Python wavfile.write方法代码示例

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def write_file(output_file_path, input_file_name, name_attribute, sig, fs):
    """
    Read wave file as mono.

    Args:
        - output_file_path (str) : path to save resulting wave file to.
        - input_file_name  (str) : name of processed wave file,
        - name_attribute   (str) : attribute to add to output file name.
        - sig            (array) : signal/audio array.
        - fs               (int) : sampling rate.

    Returns:
        tuple of sampling rate and audio data.
    """
    # set-up the output file name
    fname = os.path.basename(input_file_name).split(".wav")[0] + name_attribute
    fpath = os.path.join(output_file_path, fname)
    write(filename=fpath, rate=fs, data=sig)
    print("Writing data to " + fpath + ".") 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def synthesize(frames,filename,stride,sr=16000,deemph=0,ymax=0.98,normalize=False):
    # Generate stream
    y=torch.zeros((len(frames)-1)*stride+len(frames[0]))
    for i,x in enumerate(frames):
        y[i*stride:i*stride+len(x)]+=x
    # To numpy & deemph
    y=y.numpy().astype(np.float32)
    if deemph>0:
        y=deemphasis(y,alpha=deemph)
    # Normalize
    if normalize:
        y-=np.mean(y)
        mx=np.max(np.abs(y))
        if mx>0:
            y*=ymax/mx
    else:
        y=np.clip(y,-ymax,ymax)
    # To 16 bit & save
    wavfile.write(filename,sr,np.array(y*32767,dtype=np.int16))
    return y

######################################################################################################################## 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def split_to_mix(audio_path_list,database_repo=DATABASE_REPO_PATH,partition=2):
    # return split_list : (part1,part2,...)
    # each part : (idx,path)
    length = len(audio_path_list)
    part_len = length // partition
    head = 0
    part_idx = 0
    split_list = []
    while((head+part_len)<length):
        part = audio_path_list[head:(head+part_len)]
        split_list.append(part)
        with open('%s/single_TF_part%d.txt'%(database_repo,part_idx),'a') as f:
            for idx, _ in part:
                name = 'single-%05d' % idx
                f.write('%s.npy' % name)
                f.write('\n')
        head += part_len
        part_idx += 1
    return split_list

# mix single TF data 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def wavWrite(y, fs, nbits, audioFile):
		""" Write samples to WAV file
        Args:
            samples: (ndarray / 2D ndarray) (floating point) sample vector
                    	mono: DIM: nSamples
                    	stereo: DIM: nSamples x nChannels

            fs: 	(int) Sample rate in Hz
            nBits: 	(int) Number of bits
            fnWAV: 	(string) WAV file name to write
		"""
		if nbits == 8:
			intsamples = (y+1.0) * AudioIO.normFact['int' + str(nbits)]
			fX = np.int8(intsamples)
		elif nbits == 16:
			intsamples = y * AudioIO.normFact['int' + str(nbits)]
			fX = np.int16(intsamples)
		elif nbits > 16:
			fX = y

		write(audioFile, fs, fX) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def sound(x,fs):
		""" Plays a wave file using the pyglet library. But first, it has to be written.
			Termination of the playback is being performed by any keyboard input and Enter.
			Args:
			x: 		   (array) Floating point samples
			fs:		   (int) The sampling rate
		"""
		import pyglet as pg
		global player
		# Call the writing function
		AudioIO.wavWrite(x, fs, 16, 'testPlayback.wav')
		# Initialize playback engine
		player = pg.media.Player()
		# Initialize the object with the audio file
		playback = pg.media.load('testPlayback.wav')
		# Set it to player
		player.queue(playback)
		# Sound call
		player.play()
		# Killed by "keyboard"
		kill = raw_input()
		if kill or kill == '':
			AudioIO.stop()
		# Remove the dummy wave write
		os.remove('testPlayback.wav') 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def _wavdata_rs(self):
        if self.fs_rs is not None:
            # Number of points in resample
            ns_rs = np.int_(np.ceil(self.ns * self.fs_rs / self.fs))
            # Do resample
            # XXX: Tried using a Hamming window as a low pass filter, but it
            #      didn't seem to make a big difference, so it's not used
            #      here.
            data_rs = resample(self.wavdata, ns_rs)
            wavpath_rs = self.wavpath.split('.')[0] + '-resample-' + str(self.fs_rs) + 'Hz.wav'
            # Write resampled data to wav file
            # Convert data from 32-bit floating point to 16-bit PCM
            data_rs_int = np.int16(data_rs * 32768)
            wavfile.write(wavpath_rs, self.fs_rs, data_rs_int)
            # XXX: Was worried that Python might continue executing code
            #      before the file write is finished, but it seems like it's
            #      not an issue.
            return wavpath_rs, data_rs, data_rs_int, ns_rs
        else:
            return None, None, None, None 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def run_phase_reconstruction_example():
    fs, d = fetch_sample_speech_tapestry()
    # actually gives however many components you say! So double what .m file
    # says
    fftsize = 512
    step = 64
    X_s = np.abs(stft(d, fftsize=fftsize, step=step, real=False,
                      compute_onesided=False))
    X_t = iterate_invert_spectrogram(X_s, fftsize, step, verbose=True)

