コード例 #1
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    def gain(self, level, mode: 'multiply'):
        ''' Increases the volume of the audio by a specified amount '''

        amplifier = {
            'multiply':
            lambda d, l: d * l,
            'percent':
            lambda d, l: d * l / 100.0,
            'additive':
            lambda d, l: d + l,
            'subtractive':
            lambda d, l: d - l,
            'dB_additive':
            lambda d, l: librosa.db_to_amplitude(
                (librosa.amplitude_to_db(d) + l)),
            'dB_subtractive':
            lambda d, l: librosa.db_to_amplitude(
                (librosa.amplitude_to_db(d) - l)),
            'dB_multiply':
            lambda d, l: (10**(l / 20)) * d
        }

        data = amplifier[mode](self.data, level)
        return Audio().populate(
            data, self.sample_rate, self.source_path,
            self.operations + [_gainOperation(level, mode)])
コード例 #2
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def pesq_on_batch(y_denoised, ytest, test_phase, sr=16000):

    pesqvalue = 1
    try:

        y_denoised = np.squeeze(y_denoised, axis=3)
        y_denoised = np.squeeze(y_denoised, axis=0)

        y_denoised = librosa.db_to_amplitude(y_denoised)
        ytest = librosa.db_to_amplitude(ytest)

        denoised = y_denoised * test_phase
        original = ytest * test_phase

        denoised = librosa.istft(denoised)
        original = librosa.istft(original)

        #print(denoised)
        #print(original)

        denoised = librosa.util.normalize(denoised)
        original = librosa.util.normalize(original)

        #pesqvalue=pesq(sr, original, denoised, 'wb')

        pesqvalue = pesq(sr, original, denoised, 'wb')
        #print(pesqvalue)
    except:
        print("pesq didnt work")
        presqvalue = 1

    return pesqvalue
コード例 #3
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def stoi_on_batch(y_denoised,ytest,test_phase,sr=16000):

    stoivalue=0

    y_denoised = np.squeeze(y_denoised,axis=3)
    y_denoised = np.squeeze(y_denoised,axis=0)



    y_denoised = librosa.db_to_amplitude(y_denoised)
    ytest = librosa.db_to_amplitude(ytest)

    denoised = y_denoised*test_phase
    original = ytest*test_phase

    denoised = librosa.istft(denoised)
    original = librosa.istft(original)

    #print(denoised)
    #print(original)

    denoised = librosa.util.normalize(denoised)
    original = librosa.util.normalize(original)

    #pesqvalue=pesq(sr, original, denoised, 'wb')


    stoivalue=stoi( original, denoised,sr, 'wb')
    #print(pesqvalue)

    #print("stoi didnt work")
    #stoivalue=0

    return stoivalue
def pesq_from_fft(noisy,phase_noisy,clean,phase_clean,out=False):
     """
     Calculate PESQ Metric on stft batch
     """
     phase_noisy=np.array(phase_noisy)
     noisy=librosa.db_to_amplitude(noisy)

     noisy=noisy*phase_noisy
     noisy = librosa.istft(noisy)

     clean=np.array(clean)
     phase_clean=np.array(phase_clean)
     clean=librosa.db_to_amplitude(clean)
     clean=clean*phase_clean
     clean = librosa.istft(clean)

     if out==True:
         global cn

         scipy.io.wavfile.write(path+'\\gvepre\\predictGVE'+str(cn)+'.wav',16000,noisy)
         cn=cn+1

     sr =16000

     pesqvalue=pesq(sr, clean, noisy, 'wb')
     #print(pesqvalue)
     return pesqvalue
コード例 #5
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def source_to_distortion(batch_predicted,target_gt):

    batch_predicted = librosa.db_to_amplitude(batch_predicted)
    target_gt = librosa.db_to_amplitude(target_gt)

    distortion = (batch_predicted-target_gt)**2

    return 10*np.log10(np.divide(target_gt, distortion, out=(np.ones_like(Noisy))*50, where=distortion!=0))
コード例 #6
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def extract_audio(
        Z, feature,
        params):  # if normalized Z: unnormalize first, then pass to func.

