, n_samples_new)
else:
n_samples_new = int(np.round(x.shape[1] * ratio))
return signal.resample(x, n_samples_new, axis=axis)
def play(self):
if have_sk_audiolab and have_sk_samplerate:
play(
np.array(resample(self.data, 44100. / self.fs, 'sinc_best'),
dtype=np.float64))
else:
print(
'Warning: scikits.audiolab and scikits.samplerate are required to play audiofiles.'
)
def resample(y, orig_sr, target_sr, res_type='sinc_fastest', fix=True,
**kwargs):
"""Resample a time series from orig_sr to target_sr
:usage:
>>> # Downsample from 22 KHz to 8 KHz
>>> y, sr = librosa.load('file.wav', sr=22050)
>>> y_8k = librosa.resample(y, sr, 8000)
:parameters:
- y : np.ndarray [shape=(n,)]
audio time series
- orig_sr : int > 0 [scalar]
original sampling rate of ``y``
- target_sr : int > 0 [scalar]
target sampling rate
- res_type : str
resample type (see note)
- fix : bool
adjust the length of the resampled signal to be of size exactly
``ceil(target_sr * len(y) / orig_sr)``
- *kwargs*
If ``fix==True``, additional keyword arguments to pass to
:func:`librosa.util.fix_length()`.
:returns:
- y_hat : np.ndarray [shape=(n * target_sr / orig_sr,)]
``y`` resampled from ``orig_sr`` to ``target_sr``
.. note::
If `scikits.samplerate` is installed, :func:`librosa.core.resample`
will use ``res_type``.
Otherwise, it will fall back on `scipy.signal.resample`
"""
if orig_sr == target_sr:
return y
n_samples = int(np.ceil(y.shape[-1] * float(target_sr) / orig_sr))
if _HAS_SAMPLERATE:
y_hat = samplerate.resample(y.T,
float(target_sr) / orig_sr,
res_type).T
else:
y_hat = scipy.signal.resample(y, n_samples, axis=-1)
if fix:
y_hat = util.fix_length(y_hat, n_samples, **kwargs)
return np.ascontiguousarray(y_hat, dtype=y.dtype)
def resample(self, samplerate, resample_type='sinc_best'):
'''
Returns a resampled version of the sound.
'''
if not have_scikits_samplerate:
raise ImportError('Need scikits.samplerate package for resampling')
y = array(resample(self, float(samplerate / self.samplerate), resample_type),
dtype=float64)
return Sound(y, samplerate=samplerate)
def set_note(self, note, hstring):
octave = note[-1:]
self.note_dict[note]['octave']=octave
pure_note = note.strip(octave)
ratio = self.note_dict[note]['ratio']
globals()[note] = resample(hstring, ratio, "sinc_best").astype(hstring.dtype)
globals()[note+"_note"] = self.pygame.sndarray.make_sound(globals()[note])
globals()[note+"_note"].set_volume(.15)
self.note_dict[note]['snd_arry']=note+"_note"
def resample(self, samplerate, resample_type='sinc_best'):
'''
Returns a resampled version of the sound.
'''
if not have_scikits_samplerate:
raise ImportError('Need scikits.samplerate package for resampling')
y = np.array(resample(self, float(samplerate / self.samplerate), resample_type),
dtype=np.float64)
return Sound(y, samplerate=samplerate)
def rce2wav(rcefile,
samplerate,
resample=None,
split=False,
splitsuffixes=None):
"""
a rce file is a raw, float64 file containing pressure level at each frame.
FORMAT:
1 double: number of sources
Each frame then contains the data for each source, interleaved
rcefile : path to the .rce file
samplerate : samplerate of the simulation (sim.samplerate)
resample : INT (new samplerate).
If given, file will be resampled to this sample-rate.
NOTE:
an .rce file is simply raw data. To load it: numpy.fromfile(path, dtype=float).
The samplerate is not saved with the data, but it is the same used by the
simulation.
"""
def name_with_suffix(origname, ext='wav', suffix=""):
if suffix:
suffix = "-%s" % str(suffix)
base = os.path.splitext(origname)[0]
return "%s%s.%s" % (base, suffix, ext)
raw = rce2array(rcefile)
from e import audiosample
if resample is not None:
try:
from scikits import samplerate
raw = samplerate.resample(raw, resample / samplerate, 'sync_fast')
samplerate = resample
except ImportError:
warnings.warn(
"resampling not available (install scikits.samplerate to enable). Keeping original samplerate"
)
numch = raw.shape[1] if len(raw.shape) > 1 else 1
if not split or numch == 1:
out = name_with_suffix(rcefile)
audiosample.Sample(raw, samplerate).write(out)
return out
else:
outs = []
if splitsuffixes is None:
splitsuffixes = [str(i) for i in range(numch)]
else:
splitsuffixes = splitsuffixes[:numch]
for i, suffix in enumerate(splitsuffixes):
out = name_with_suffix(rcefile, suffix=suffix)
outs.append(out)
audiosample.Sample(raw[:, i], samplerate).write(out)
return outs
def playSound(self, pitch, noteIndex):
self.getControlVals()
toneArray = self.soundArrays[self.selectedSound]
if self.attack != 0:
changedArray = resample(toneArray*self.attack,
pitch,"sinc_fastest").astype(toneArray.dtype)
else:
changedArray = resample(toneArray,
pitch,"sinc_fastest").astype(toneArray.dtype)#makes correct type
newSound = pygame.sndarray.make_sound(changedArray)
newSound.set_volume(self.volume)
if (self.isRecording and self.songCount < len(self.songs) and
self.selectedSong != None):
currentTime = time.time()
timePlayed = currentTime - self.startTime
self.makeRecording(newSound,timePlayed,noteIndex,self.volume,
self.length,self.fade,self.isSustained)
if self.isSustained: self.length = -1
if self.fade != 0: newSound.play(0,self.length,0).fadeout(self.fade)
elif self.isSustained: newSound.play(0,-1,0)
else: newSound.play(0,self.length,0)
def resample(s, fs, fs_new, algorithm='sinc_best'):
'''
sinc_medium : Band limited sinc interpolation, medium quality, 121dB SNR, 90% BW.