    """
    import matplotlib.pyplot as plt
    plt.specgram(d, cmap="gray")
    plt.savefig("1.png")
    plt.close()
    plt.imshow(X_s, cmap="gray")
    plt.savefig("2.png")
    plt.close()
    """

    wavfile.write("phase_original.wav", fs, soundsc(d))
    wavfile.write("phase_reconstruction.wav", fs, soundsc(X_t)) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def run_fft_dct_example():
    random_state = np.random.RandomState(1999)

    fs, d = fetch_sample_speech_fruit()
    n_fft = 64
    X = d[0]
    X_stft = stft(X, n_fft)
    X_rr = complex_to_real_view(X_stft)
    X_dct = fftpack.dct(X_rr, axis=-1, norm='ortho')
    X_dct_sub = X_dct[1:] - X_dct[:-1]
    std = X_dct_sub.std(axis=0, keepdims=True)
    X_dct_sub += .01 * std * random_state.randn(
        X_dct_sub.shape[0], X_dct_sub.shape[1])
    X_dct_unsub = np.cumsum(X_dct_sub, axis=0)
    X_idct = fftpack.idct(X_dct_unsub, axis=-1, norm='ortho')
    X_irr = real_to_complex_view(X_idct)
    X_r = istft(X_irr, n_fft)[:len(X)]

    SNR = 20 * np.log10(np.linalg.norm(X - X_r) / np.linalg.norm(X))
    print(SNR)

    wavfile.write("fftdct_orig.wav", fs, soundsc(X))
    wavfile.write("fftdct_rec.wav", fs, soundsc(X_r)) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def run_ltsd_example():
    fs, d = fetch_sample_speech_tapestry()
    winsize = 1024
    d = d.astype("float32") / 2 ** 15
    d -= d.mean()

    pad = 3 * fs
    noise_pwr = np.percentile(d, 1) ** 2
    noise_pwr = max(1E-9, noise_pwr)
    d = np.concatenate((np.zeros((pad,)) + noise_pwr * np.random.randn(pad), d))
    _, vad_segments = ltsd_vad(d, fs, winsize=winsize)
    v_up = np.where(vad_segments == True)[0]
    s = v_up[0]
    st = v_up[-1] + int(.5 * fs)
    d = d[s:st]

    bname = "tapestry.wav".split(".")[0]
    wavfile.write("%s_out.wav" % bname, fs, soundsc(d)) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def to_wav(filename,rate,x):
    """
    Write a wave file.

    A wrapper function for scipy.io.wavfile.write
    that also includes int16 scaling and conversion.
    Assume input x is [-1,1] values.

    Parameters
    ----------
    filename : file name string
    rate : sampling frequency in Hz

    Returns
    -------
    Nothing : writes only the *.wav file

    Examples
    --------
    >>> to_wav('test_file.wav', 8000, x)

    """
    x16 = np.int16(x*32767)
    wavfile.write(filename, rate, x16) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def getMusicSegmentsFromFile(inputFile):	
	modelType = "svm"
	modelName = "data/svmMovies8classes"
	
	dirOutput = inputFile[0:-4] + "_musicSegments"
	
	if os.path.exists(dirOutput) and dirOutput!=".":
		shutil.rmtree(dirOutput)	
	os.makedirs(dirOutput)	
	