    # convert to audio
    if feature == "Stft":
        # undo log-magnitude scaling
        S = librosa.db_to_amplitude(Z)

        # upsample
        S = _upsample_fft(S, params["fft_sample_rate"],
                          params["stft_window_length"])

        yhat = librosa.griffinlim(S, hop_length=params["stft_hop_length"])

    elif feature == "Mel":
        # undo log-power scaling
        S = librosa.db_to_power(Z)

        yhat = librosa.feature.inverse.mel_to_audio(
            S,
            sr=params["fft_sample_rate"],
            n_fft=params["stft_window_length"],
            hop_length=params["stft_hop_length"],
        )

    elif feature == "Cqt":
        # undo log-amplitude scaling
        S = librosa.db_to_amplitude(Z)

        yhat = librosa.griffinlim_cqt(
            S,
            sr=params["fft_sample_rate"],
            hop_length=params["stft_hop_length"],
            fmin=librosa.note_to_hz(params["cqt_min_frequency"]),
        )

    elif feature == "Mfcc":

        yhat = librosa.feature.inverse.mfcc_to_audio(
            Z,
            n_mels=params["frequency_bins"],
            sr=params["fft_sample_rate"],
            n_fft=params["stft_window_length"],
            hop_length=params["stft_hop_length"],
        )

    else:
        print("Error: feature invalid")
        # throw/raise something
        return -1

    return yhat, params["fft_sample_rate"]
コード例 #7
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def spectrogram_to_audio_db(data, data_recon, output_dir):
    data_np = (data.squeeze(0).to(torch.device("cpu"))).detach().numpy()
    data_np = librosa.db_to_amplitude(data_np)
    data_griffin_lim = librosa.griffinlim(data_np)
    data_recon_np = (data_recon.squeeze(0).to(
        torch.device("cpu"))).detach().numpy()
    data_recon_np = librosa.db_to_amplitude(data_recon_np)
    data_recon_griffin_lim = librosa.griffinlim(data_recon_np)

    source_aud_path = output_dir + '_input_' + '.wav'
    target_aud_path = output_dir + '_output_' + '.wav'

    librosa.output.write_wav(source_aud_path, data_griffin_lim, 16384)
    librosa.output.write_wav(target_aud_path, data_recon_griffin_lim, 16384)
    return source_aud_path, target_aud_path
コード例 #8
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def main():

    sample_rate = 44100

    n_fft = 512
    n_frame = 1000

    # create arbitrary spectrogram as Numpy array
    # for convenience, create spectrogram in range of dB and convert to power

    spec = np.random.normal(-20, 2, [1 + n_fft // 2, n_frame])
    spec[20:30] += 50

    spec = librosa.db_to_amplitude(spec)

    # plot created spectrogram

    librosa.display.specshow(librosa.amplitude_to_db(spec))
    plt.colorbar()
    plt.show()

    # inverse STFT and playback

    wave = librosa.istft(spec)
    print('wave shape: {}'.format(wave.shape))