linear : Linear interpolator, very fast, poor quality.
sinc_fastest : Band limited sinc interpolation, fastest, 97dB SNR, 80% BW.
zero_order_hold : Zero order hold interpolator, very fast, poor quality.
sinc_best : Band limited sinc interpolation, best quality, 145dB SNR, 96% BW.
'''
from scikits.samplerate import resample
if fs_new == fs:
return s
return resample(s, fs_new / fs, 'sinc_best')
def load_song(song_path):
"""
Loads an individual song, normalizes it, and returns it.
"""
fs, data = wavfile.read(song_path)
data = data.mean(axis=1).flatten()
data = samplerate.resample(data, TARGET_FS / fs, 'linear')
# Normalize
data = data / np.max(data) * .95
print "\t- Loaded %s" % song_path
return data
def _map(f, n_filters=40, n_ceps=13, fs=8000, downsample='sinc_best',
win=0.025, shift=0.01, delta_order=2, energy=True, vad=True,
dtype='float32', get_spec=False, get_mspec=True, get_mfcc=False,
robust=True):
'''
Return
------
(name, features, vad, sum1, sum2)
'''
try:
audio_path, segments = f
# load audio data
s, _ = speech.read(audio_path)
# check frequency for downsampling (if necessary)
if _ is not None:
if fs is not None and fs != _:
if fs < _: # downsample
s = resample(s, fs / _, 'sinc_best')
else:
raise ValueError('Cannot perform upsample from frequency: '
'{}Hz to {}Hz'.format(_, fs))
else:
fs = _
N = len(s)
features = []
for name, start, end, channel in segments:
start = int(float(start) * fs)
end = int(N if end < 0 else end * fs)
data = s[start:end, channel] if s.ndim > 1 else s[start:end]
tmp = speech_features_extraction(data.ravel(), fs=fs,
n_filters=n_filters, n_ceps=n_ceps,
win=win, shift=shift, delta_order=delta_order,
energy=energy, vad=vad, dtype=dtype,
get_spec=get_spec, get_mspec=get_mspec, get_mfcc=get_mfcc)
if tmp is not None:
features.append((name,) + tmp)
else:
msg = 'Ignore segments: %s, error: NaN values' % name
warnings.warn(msg)
return features
except Exception, e:
msg = 'Ignore file: %s, error: %s' % (f[0], str(e))
warnings.warn(msg)
if robust:
return None
else:
import traceback; traceback.print_exc()
raise e
def resample_data(self):
for i_series in xrange(8):
X_series = np.array(self.train_X_series[i_series]).astype(np.float32)
X_series = resample(X_series, 250.0/500.0, 'sinc_fastest')
self.train_X_series[i_series] = X_series
y_series = np.array(self.train_y_series[i_series]).astype(np.int32)
# take later predictions ->
# shift all predictions backwards compared to data.
# this ensures you are not using data from the future to make a prediciton
# rather in a bad case maybe you do not even have all data up to the sample
# to make the prediction
y_series = y_series[1::2]
# maybe some border effects remove predictions
y_series = y_series[-len(X_series):]
self.train_y_series[i_series] = y_series
def test_mono():
fs = 16000.
fr = 8000.
f0 = 440.
# Create a small stereo audio array with dephased channel
x = np.sin(2 * np.pi * f0/fs * np.arange(0, 2 * fs))
# Upsampled reference
z_r = np.sin(2 * np.pi * f0/fr * np.arange(0, 2 * fr))
z = resample(x, fr /fs, 'sinc_best')
assert np.max(np.abs(z_r[10:-1] - z[10:])) < 1e-2
def resample_data(self):
for i_series in xrange(8):
X_series = np.array(self.train_X_series[i_series]).astype(
np.float32)
X_series = resample(X_series, 250.0 / 500.0, 'sinc_fastest')
self.train_X_series[i_series] = X_series
y_series = np.array(self.train_y_series[i_series]).astype(np.int32)
