How to use the librosa.load function in librosa

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github pkmital / time-domain-neural-audio-style-transfer / models / fourier.py View on Github external
output_fname,
        n_fft=4096,
        n_layers=1,
        n_filters=4096,
        hop_length=256,
        alpha=0.05,
        k_w=15,
        k_h=3,
        optimizer='bfgs',
        stride=1,
        iterations=300,
        sr=22050):

    frame_size = n_fft // 2

    audio, fs = librosa.load(content_fname, sr=sr)
    content = chop(audio, hop_size=hop_length, frame_size=frame_size)
    audio, fs = librosa.load(style_fname, sr=sr)
    style = chop(audio, hop_size=hop_length, frame_size=frame_size)

    n_frames = min(content.shape[0], style.shape[0])
    n_samples = min(content.shape[1], style.shape[1])
    content = content[:n_frames, :n_samples]
    style = style[:n_frames, :n_samples]

    content_features, style_gram, kernels, freqs = compute_features(
        content=content,
        style=style,
        stride=stride,
        n_fft=n_fft,
        n_layers=n_layers,
        n_filters=n_filters,
github interactiveaudiolab / nussl / tests / tests_core / test_audio_signal.py View on Github external
def test_resample_vs_librosa_load(self):
        # Check against librosa load function
        a = nussl.AudioSignal(self.audio_input1)
        a.resample(48000)
        b_audio_data, b_sample_rate = librosa.load(self.audio_input1, sr=48000)
        assert (a.sample_rate == b_sample_rate)
        assert (np.allclose(a.audio_data, b_audio_data))
github algorithmic-music-exploration / amen / tests / test_audio.py View on Github external
def test_sample_data():
    y, sr = librosa.load(EXAMPLE_FILE)
    assert audio.analysis_samples.all() == y.all()
github LemonATsu / pop-to-8bit / py8bits / rpca_mask.py View on Github external
if mask_type == 1:
        mask = np.greater(np.abs(S), (gain * np.abs(L)))
        S = S * mask
        L = S_mix - S
    
    voice = librosa.core.istft(S.T, hop_length=hop_size, win_length=win_size)
    accom = librosa.core.istft(L.T, hop_length=hop_size, win_length=win_size)
    voice = voice / np.max(np.abs(voice))
    accom = accom / np.max(np.abs(accom))

    return voice, accom


if __name__ == '__main__':
    clip, fs = librosa.load('examples/c1.wav', mono=False,sr=44100)
    clip = clip.T
    clip = clip[:,1] + clip[:,0]
    voice, accom = svs_RPCA(clip)
    
    librosa.output.write_wav('voice.wav', voice, sr=44100)
    librosa.output.write_wav('accom.wav', accom, sr=44100)
github librosa / librosa / examples / analyzer.py View on Github external
Arguments
    ---------
    infile  -- (str) path to input file


    Returns
    -------
    analysis -- (dict) of various useful things
    '''

    A = {}
    
    A['filename'] = infile

    y, sr = librosa.load(infile, sr=SR)
    
    # First, get the track duration
    A['duration'] = float(len(y)) / sr

    # Then, get the beats
    tempo, beats = librosa.beat.beat_track(y, sr, hop_length=HOP)

    # Push the last frame as a phantom beat
    A['tempo'] = tempo
    A['beats'] = librosa.frames_to_time(beats, sr, hop_length=HOP).tolist()

    
    S = librosa.feature.melspectrogram(y, sr,   n_fft=2048, 
                                                hop_length=HOP, 
                                                n_mels=80, 
                                                fmax=8000)
github haoxiangsnr / Build-SE-Dataset / time_domain.py View on Github external
offset=clean_cfg["offset"]
        )
        random.shuffle(clean_speech_paths)
        clean_ys = load_wavs(
            file_paths=clean_speech_paths,
            sr=clean_cfg["sampling_rate"],
            min_sampling=clean_cfg["min_sampling"],
        )
        print("Loaded clean speeches.")

        # 加载噪声信号,存至 dict 中
        noise_cfg = dataset_cfg["noise"]
        noise_database_dir = Path(noise_cfg["database"])
        noise_ys = {}
        for noise_type in tqdm(noise_cfg["types"], desc="Loading noise files"):
            mixture, _ = librosa.load(
                (noise_database_dir / (noise_type + ".wav")).as_posix(),
                sr=noise_cfg["sampling_rate"])
            noise_ys[noise_type] = mixture
        print("Loaded noise.")

        # 合成带噪语音
        mixture_store = {}
        clean_store = {}
        for i, clean in tqdm(enumerate(clean_ys, start=1), desc="合成带噪语音"):
            num = str(i).zfill(4)
            for snr in dataset_cfg["snr"]:
                for noise_type in noise_ys.keys():
                    basename_text = f"{num}_{noise_type}_{snr}"

                    clean, noise = corrected_the_length_of_noise_and_clean_speech(
                        clean_y=clean,
github santi-pdp / segan_pytorch / segan / transforms.py View on Github external
def __init__(self, noises_dir, snr_levels=[0, 5, 10], do_IRS=False,
                 prob=1):
        self.prob = prob
        self.noises_dir = noises_dir
        self.snr_levels = snr_levels
        self.do_IRS = do_IRS
        # read noises in dir
        noises = glob.glob(os.path.join(noises_dir, '*.wav'))
        if len(noises) == 0:
            raise ValueError('[!] No noises found in {}'.format(noises_dir))
        else:
            print('[*] Found {} noise files'.format(len(noises)))
            self.noises = []
            for n_i, npath in enumerate(noises, start=1):
                #nwav = wavfile.read(npath)[1]
                nwav = librosa.load(npath, sr=None)[0]
                self.noises.append({'file':npath, 
                                    'data':nwav.astype(np.float32)})
                log_noise_load = 'Loaded noise {:3d}/{:3d}: ' \
                                 '{}'.format(n_i, len(noises),
                                             npath)
                print(log_noise_load)
        self.eps = 1e-22
github daylen / audio-style-classifier / data_manager.py View on Github external
def thread_main(self, sess):
		stop = False
		while not stop:
			for fname in self.fnames:
				if self.coord.should_stop():
					stop = True
					break
				try:
					audio = librosa.load(fname, sr=self.sample_rate)[0]
				except:
					print 'could not read file', fname, 'skipping...'
					continue
				rand_start_idx = np.random.randint(0, len(audio) - self.sample_rate * self.seconds_of_audio)
				audio = audio[rand_start_idx:rand_start_idx + self.sample_rate * self.seconds_of_audio]
				label = self.data_dict[fname]
				sess.run(self.enqueue_op, feed_dict={self.x: audio, self.y: label})
github colinsongf / keyword_spotting / process_wav.py View on Github external
def process_stft(f):

    y, sr = librosa.load(f, sr=config.samplerate)
    if config.pre_emphasis:
        y = pre_emphasis(y)
    linearspec = np.transpose(np.abs(
        librosa.core.stft(y, config.fft_size,
                          config.hop_size)))

    return linearspec, y