    sd.play(wave, samplerate=sample_rate, blocking=True)
コード例 #9
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def mel_spectro2wav(mel_spectro, preemphasize=hparams.PREEMPHASIZE,
                    ref_db=hparams.REF_DB,
                    max_db=hparams.MAX_DB,
                    n_iter_griffin_lim=hparams.N_ITER_GRIFFIN_LIM,
                    gl_power=hparams.GL_POWER,
                    sample_rate=hparams.SAMPLE_RATE,
                    n_fft=hparams.N_FFT,
                    n_mels=hparams.SYNTHESIZER_N_MELS,
                    hop_length=hparams.HOP_LENGTH,
                    win_length=hparams.WIN_LENGTH,
                    window=hparams.WINDOW):
    mel_spectro = mel_spectro.T
    mel_spectro = (np.clip(mel_spectro, 0, 1) * max_db) - max_db + ref_db
    amp_mel = librosa.db_to_amplitude(mel_spectro)
    inv_mel_basis = np.linalg.pinv(librosa.filters.mel(sample_rate, n_fft=n_fft, n_mels=n_mels))
    mag_spectro = np.maximum(1e-10, np.dot(inv_mel_basis, amp_mel))
    mag_spectro = mag_spectro ** gl_power
    wav = griffin_lim(mag_spectro,
                      n_iter_griffin_lim=n_iter_griffin_lim,
                      n_fft=n_fft,
                      hop_length=hop_length,
                      win_length=win_length,
                      window=window)
    wav = signal.lfilter([1], [1, -preemphasize], wav)
    wav, _ = librosa.effects.trim(wav)
    return wav.astype(np.float32)
コード例 #10
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ファイル: filter.py プロジェクト: taalua/hifi-gan-denoising
def random_eq(audio, fs, fraction, order, limits_freq, distribtion):
    bands, _, _ = filterbank(audio, fs, fraction, order, limits_freq)
    bands = np.stack(bands, axis=0)
    coeffs = distribtion(size=(bands.shape[0], 1))
    eq_coeffs = librosa.db_to_amplitude(coeffs)
    bands *= eq_coeffs
    return librosa.util.normalize(np.sum(bands, axis=0))
コード例 #11
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ファイル: similarity.py プロジェクト: w0lramD/pattern-radio
def fetch_spectrogram_image(row):
    """ Fetch spectrogram at specific path from cloud storage """

    try:
        filename = row['filename'].replace('.mp3', '.wav') + ".png"
        # filename = row['filename']
        logging.info("Fetching " + filename)
        blob = pool_bucket.get_blob(filename)
        if not blob:
            logging.error(filename + " not found")

        tmp = tempfile.NamedTemporaryFile(suffix=".png")
        blob.download_to_file(tmp)
        tmp.seek(0)
        img = np.asarray(Image.open(tmp))

        assert blob.metadata['db_min']
        assert blob.metadata['db_max']

        img_mapped = np.interp(
            img, (0, 255),
            (int(blob.metadata['db_min']), int(blob.metadata['db_max'])))

        cqt = librosa.db_to_amplitude(img_mapped)
        cqt = np.flipud(cqt).T

        tmp.close()

        return [cqt, blob.metadata]
    except Exception as e:
        logging.error("Could not download " + filename)
        logging.error(e)
        return
コード例 #12
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ファイル: utils.py プロジェクト: Jerryhai/tacotron2
def spectrogram2wav(mag):
    '''# Generate wave file from spectrogram'''
    # transpose
    mag = mag.T

    # de-noramlize
    mag = (np.clip(mag, 0, 1) * hp.max_db) - hp.max_db + hp.ref_db

    # to amplitude
    mag = librosa.db_to_amplitude(mag)
    # print(np.max(mag), np.min(mag), mag.shape)
    # (1025, 812, 16)

    # wav reconstruction
    wav = griffin_lim(mag)

    #wav = butter_bandpass_filter(wav, hp.lowcut, hp.highcut, hp.sr, order=6)

    # de-preemphasis
    wav = signal.lfilter([1], [1, -hp.preemphasis], wav)

    # trim
    wav, _ = librosa.effects.trim(wav)

    return wav
コード例 #13
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def postpro_and_gen(S,
                    phase,
                    returnS=0,
                    dBscale=1,
                    denormalize=1,
                    complex_phase=0,
                    clip_phase=0):  # T, F
    if dBscale:
        if denormalize:
            # denormalization
            S = S * hparams.max_db - hparams.max_db + hparams.ref_db
        S = librosa.db_to_amplitude(S)

    # pad with 0
    Sfull = np.concatenate((S, np.zeros(shape=(S.shape[0], 1))), axis=-1)

    if clip_phase:
        phase = np.concatenate((phase, np.zeros(shape=(2, phase.shape[1], 1))),
                               axis=-1)