# take later predictions ->
# shift all predictions backwards compared to data.
# this ensures you are not using data from the future to make a prediciton
# rather in a bad case maybe you do not even have all data up to the sample
# to make the prediction
y_series = y_series[1::2]
# maybe some border effects remove predictions
y_series = y_series[-len(X_series):]
self.train_y_series[i_series] = y_series
def create_specgram_image_of_audio_file(fid):
# Read wav file
wave_obj = wave.open(fid, 'r')
# Vectorize the wave file
sample_vec = np.fromstring(wave_obj.readframes(wave_obj.getnframes()), np.short)/32780.0
sample_rate = wave_obj.getframerate() # get sample rate
wave_obj.close()
# Re-sample the vector to 22050 Hz
if not sample_rate == 22050:
sample_vec = resample(sample_vec, FS/sample_rate, 'sinc_best')
num_of_samples = len(sample_vec)
# Normalize the vector
sample_vec = 0.9*sample_vec/sample_vec.max()
# Create spectrogram
# the window size (FFT size) is 512, and FFT HOP size is 128 -> 128/512 = 0.25 meaning a 75% overlap
num_of_specgram_frames = int((num_of_samples-FFT_SIZE)/FFT_HOP_SIZE) + 1
# Hanning window function
window = 0.5 * (1.0 - np.cos(np.array(range(FFT_SIZE))*2.0*np.pi/(FFT_SIZE-1)))
# Apply STFT
specgram = []
for j in range(num_of_specgram_frames):
vec = sample_vec[j*FFT_HOP_SIZE: j*FFT_HOP_SIZE + FFT_SIZE] * window
real_fft = np.fft.rfft(vec, FFT_SIZE)
real_fft = abs(real_fft[:int(FFT_SIZE/2)])
specgram.append(real_fft)
# Create spectrogram image
specgram_transposed = np.transpose(np.array(specgram))
specgram_image = np.zeros_like(specgram_transposed)
for i in range(specgram_transposed.shape[0]):
specgram_image[i] = specgram_transposed[-i - 1]
if np.max(specgram_image) <= 0.0:
print('Problem: np.max(specgram_image) <= 0.0')
else:
specgram_image /= np.max(specgram_image)
return specgram_image
def TestBPDN2():
# ________________________________________
print 'Test: basis pursuit decomposition'
fs = 8000
btmp = audiolab.wavread('glockenspiel.wav')[0]
b = samplerate.resample(btmp, fs/44100.,'sinc_best')
L = len(b)
A = GaborBlock(L,1024)
B = GaborBlock(A.M,64)
C = DictionaryUnion(A,B)
b = np.hstack((b,np.zeros(C.M-L)))
e = 1e-2
x = BPDN(C,b,e,100)
ye = np.real(C.dot(x))
print 'Error (should be <= %f): %f' % (e,np.sum((b-ye)**2))
print '----------------------------------------'
xtone = x[:A.N]
xtrans = x[A.N:]
ytone = np.real(A.dot(xtone))
ytrans = np.real(B.dot(xtrans))
audiolab.wavwrite(ytone,'ytone.wav',fs)
audiolab.wavwrite(ytrans,'ytrans.wav',fs)
# tonal decomp
m = np.log10(np.abs(A.conj().transpose().dot(ytone)))
tfgrid = np.reshape(range(0,A.N),(A.N/A.fftLen,A.fftLen))
tfgrid = tfgrid[:,:A.fftLen/2+1]
pyplot.subplot(2,1,1)
pyplot.imshow(m[tfgrid].transpose(), aspect='auto', interpolation='bilinear', origin='lower')
# transient decomp
m = np.log10(np.abs(B.conj().transpose().dot(ytrans)))
tfgrid = np.reshape(range(0,B.N),(B.N/B.fftLen,B.fftLen))
tfgrid = tfgrid[:,:B.fftLen/2+1]
pyplot.subplot(2,1,2)
pyplot.imshow(m[tfgrid].transpose(), aspect='auto', interpolation='bilinear', origin='lower')
pyplot.show()
def resample_audio(samples, current_sample_rate, new_sample_rate):
'''Resamples audio samples
Given an audio sample and the current sample rate, resample to a new
sample rate.
Args:
samples: The audio samples from wavfile.read(<file>)[0]
current_sample_rate: The current sample rate from wavfile.read(<file>)[1]
new_sample_rate: The new sample rate
Returns:
A tuple with the new sample rate and the new audio samples. This
is exactly similar to wavfile.read(<filename>)
'''
return new_sample_rate, resample(samples,
new_sample_rate / current_sample_rate,
'sinc_best')
def resample(y, orig_sr, target_sr, res_type='sinc_fastest'):
"""Resample a time series from orig_sr to target_sr
:usage:
>>> # Downsample from 22 KHz to 8 KHz
>>> y, sr = librosa.load('file.wav', sr=22050)
>>> y_8k = librosa.resample(y, sr, 8000)
:parameters:
- y : np.ndarray
audio time series
- orig_sr : int
original sampling rate of ``y``
- target_sr : int
target sampling rate
- res_type : str
resample type (see note)
:returns:
- y_hat : np.ndarray
``y`` resampled from ``orig_sr`` to ``target_sr``
.. note::
If scikits.samplerate is installed, resample will use res_type
otherwise, it will fall back on scipy.signal.resample
"""
if orig_sr == target_sr:
return y
if _HAS_SAMPLERATE:
y_hat = samplerate.resample(y.T,
float(target_sr) / orig_sr,
res_type).T
else:
n_samples = y.shape[-1] * target_sr / orig_sr
y_hat = scipy.signal.resample(y, n_samples, axis=-1)
return np.ascontiguousarray(y_hat)
def resample(self, target_fs, mode="sinc_best"):
"""
Resample the signal. One implication of the signal being digital, is that the resulting sampling
frequency is not guaranteed to be exactly at `target_fs`.