	[Fs, x] = audioBasicIO.readAudioFile(inputFile)	

	if modelType=='svm':
		[Classifier, MEAN, STD, classNames, mtWin, mtStep, stWin, stStep, compute_beat] = aT.load_model(modelName)
	elif modelType=='knn':
		[Classifier, MEAN, STD, classNames, mtWin, mtStep, stWin, stStep, compute_beat] = aT.load_model_knn(modelName)

	flagsInd, classNames, acc, CM = aS.mtFileClassification(inputFile, modelName, modelType, plotResults = False, gtFile = "")
	segs, classes = aS.flags2segs(flagsInd, mtStep)

	for i, s in enumerate(segs):
		if (classNames[int(classes[i])] == "Music") and (s[1] - s[0] >= minDuration):
			strOut = "{0:s}{1:.3f}-{2:.3f}.wav".format(dirOutput+os.sep, s[0], s[1])	
			wavfile.write( strOut, Fs, x[int(Fs*s[0]):int(Fs*s[1])]) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def annotation2files(wavFile, csvFile):
    '''
        Break an audio stream to segments of interest, 
        defined by a csv file
        
        - wavFile:    path to input wavfile
        - csvFile:    path to csvFile of segment limits
        
        Input CSV file must be of the format <T1>\t<T2>\t<Label>
    '''    
    
    [Fs, x] = audioBasicIO.read_audio_file(wavFile)
    with open(csvFile, 'r') as csvfile:
        reader = csv.reader(csvfile, delimiter='\t', quotechar='|')
        for j, row in enumerate(reader):
            T1 = float(row[0].replace(",","."))
            T2 = float(row[1].replace(",","."))            
            label = "%s_%s_%.2f_%.2f.wav" % (wavFile, row[2], T1, T2)
            label = label.replace(" ", "_")
            xtemp = x[int(round(T1*Fs)):int(round(T2*Fs))]            
            print(T1, T2, label, xtemp.shape)
            wavfile.write(label, Fs, xtemp) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def BeamformIt(mixture, fs=8000, basedir='/Data/software/BeamformIt/', verbose=False):
    mixture /= np.max(np.abs(mixture))

    if not os.path.exists('/tmp/audios/'):
        os.mkdir('/tmp/audios/')

    wavfile.write('/tmp/audios/rec.wav', fs, mixture.T)

    p = subprocess.Popen("cd {}; bash do_beamforming.sh /tmp/audios/ temps".format(basedir),
                         stdout=subprocess.PIPE, shell=True)

    (output, err) = p.communicate()
    p_status = p.wait()
    ref_ch = int(str(output).split("Selected channel ")[1].split(' as the reference channel')[0])
    if verbose:
        print("Output: {}".format(output))
        print("Error: {}".format(err))
        print("Status: {}".format(p_status))

    s, _ = sf.read('{}/output/temps/temps.wav'.format(basedir))

    return s, ref_ch 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def generate_and_save_samples(sample_fn, length, count, dir, rate, levels):
    def save_samples(data):
        data = (data * np.reshape(np.arange(levels) / (levels-1), [levels, 1, 1])).sum(
            axis=1, keepdims=True)
        value = np.iinfo(np.int16).max
        audio = (utils.inverse_mulaw(data * 2 - 1) * value).astype(np.int16)
        for idx, sample in enumerate(audio):
            filename = os.path.join(dir, 'sample_{}.wav'.format(idx))
            wavfile.write(filename, rate, np.squeeze(sample))

    samples = chainer.Variable(
        chainer.cuda.cupy.zeros([count, levels, 1, length], dtype='float32'))
    one_hot_ref = chainer.cuda.cupy.eye(levels).astype('float32')

    with tqdm.tqdm(total=length) as bar:
        for i in range(length):
            probs = F.softmax(sample_fn(samples))[:, :, 0, 0, i]
            samples.data[:, :, 0, i] = one_hot_ref[utils.sample_from(probs.data.get())]
            bar.update()

    samples.to_cpu()
    save_samples(samples.data) 

# 需要导入模块: from scipy.io import wavfile [as 别名]
# 或者: from scipy.io.wavfile import write [as 别名]
def save_wav(audio, output_wav_file):
    wav.write(output_wav_file, 16000, np.array(np.clip(np.round(audio), -2**15, 2**15-1), dtype=np.int16))
    print('output dB', db(audio))