    # generate waveform
    wav = genWaveclip(Sfull, phase, complex_phase)
    if not returnS:
        return wav
    else:
        return wav, Sfull, phase
コード例 #14
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 def extract_loudness(self, audio):
     S = librosa.stft(audio)
     power = np.abs(S)**2
     p_mean = np.sum(power, axis=0, keepdims=True)
     db = librosa.power_to_db(p_mean, ref=np.max(power))
     amp = librosa.db_to_amplitude(db)
     return amp[0]
コード例 #15
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def make_test_folder(X_in, X_out, name, step, n_samples=5):
    path_to_step_folder = mkdir(path_to_log, str(step + 1))
    path_to_step_folder_name = mkdir(path_to_step_folder, name)
    for i in range(n_samples):
        plt.subplot(2, n_samples, 1 + i)
        plt.axis('off')
        plt.imshow(X_in[i].reshape(spec_dim, spec_dim), cmap='gray')
        imageio.imwrite(
            path.join(path_to_step_folder_name,
                      str(i) + "_real_" + name.split('_')[0] + ".jpg"),
            X_in[i].reshape(spec_dim, spec_dim))
        im = cv2.imread(
            path.join(path_to_step_folder_name,
                      str(i) + "_real_" + name.split('_')[0] + ".jpg"), -1)
        im = im[:fft_len // 2 + 1, :fft_len // 2 + 1]
        im = (im * 80.0 / 255.0) - 80.0
        im = librosa.db_to_amplitude(im)
        y2 = griffinlim(im, hop_length=hop_length)
        write(
            path.join(path_to_step_folder_name,
                      str(i) + "_real_" + name.split('_')[0] + ".wav"), 16000,
            y2 * 1.5)
    # plot translated image
    for i in range(n_samples):
        plt.subplot(2, n_samples, 1 + n_samples + i)
        plt.axis('off')
        plt.imshow(X_out[i].reshape(spec_dim, spec_dim), cmap='gray')
        imageio.imwrite(
            path.join(path_to_step_folder_name,
                      str(i) + "_generated_" + name.split('_')[2] + ".jpg"),
            X_out[i].reshape(spec_dim, spec_dim))
        im = cv2.imread(
            path.join(path_to_step_folder_name,
                      str(i) + "_generated_" + name.split('_')[2] + ".jpg"),
            -1)
        im = im[:fft_len // 2 + 1, :fft_len // 2 + 1]
        im = (im * 80.0 / 255.0) - 80.0
        im = librosa.db_to_amplitude(im)
        y2 = griffinlim(im, hop_length=hop_length)
        write(
            path.join(path_to_step_folder_name,
                      str(i) + "_generated_" + name.split('_')[2] + ".wav"),
            16000, y2 * 1.5)
    # save plot to file
    filename1 = '%s_generated_plot_%06d.png' % (name, (step + 1))
    plt.savefig(path.join(path_to_step_folder, filename1), dpi=300)
    plt.close()
コード例 #16
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ファイル: dsp.py プロジェクト: taalua/hifi-gan-denoising
def extend_envelope(envelope_estimated, noise_floor_onset):
    # Fit straight line in dB scaling, return exponential, extended envelope.

    env_db = librosa.amplitude_to_db(envelope_estimated, ref=1.0)
    a = (env_db[noise_floor_onset] - env_db[0]) / np.max(
        (noise_floor_onset, 1.))
    b = env_db[0]
    return librosa.db_to_amplitude(a * np.arange(len(envelope_estimated)) + b)
コード例 #17
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def inv_spectrogram_librosa(spectrogram, fs, hparams):
    """Converts spectrogram to waveform using librosa"""
    S_denorm = _denormalize(spectrogram, hparams)
    S = librosa.db_to_amplitude(
        S_denorm + hparams.ref_level_db
    )  # Convert back to linear
    # Reconstruct phase
    return griffinlim_librosa(S, fs, hparams)
コード例 #18
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def SDR(original, predicted):