This method wraps :func:`~samplerate.resample`
:param target_fs: The new sampling frequency
:type target_fs: float
:param mode: to be passed to :func:`~scikits.samplerate.resample`
:type mode: str
:return:
"""
# num = target_fs * self.size / self.fs
ratio = target_fs / self.fs
out = resample(self,ratio,mode)
real_fs = out.size * self.fs / float(self.size)
out = self._copy_attrs_to_array(out, fs=real_fs)
return out
def load_wav(filename):
'''
'''
f = open(filename, "r")
nchannels, samplewidth, framerate, nframes,\
_, _ = f.getparams()
frames = f.readframes(nframes)
f.close()
if samplewidth == 1:
dtype = np.uint8
maxvalue = 2 ** 8
elif samplewidth == 2:
dtype = np.int16
maxvalue = 2 ** (16 - 1)
elif samplewidth == 4:
dtype = np.int32
maxvalue = 2 ** (32 - 1)
else:
print "Error: unsupported samplewidth {}".format(samplewidth)
return None
all_samples = np.fromstring(frames, dtype)
combined_samples = np.zeros(nframes)
for i in xrange(nchannels):
a = combined_samples
b = all_samples[i::nchannels]
condition = np.less(np.abs(a), np.abs(b))
combined_samples = np.choose(condition, [a, b])
normalized_samples = combined_samples / maxvalue
if dtype == np.uint8:
normalized_samples = 2 * normalized_samples - 1
desired_framerate = 16384.
resampled_samples = resample(normalized_samples,
desired_framerate / framerate, 'sinc_best')
return Signal(resampled_samples, desired_framerate, filename)
def generate_scale(self):
"""
Given the initial note, middle C, create the rest of the musical scale by
resampling.
Returns: Dictionary of musical scale with the key being the name of the note
and the value being the corresponding sound object.
"""
pygame.mixer.init()
wav = util.get_app_path() + "res/piano-c.wav"
sound = pygame.mixer.Sound(wav)
pygame.mixer.set_num_channels(32)
sndarray = pygame.sndarray.array(sound)
ratio_dict = {
'low_c': 1,
'c_sharp': .944,
'd': .891,
'd_sharp': .841,
'e': .794,
'f': .749,
'f_sharp': .707,
'g': .667,
'g_sharp': .63,
'a': .594,
'a_sharp': .561,
'b': .53,
'high_c': .5
}
# Generate the Sound objects from the dictionary.
scale = {}
for key, value in ratio_dict.iteritems():
smp = resample(sndarray, value,
"sinc_fastest").astype(sndarray.dtype)
# Use the key, currently a string, as a variable
scale[key] = pygame.sndarray.make_sound(smp)
self.scale_dict = scale
def resample(y, orig_sr, target_sr, fix=True, scale=False):
"""Resample a time series from orig_sr to target_sr
Args:
y : np.ndarray [shape=(n,) or shape=(2, n)]
audio time series. Can be mono or stereo.
orig_sr : number > 0 [scalar]
original sampling rate of `y`
target_sr : number > 0 [scalar]
target sampling rate
fix : bool
adjust the length of the resampled signal to be of size exactly
`ceil(target_sr * len(y) / orig_sr)`
scale : bool
Scale the resampled signal so that `y` and `y_hat` have
approximately equal total energy.
Returns
y_hat : np.ndarray [shape=(n * target_sr / orig_sr,)]
`y` resampled from `orig_sr` to `target_sr`
"""
if y.ndim > 1:
return np.vstack([resample(yi, orig_sr, target_sr, fix=fix)
for yi in y])
valid_audio(y, mono=True)
if orig_sr == target_sr:
return y
ratio = float(target_sr) / orig_sr
n_samples = int(np.ceil(y.shape[-1] * ratio))
try:
# Try to use scikits.samplerate if available
import scikits.samplerate as samplerate
y_hat = samplerate.resample(y.T, ratio, 'sinc_best').T
except ImportError:
warnings.warn('Could not import scikits.samplerate. '
'Falling back to scipy.signal')
import scipy.signal
y_hat = scipy.signal.resample(y, n_samples, axis=-1)
if fix:
y_hat = fix_length(y_hat, n_samples)
if scale:
y_hat /= np.sqrt(ratio)
return np.ascontiguousarray(y_hat, dtype=y.dtype)
def resample(y, orig_sr, target_sr, res_type='sinc_fastest'):
"""Resample a signal from orig_sr to target_sr
:usage:
>>> # Downsample from 22 KHz to 8 KHz
>>> y, sr = librosa.load('file.wav', sr=22050)
>>> y_8k = librosa.resample(y, sr, 8000)
:parameters:
- y : np.ndarray
audio time series
- orig_sr : int
original sampling rate of ``y``
- target_sr : int
target sampling rate
- res_type : str
resample type (see note)
:returns:
- y_hat : np.ndarray
``y`` resampled from ``orig_sr`` to ``target_sr``
.. note::
If scikits.samplerate is installed, resample will use res_type
otherwise, it will fall back on scipy.signal.resample
"""
if orig_sr == target_sr:
return y
if _HAS_SAMPLERATE:
y_hat = samplerate.resample(y.T, float(target_sr) / orig_sr, res_type).T
else:
n_samples = y.shape[-1] * target_sr / orig_sr
y_hat = scipy.signal.resample(y, n_samples, axis=-1)
return y_hat
def prepareData(self):
if self.newSampleRate is not None:
resamplingFactor = float(self.newSampleRate)/float(self.sampleRate)
elif self.filterb is None:
print("skipping data preparation, nothing to do")
return
newChannels = [None for x in xrange(len(self.channels))]
for i, channel in itertools.izip(xrange(len(self.channels)), self.channels):
print("channel {0}".format(i))
newData = channel
if self.filterb is not None:
print(" lowpass")
newData = signal.lfilter(self.filterb, self.filtera, channel)
if self.newSampleRate is not None:
print(" resampling")
newData = samplerate.resample(numpy.array(channel), resamplingFactor, 'sinc_best' if self.filterb is None else 'sinc_medium')
newChannels[i] = newData
self.channels = newChannels
if self.newSampleRate is not None:
self.sampleRate = self.newSampleRate
def test_multi_channel():
fs = 16000.