    original = librosa.db_to_amplitude(original)
    predicted = librosa.db_to_amplitude(predicted)

    distortion = predicted - original

    original = original**2
    distortion = distortion**2

    sdr = np.divide(original, distortion)

    sdr = np.nan_to_num(sdr, nan=60.0, posinf=60.0, neginf=60.0)
    #print(sdr)
    sdr = 10 * np.log10(sdr)

    return np.mean(sdr)
コード例 #19
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def SNR(Noisy, Clean, mask=True, Bark=False):
    """Function to Calculate Signal-to-Noise Ratio, mask==True puts out IBM"""
    m_ibm = []

    if Bark == False:
        Noisy = librosa.db_to_amplitude(Noisy)
        Clean = librosa.db_to_amplitude(Clean)

    N = np.subtract(Noisy, Clean)

    m_ibm = 20 * np.log10(
        np.divide(Clean, N, out=np.zeros_like(Noisy), where=N != 0))

    print("masking output")
    if mask == True:
        m_ibm = (m_ibm >= 0).astype(int)
    return m_ibm
コード例 #20
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def generateWavFromSTFT(amp, phase, wl, hl):

    Mdb_inormed = np.interp(amp, (amp.min(), amp.max()), (-15, 65))
    iM = librosa.db_to_amplitude(Mdb_inormed)
    iC = iM * np.exp(1j * phase)

    iy = librosa.istft(iC, hop_length=hl, win_length=wl)
    iy = librosa.util.normalize(iy)
    return iy
コード例 #21
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def spec_to_audio(X_out, name_gen):
    imageio.imwrite(path.join(path_to_results, name_gen + ".jpg"),
                    X_out.reshape(260, 260))
    im = cv2.imread(path.join(path_to_results, name_gen + ".jpg"), -1)
    im = im[:257, :257]
    im = (im * 80.0 / 255.0) - 80.0
    im = librosa.db_to_amplitude(im)
    y2 = griffinlim(im, hop_length=256)
    write(path.join(path_to_results, name_gen + ".wav"), 16000, y2 * 3)
コード例 #22
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def do_convert(predictor, input_name, logdir2):
    convert_s = datetime.datetime.now()

    # Load input audio
    input_audio, _ = librosa.load(input_name, sr=hp.default.sr, dtype=np.float64)

    # Extract F0 from input audio first
    input_f0, t_table = pw.dio(input_audio, hp.default.sr)
    input_f0 = pw.stonemask(input_audio, input_f0, t_table, hp.default.sr)

    # Get MFCC, Spectral Envelope, and Aperiodicity
    mfcc = _get_mfcc(input_audio, hp.default.n_fft, hp.default.win_length, hp.default.hop_length)
    mfcc = np.expand_dims(mfcc, axis=0)

    input_ap = pw.d4c(input_audio, input_f0, t_table, hp.default.sr, fft_size=hp.default.n_fft)

    input_sp_en = _get_spectral_envelope(preemphasis(input_audio, coeff=hp.default.preemphasis), hp.default.n_fft)
    plt.imsave('./converted/debug/input_sp_en_original.png', input_sp_en, cmap='binary')
    input_sp_en = np.expand_dims(input_sp_en, axis=0)

    # Convert Spectral Envelope
    output_sp_en, ppgs = convert_spectral_envelope(predictor, mfcc, input_sp_en)
    output_sp_en = np.squeeze(output_sp_en.astype(np.float64), axis=0)

    preproc_s = datetime.datetime.now()
    # Denormalization
    output_sp_en = denormalize_db(output_sp_en, hp.default.max_db, hp.default.min_db)

    # Db to amp
    output_sp_en = librosa.db_to_amplitude(output_sp_en)

    # Emphasize the magnitude
    output_sp_en = np.power(output_sp_en, hp.convert.emphasis_magnitude)

    preproc_e = datetime.datetime.now()
    preproc_t = preproc_e - preproc_s
    print("Pre-Processing time:{}s".format(preproc_t.seconds))

    # F0 transformation with WORLD Vocoder
    output_f0 = f0_adapt(input_f0, logdir2)

    # Synthesize audio and de-emphasize
    output_audio = pw.synthesize(output_f0, output_sp_en, input_ap, hp.default.sr)
    output_audio = inv_preemphasis(output_audio, coeff=hp.default.preemphasis)

    # Saving output_audio to 32-bit Float wav file
    output_audio = output_audio.astype(np.float32)
    librosa.output.write_wav(path="./converted/"+input_name,y=output_audio,sr=hp.default.sr)

    # Saving PPGS data to Grayscale Image and raw binary file
    ppgs = np.squeeze(ppgs, axis=0)
    plt.imsave('./converted/debug/'+input_name+'.png', ppgs, cmap='binary')
    np.save('./converted/debug/'+input_name+'.npy', ppgs)

    convert_e = datetime.datetime.now()
    convert_time = convert_e - convert_s
    print("Total Converting Time:{}s".format(convert_time.seconds))
コード例 #23
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ファイル: util_audio.py プロジェクト: RobertKajnak/AMT-SAGA
 def mag(self):
     if self._mag is None:
         if self._D is not None and self._ph is not None:
             if self._ref_mag is None:
                 self._ref_mag = 1.0
             self._mag = librosa.db_to_amplitude(self._D, ref=self._ref_mag)
         else:
             self._mag, self._ph = librosa.core.magphase(self.F)
     return self._mag
def SDR(original,predicted):
    """
    This function calculated the source-to-distortion ratio on a batchwise stft basis.
    Herefore SDR on all tf-units is found and the mean is put out.
    """
    original = librosa.db_to_amplitude(original)
    predicted = librosa.db_to_amplitude(predicted)

    distortion = predicted-original

    # power spectrum:
    original = original**2
    distortion = distortion**2

    sdr=np.divide(original,distortion)
    # Fixing NAN Values:
    sdr= np.nan_to_num(sdr,nan=60.0,posinf=60.0,neginf=60.0)
    sdr = 10*np.log10(sdr)
    return np.mean(sdr)
コード例 #25
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 def convert_specs_to_audio(self, spectrograms, min_max_values):
     signals = []
     for spec, min_max_value in zip(spectrograms, min_max_values):
         log_spec = spec[:, :, 0]
         denorm_log_spec = self.min_max_normalizer.denormalise(
             log_spec, min_max_value["min"], min_max_value["max"])
         spectr = librosa.db_to_amplitude(denorm_log_spec)
         signal = librosa.istft(spectr, hop_length=self.hop_length)
         signals.append(signal)
     return signals
コード例 #26
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    def img2audio(self, mode):  # DECODER
        magma_dif_list = self.readMagmaDiff()
        mtx_rgb_sum = self.readImg(self.trans_img_path)
        mtx_value = self.curver(mtx_rgb_sum, magma_dif_list)

        mtx_unit = np.ones(mtx_value.shape)
        mtx_db = (mtx_value / np.max(mtx_value) - mtx_unit) * float(
            abs(self.limit))  # get "normalized" db matrix
        mtx_amp = librosa.db_to_amplitude(mtx_db, ref=1.0)

        return self.reconstructer(mtx_amp, mode)
コード例 #27
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def generateWavFromCQT(amp, phase, sr, hl):
    