fr = 8000.
f0 = 440.
# Create a small stereo audio array with dephased channel
xleft = np.sin(2 * np.pi * f0/fs * np.arange(0, 2 * fs))
xright = np.cos(2 * np.pi * f0/fs * np.arange(0, 2 * fs))
y = np.empty((xleft.size, 2), np.float)
y[:,0] = xleft
y[:,1] = xright
# Upsampled reference
z_rleft = np.sin(2 * np.pi * f0/fr * np.arange(0, 2 * fr))
z_rright = np.cos(2 * np.pi * f0/fr * np.arange(0, 2 * fr))
z_r = np.vstack((z_rleft, z_rright)).T
z = resample(y, fr /fs, 'sinc_best')
assert np.max(np.abs(z_r[100:-100] - z[100:-99])) < 1e-2
def main(dbFilename, targetFs, force=False):
util.createDirectory(NormalizeDir)
rirDb = json.load(open(dbFilename))
bar = util.ConsoleProgressBar()
bar.start('Normalize RIRs')
i = 0
for rirId, rir in rirDb.items():
targetFilename = os.path.join(NormalizeDir, rir['id'] + '.wav')
if not force:
if rir['filename'] == targetFilename and \
rir['fs'] == targetFs and \
targetFilename:
continue
x, fs_x = sf.read(os.path.join(ImportDir, rir['id'] + '.wav'), dtype='float32')
y, fs_y = x, fs_x
if fs_y != targetFs:
y = resample(y, targetFs / fs_y, 'sinc_best')
fs_y = targetFs
rir['length_org'] = len(y) / fs_y
y = util.trimSilence(y, 0.001, trimRight=False)
y = util.normalizeAmplitude(y)
sf.write(targetFilename, y, fs_y)
rir['filename'] = targetFilename
rir['fs'] = fs_y
rir['length'] = len(y) / fs_y
i += 1
bar.progress(i / len(rirDb))
bar.end()
with open(dbFilename, 'w') as dbFile:
json.dump(rirDb, dbFile, sort_keys=True, indent=4)
def resample(self):
plotBA = False
if type(self.original) is list:
self.original = numpy.asarray(self.original)
if self.logs :
self.logs.write("Rec.py : resampling recording")
aa,b,c=mlab.specgram(self.original,NFFT=256,Fs=self.sample_rate)
to_sample = self.calc_resample_factor()
self.original = resample(self.original, float(to_sample)/float(self.sample_rate) , 'sinc_best')
if plotBA:
a,b,c=mlab.specgram(self.original,NFFT=256,Fs=to_sample)
figure(figsize=(25,15))
subplot(211)
imshow(20*log10(numpy.flipud(aa)), interpolation='nearest', aspect='auto')
subplot(212)
imshow(20*log10(numpy.flipud(a)),interpolation='nearest', aspect='auto')
savefig(''+self.filename+'.png', dpi=100)
close()
self.samples = len(self.original)
self.sample_rate = to_sample
def analyze(sig, samplerate=8000, resample_to=8000):
e_min_scale = 0.3
e_min_distance = 1
e_a_scale = 0.5
e_W_ms = 2000
min_len = 30
if samplerate != resample_to:
print("resampling")
sig = srate.resample(sig, resample_to / samplerate, 'sinc_best')
frames, frame_size = ad.frames(sig, samplerate, 64)
#ac_peaks = [ad.ac_peaks(frame) for frame in frames]
#energy = ad.energies(frames, ...)