    Mdb_inormed = np.interp(amp, (amp.min(), amp.max()), (-15, 65))   
    iM = librosa.db_to_amplitude(Mdb_inormed)

    bins_per_octave = 12 * 12
    D = iM * np.exp(1j*phase)
    iy = librosa.icqt(C=D, sr=sr, hop_length=hl, bins_per_octave=bins_per_octave)
    iy = librosa.util.normalize(iy)

    return iy
コード例 #28
0
def pesq_on_batch(y_denoised,ytest,test_phase,sr=16000):

    pesqvalue=1
    try:


        y_denoised = np.squeeze(y_denoised,axis=3)
        y_denoised = np.squeeze(y_denoised,axis=0)



        y_denoised = librosa.db_to_amplitude(y_denoised)
        ytest = librosa.db_to_amplitude(ytest)

        denoised = y_denoised*test_phase
        original = ytest*test_phase

        denoised = librosa.istft(denoised)
        original = librosa.istft(original)

        #print(denoised)
        #print(original)

        denoised = librosa.util.normalize(denoised)
        original = librosa.util.normalize(original)

        #pesqvalue=pesq(sr, original, denoised, 'wb')


        pmsqe.init_constants(Fs=sr, Pow_factor=pmsqe.perceptual_constants.Pow_correc_factor_Hann, apply_SLL_equalization=True,
                           apply_bark_equalization=True, apply_on_degraded=True, apply_degraded_gain_correction=True)
        
        
        #pesqvalue=pesq(sr, original, denoised, 'wb')
        pesqvalue=per_frame_PMSQE(original,denoised)
        #print(pesqvalue)
    except:
        print("pesq didnt work")
        presqvalue=1

    return pesqvalue
コード例 #29
0
def magnitude_db_and_phase_to_audio(frame_length, hop_length_fft,
                                    stftaudio_magnitude_db, stftaudio_phase):

    stftaudio_magnitude_rev = librosa.db_to_amplitude(stftaudio_magnitude_db,
                                                      ref=1.0)

    audio_reverse_stft = stftaudio_magnitude_rev * stftaudio_phase
    audio_reconstruct = librosa.core.istft(audio_reverse_stft,
                                           hop_length=hop_length_fft,
                                           length=frame_length)

    return audio_reconstruct
コード例 #30
0
def rebuild_audio_from_spectro_clips(spectrogram_clips, is_dB_format=False):
    """rebuild waveform solely from magnitude spectrogram"""
    # audio spectrogram format:
    # 1. normal stft spectromgram
    # 2. dB-scaled spectrogram log(epilon + S*2)
    spectrogram = np.concatenate(spectrogram_clips, axis=1)
    if is_dB_format:
        spectrogram = librosa.db_to_amplitude(spectrogram)
    waveform = librosa.istft(spectrogram,
                             hop_length=HOP_LEN,
                             win_length=WIN_LEN)
    return waveform
コード例 #31
0
ファイル: test_core.py プロジェクト: dpwe/librosa
def test_db_to_amplitude():

    srand()

    NOISE_FLOOR = 1e-6

    # Make some noise
    x = np.abs(np.random.randn(1000)) + NOISE_FLOOR

    db = librosa.amplitude_to_db(x, top_db=None)
    x2 = librosa.db_to_amplitude(db)

    assert np.allclose(x, x2)
コード例 #32
0
ファイル: utils.py プロジェクト: WeCognize/deepvoice3
def spectrogram2wav(mag):
    '''# Generate wave file from spectrogram'''
    # transpose
    mag = mag.T

    # de-noramlize
    mag = (np.clip(mag, 0, 1) * hp.max_db) - hp.max_db + hp.ref_db

    # to amplitude
    mag = librosa.db_to_amplitude(mag)
    # print(np.max(mag), np.min(mag), mag.shape)
    # (1025, 812, 16)

    # wav reconstruction
    wav = griffin_lim(mag)

    # de-preemphasis
    wav = signal.lfilter([1], [1, -hp.preemphasis], wav)

    # trim
    wav, _ = librosa.effects.trim(wav)

    return wav
コード例 #33
0
ファイル: test_core.py プロジェクト: dpwe/librosa
    def __test(ref):

        db = librosa.amplitude_to_db(xp, ref=ref, top_db=None)
        xp2 = librosa.db_to_amplitude(db, ref=ref)

        assert np.allclose(xp, xp2)