#energy = log_energy
print("getting normalized spectra")
normalized_spectrum = ad.normalized_spectrum(frames, samplerate)
print("spectral entropy")
spectral_entropy = np.fromiter(
ad.entropy(frame) for frame in normalized_spectrum)
entopy_t = np.percentile(spectral_entropy, 80)
se_segments = ad.segments_to_seconds(
ad.entropy_segment_indexes(spectral_entropy, entropy_t))
print("energy based computations")
energy, smooth, en_a, en_t, en_lmin = ad.energy_thresholds(
sig,
noise_dist=e_min_distance,
a_scale=e_a_scale,
min_scale=e_min_scale,
W_ms=e_W_ms)
energy_segments = ad.get_voice_segments(
smooth, en_t, ad.get_segment_indexes(smooth, en_t, min_len=10))
#energy_t = ad.thresholds(frames)
#entropy_t = ad.entropy_t(frames)
#energy_indexes = ad.get_segment_indexes(x, t, min_len=30)
#entropy_indexes = ad.get_entropy_indexes(x, t, min_len=30)
#combination_indexes = ad.get_combined_indexes(x, t, min_len=30)
return spectral_entropy, se_t, se_segments, energy, en_t, energy_segments
def perform_mfcc_on_filename(filename, opts):
(samplerate, sig) = read_audio_to_numpy(filename)
opts['samplerate'] = samplerate
if sig.ndim > 1:
# Mix to mono
# TODO: Multi-channel
nchannels = sig.shape[1]
sig = n.mean(sig, axis=1)
else:
nchannels = 1
print "Read %s with sample rate %s, #channels = %d" % (filename, samplerate, nchannels)
if (samplerate != desired_samplerate and not FORCE_RESAMPLE):
print "%s has the wrong samplerate, ignoring" % filename
return None
if (samplerate != desired_samplerate and FORCE_RESAMPLE):
origsig = sig
sig = resample(origsig, 1.0 * desired_samplerate/samplerate, 'sinc_best')
print "Resampled file from rate %d to rate %d, shape %s to %s" % (samplerate, desired_samplerate, origsig.shape, sig.shape)
mfcc_feat = mfcc(sig, **opts)
print "Computed mfccs on %s" % filename
return mfcc_feat
def batchChordGenerate(taskList, prefix):
'''
inputs: a list of lists
each sublist contains the following values @ indices:
0 : vox collection index
1 : vox sample index
2 : a list: [sample #, transposition semitones, transposition cents]
3 : ... and so on ...
'''
for i, t in enumerate(taskList):
print "begin task for vocal gesture " + p(t[0]) + '_' + p(t[1])
# grab the vocal gesture:
gestur = vox[t[0]][t[1]]
# compute clicks to be applied in all files:
idx, l = clicks(gestur)
clk = [idx, l]
# compute envelope to be applied to all files:
env = envelope(gestur, 10, 30)
env *= (1 / env.max()) # normalize envelope
# proceed with computing:
fp = prefix + '_' + p(i) + '__' + p(t[0]) + '_' + p(t[1]) + '__'
sr = None
enc = None
for j in xrange(2, len(t)):
fname = fp + str(j - 2) + '__' + str(t[j][0]) + '.wav'
print " computing " + str(j) + ' : ' + fname
transpratio = tr(t[j][1], t[j][2])
# print " transposition : " + str(t[j][1]) + ' ' + str(t[j][2]) + ' = ' + str(transpratio)
rezs = [] # array of np.arrays containing transposed resonances
for f in coll_safe[t[j][0]]:
# print " file : " + f
x, sr, enc = wavread(f)
rezs.append(resample(x, transpratio, 'sinc_best'))
wavwrite(resonancesChord(None, rezs, 30, clk, env), fname, 44100,
'pcm24')
return
def preprocess(db, stations, comps, goal_day, params, responses=None):
"""
Fetches data for each ``stations`` and each ``comps`` using the
data_availability table in the database.
To correct for instrument responses, make sure to set ``remove_response``
to "Y" in the config and to provide the ``responses`` DataFrame.
:Example:
>>> from msnoise.api import connect, get_params, preload_instrument_responses
>>> from msnoise.preprocessing import preprocess
>>> db = connect()
>>> params = get_params(db)
>>> responses = preload_instrument_responses(db)
>>> st = preprocess(db, ["YA.UV06","YA.UV10"], ["Z",], "2010-09-01", params, responses)
>>> st
2 Trace(s) in Stream:
YA.UV06.00.HHZ | 2010-09-01T00:00:00.000000Z - 2010-09-01T23:59:59.950000Z | 20.0 Hz, 1728000 samples
YA.UV10.00.HHZ | 2010-09-01T00:00:00.000000Z - 2010-09-01T23:59:59.950000Z | 20.0 Hz, 1728000 samples
:type db: :class:`sqlalchemy.orm.session.Session`
:param db: A :class:`~sqlalchemy.orm.session.Session` object, as
obtained by :func:`msnoise.api.connect`.
:type stations: list of str
:param stations: a list of station names, in the format NET.STA.
:type comps: list of str
:param comps: a list of component names, in Z,N,E,1,2.
:type goal_day: str
:param goal_day: the day of data to load, ISO 8601 format: e.g. 2016-12-31.
:type params: class
:param params: an object containing the config parameters, as obtained by
:func:`msnoise.api.get_params`.
:type responses: :class:`pandas.DataFrame`
:param responses: a DataFrame containing the instrument responses, as
obtained by :func:`msnoise.api.preload_instrument_responses`.
:rtype: :class:`obspy.core.stream.Stream`
:return: A Stream object containing all traces.
"""
datafiles = {}
output = Stream()
for station in stations:
datafiles[station] = {}
net, sta = station.split('.')
gd = datetime.datetime.strptime(goal_day, '%Y-%m-%d')
files = get_data_availability(db,
net=net,
sta=sta,
starttime=gd,
endtime=gd)
for comp in comps:
datafiles[station][comp] = []
for file in files:
if file.comp[-1] not in comps:
continue
fullpath = os.path.join(file.path, file.file)
datafiles[station][file.comp[-1]].append(fullpath)
for istation, station in enumerate(stations):
net, sta = station.split(".")
for comp in comps:
files = eval("datafiles['%s']['%s']" % (station, comp))
if len(files) != 0:
logging.debug("%s.%s Reading %i Files" %
(station, comp, len(files)))
stream = Stream()
for file in sorted(files):
try:
st = read(file,
dytpe=np.float,
starttime=UTCDateTime(gd),
endtime=UTCDateTime(gd) + 86400)
except:
logging.debug("ERROR reading file %s" % file)
continue
for tr in st:
if len(tr.stats.channel) == 2:
tr.stats.channel += tr.stats.location
tr.stats.location = "00"
tmp = st.select(network=net, station=sta, component=comp)
if not len(tmp):
for tr in st:
tr.stats.network = net
st = st.select(network=net,
station=sta,
component=comp)
else:
st = tmp
for tr in st:
tr.data = tr.data.astype(np.float)
tr.stats.network = tr.stats.network.upper()
tr.stats.station = tr.stats.station.upper()
tr.stats.channel = tr.stats.channel.upper()
stream += st
del st
stream.sort()
try:
# HACK not super clean... should find a way to prevent the
# same trace id with different sps to occur
stream.merge(method=1,
interpolation_samples=3,
fill_value=None)
except:
logging.debug("Error while merging...")
traceback.print_exc()
continue
stream = stream.split()
if not len(stream):
continue
logging.debug("%s Checking sample alignment" % stream[0].id)
for i, trace in enumerate(stream):
stream[i] = check_and_phase_shift(trace)
logging.debug("%s Checking Gaps" % stream[0].id)
if len(getGaps(stream)) > 0:
max_gap = params.preprocess_max_gap * stream[
0].stats.sampling_rate
only_too_long = False
while getGaps(stream) and not only_too_long:
too_long = 0
gaps = getGaps(stream)
for gap in gaps:
if int(gap[-1]) <= max_gap:
try:
stream[gap[0]] = stream[gap[0]].__add__(
stream[gap[1]],
method=1,
fill_value="interpolate")
stream.remove(stream[gap[1]])
except:
stream.remove(stream[gap[1]])
break
else:
too_long += 1
if too_long == len(gaps):
only_too_long = True
stream = stream.split()
for tr in stream:
if tr.stats.sampling_rate < (params.goal_sampling_rate -
1):
stream.remove(tr)
taper_length = 20.0 # seconds
for trace in stream:
if trace.stats.npts < 4 * taper_length * trace.stats.sampling_rate:
stream.remove(trace)
else:
trace.detrend(type="demean")
trace.detrend(type="linear")
trace.taper(max_percentage=None, max_length=1.0)
if not len(stream):
logging.debug(" has only too small traces, skipping...")
continue
for trace in stream:
logging.debug("%s Highpass at %.2f Hz" %
(trace.id, params.preprocess_highpass))
trace.filter("highpass",
freq=params.preprocess_highpass,
zerophase=True)
if trace.stats.sampling_rate != params.goal_sampling_rate:
logging.debug("%s Lowpass at %.2f Hz" %
(trace.id, params.preprocess_lowpass))
trace.filter("lowpass",
freq=params.preprocess_lowpass,
zerophase=True,
corners=8)
if params.resampling_method == "Resample":
logging.debug(
"%s Downsample to %.1f Hz" %
(trace.id, params.goal_sampling_rate))
trace.data = resample(
trace.data, params.goal_sampling_rate /
trace.stats.sampling_rate, 'sinc_fastest')
elif params.resampling_method == "Decimate":
decimation_factor = trace.stats.sampling_rate / params.goal_sampling_rate
if not int(decimation_factor) == decimation_factor:
logging.warning(
"%s CANNOT be decimated by an integer factor, consider using Resample or Lanczos methods"
" Trace sampling rate = %i ; Desired CC sampling rate = %i"
% (trace.id, trace.stats.sampling_rate,
params.goal_sampling_rate))
sys.stdout.flush()
sys.exit()
logging.debug("%s Decimate by a factor of %i" %
(trace.id, decimation_factor))
trace.data = trace.data[::int(decimation_factor)]
elif params.resampling_method == "Lanczos":
logging.debug(
"%s Downsample to %.1f Hz" %
(trace.id, params.goal_sampling_rate))
trace.data = np.array(trace.data)
trace.interpolate(
method="lanczos",
sampling_rate=params.goal_sampling_rate,
a=1.0)
trace.stats.sampling_rate = params.goal_sampling_rate
if params.remove_response:
logging.debug('%s Removing instrument response' %
stream[0].id)
response = responses[responses["channel_id"] ==
stream[0].id]
if len(response) > 1:
response = response[
response["start_date"] <= UTCDateTime(gd)]
if len(response) > 1:
response = response[
response["end_date"] >= UTCDateTime(gd)]
elif len(response) == 0:
logging.info("No instrument response information "
"for %s, skipping" % stream[0].id)
continue
try:
datalesspz = response["paz"].values[0]
except:
logging.error("Bad instrument response information "
"for %s, skipping" % stream[0].id)
continue
stream.simulate(
paz_remove=datalesspz,
remove_sensitivity=True,
pre_filt=params.response_prefilt,
paz_simulate=None,
)
for tr in stream:
tr.data = tr.data.astype(np.float32)
output += stream
del stream
del files
clean_scipy_cache()
return output
def _downsample_and_upsample(data):
ds = numpy.array([resample(x, 0.5, 'sinc_best') for x in data[0]])
us = numpy.array([resample(x, 2, 'sinc_best') for x in ds])
return (us, )
def preprocess(db, stations, comps, goal_day, params, responses=None):
datafiles = {}
output = Stream()
for station in stations:
datafiles[station] = {}
net, sta = station.split('.')
gd = datetime.datetime.strptime(goal_day, '%Y-%m-%d')
files = get_data_availability(db,
net=net,
sta=sta,
starttime=gd,
endtime=gd)
for comp in comps:
datafiles[station][comp] = []
for file in files:
if file.comp[-1] not in comps:
continue
fullpath = os.path.join(file.path, file.file)
datafiles[station][file.comp[-1]].append(fullpath)
for istation, station in enumerate(stations):
net, sta = station.split(".")
for comp in comps:
files = eval("datafiles['%s']['%s']" % (station, comp))
if len(files) != 0:
logging.debug("%s.%s Reading %i Files" %
(station, comp, len(files)))
stream = Stream()
for file in sorted(files):
st = read(file,
dytpe=np.float,
starttime=UTCDateTime(gd),
endtime=UTCDateTime(gd) + 86400)
tmp = st.select(network=net, station=sta, component=comp)
if not len(tmp):
for tr in st:
tr.stats.network = net
st = st.select(network=net,
station=sta,
component=comp)
else:
st = tmp
for tr in st:
tr.data = tr.data.astype(np.float)
stream += st
del st
stream.sort()
stream.merge(method=1,
interpolation_samples=3,
fill_value=None)
stream = stream.split()
logging.debug("Checking sample alignment")
for i, trace in enumerate(stream):
stream[i] = check_and_phase_shift(trace)
logging.debug("Checking Gaps")
if len(getGaps(stream)) > 0:
max_gap = 10
only_too_long = False
while getGaps(stream) and not only_too_long:
too_long = 0
gaps = getGaps(stream)
for gap in gaps:
if int(gap[-1]) <= max_gap:
stream[gap[0]] = stream[gap[0]].__add__(
stream[gap[1]],
method=1,
fill_value="interpolate")
stream.remove(stream[gap[1]])
break
else:
too_long += 1
if too_long == len(gaps):
only_too_long = True
stream = stream.split()
taper_length = 20.0 # seconds
for trace in stream:
if trace.stats.npts < 4 * taper_length * trace.stats.sampling_rate:
stream.remove(trace)
else:
trace.detrend(type="demean")
trace.detrend(type="linear")
trace.taper(max_percentage=None, max_length=1.0)
if not len(stream):
logging.debug(" has only too small traces, skipping...")
continue
for trace in stream:
logging.debug("%s.%s Highpass at %.2f Hz" %
(station, comp, params.preprocess_highpass))
trace.filter("highpass",
freq=params.preprocess_highpass,
zerophase=True)
if trace.stats.sampling_rate != params.goal_sampling_rate:
logging.debug(
"%s.%s Lowpass at %.2f Hz" %
(station, comp, params.preprocess_lowpass))
trace.filter("lowpass",
freq=params.preprocess_lowpass,
zerophase=True,
corners=8)
if params.resampling_method == "Resample":
logging.debug(
"%s.%s Downsample to %.1f Hz" %
(station, comp, params.goal_sampling_rate))
trace.data = resample(
trace.data, params.goal_sampling_rate /
trace.stats.sampling_rate, 'sinc_fastest')
elif params.resampling_method == "Decimate":
decimation_factor = trace.stats.sampling_rate / params.goal_sampling_rate
if not int(decimation_factor) == decimation_factor:
logging.warning(
"%s.%s CANNOT be decimated by an integer factor, consider using Resample or Lanczos methods"
" Trace sampling rate = %i ; Desired CC sampling rate = %i"
%
(station, comp, trace.stats.sampling_rate,
params.goal_sampling_rate))
sys.stdout.flush()
sys.exit()
logging.debug("%s.%s Decimate by a factor of %i" %
(station, comp, decimation_factor))
trace.data = trace.data[::int(decimation_factor)]
elif params.resampling_method == "Lanczos":
logging.debug(
"%s.%s Downsample to %.1f Hz" %
(station, comp, params.goal_sampling_rate))
trace.data = np.array(trace.data)
trace.interpolate(
method="lanczos",
sampling_rate=params.goal_sampling_rate,
a=1.0)
trace.stats.sampling_rate = params.goal_sampling_rate
if get_config(db, 'remove_response', isbool=True):
logging.debug('%s Removing instrument response' %
stream[0].id)
response_prefilt = eval(get_config(db, 'response_prefilt'))
response = responses[responses["channel_id"] ==
stream[0].id]
if len(response) > 1:
response = response[
response["start_date"] < UTCDateTime(gd)]
response = response[
response["end_date"] > UTCDateTime(gd)]
elif len(response) == 0:
logging.info("No instrument response information "
"for %s, exiting" % stream[0].id)
sys.exit()
datalesspz = response["paz"].values[0]
stream.simulate(
paz_remove=datalesspz,
remove_sensitivity=True,
pre_filt=response_prefilt,
paz_simulate=None,
)
for tr in stream:
tr.data = tr.data.astype(np.float32)
output += stream
del stream
del files
clean_scipy_cache()
return 0, output
