关于梅尔频谱你想知道的都在这里
2023-05-03 08:07:33

在语音分析,合成,转换中,第一步往往是提取语音特征参数。

利用机器学习方法进行上述语音任务,常用到梅尔频谱。

本文介绍从音频文件提取梅尔频谱,和从梅尔频谱变成音频波形。

从音频波形提取Mel频谱:

对音频信号预加重、分帧和加窗

对每帧信号进行短时傅立叶变换STFT,得到短时幅度谱

短时幅度谱通过Mel滤波器组得到Mel频谱

从Mel频谱重建音频波形

Mel频谱转换成幅度谱

griffin_lim声码器算法重建波形

去加重

声码器有很多种,比如world,straight等,但是griffin_lim是特殊的,它不需要相位信息就可以重频谱重建波形,实际上它根据帧之间的关系估计相位信息。和成的音频质量也较高,代码也比较简单。

音频波形 到 mel-spectrogram

sr = 24000 # Sample rate.

n_fft = 2048 # fft points (samples)

frame_shift = 0.0125 # seconds

frame_length = 0.05 # seconds

hop_length = int(sr*frame_shift) # samples.

win_length = int(sr*frame_length) # samples.

n_mels = 512 # Number of Mel banks to generate

power = 1.2 # Exponent for amplifying the predicted magnitude

n_iter = 100 # Number of inversion iterations

preemphasis = .97 # or None

max_db = 100

ref_db = 20

top_db = 15 



def get_spectrograms(fpath):

    '''Returns normalized log(melspectrogram) and log(magnitude) from `sound_file`.

    Args:

      sound_file: A string. The full path of a sound file.

    Returns:

      mel: A 2d array of shape (T, n_mels) <- Transposed

      mag: A 2d array of shape (T, 1+n_fft/2) <- Transposed

 '''

    # Loading sound file

    y, sr = librosa.load(fpath, sr=sr)

    # Trimming

    y, _ = librosa.effects.trim(y, top_db=top_db)

    # Preemphasis

    y = np.append(y[0], y[1:] - preemphasis * y[:-1])

    

    # stft

    linear = librosa.stft(y=y,

                          n_fft=n_fft,

                          hop_length=hop_length,

                          win_length=win_length)

    # magnitude spectrogram

    mag = np.abs(linear)  # (1+n_fft//2, T)

    # mel spectrogram

    mel_basis = librosa.filters.mel(sr, n_fft, n_mels)  # (n_mels, 1+n_fft//2)

    mel = np.dot(mel_basis, mag)  # (n_mels, t)

    # to decibel

    mel = 20 * np.log10(np.maximum(1e-5, mel))

    mag = 20 * np.log10(np.maximum(1e-5, mag))

    # normalize

    mel = np.clip((mel - ref_db + max_db) / max_db, 1e-8, 1)

    mag = np.clip((mag - ref_db + max_db) / max_db, 1e-8, 1)

    # Transpose

    mel = mel.T.astype(np.float32)  # (T, n_mels)

    mag = mag.T.astype(np.float32)  # (T, 1+n_fft//2)

    return mel, mag 



mel-spectrogram 到 音频波形

def melspectrogram2wav(mel):

    '''# Generate wave file from spectrogram'''

    # transpose

    mel = mel.T

    # de-noramlize

    mel = (np.clip(mel, 0, 1) * max_db) - max_db + ref_db

    # to amplitude

    mel = np.power(10.0, mel * 0.05)

    m = _mel_to_linear_matrix(sr, n_fft, n_mels)

    mag = np.dot(m, mel)

    # wav reconstruction

    wav = griffin_lim(mag)

    # de-preemphasis

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

    # trim

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

    return wav.astype(np.float32)

def spectrogram2wav(mag):

    '''# Generate wave file from spectrogram'''

    # transpose

    mag = mag.T

    # de-noramlize

    mag = (np.clip(mag, 0, 1) * max_db) - max_db + ref_db

    # to amplitude

    mag = np.power(10.0, mag * 0.05)

    # wav reconstruction

    wav = griffin_lim(mag)

    # de-preemphasis

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

    # trim

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

    return wav.astype(np.float32)

 




几个辅助函数:

def _mel_to_linear_matrix(sr, n_fft, n_mels):

    m = librosa.filters.mel(sr, n_fft, n_mels)

    m_t = np.transpose(m)

    p = np.matmul(m, m_t)

    d = [1.0 / x if np.abs(x) > 1.0e-8 else x for x in np.sum(p, axis=0)]

    return np.matmul(m_t, np.diag(d))

def griffin_lim(spectrogram):

    '''Applies Griffin-Lim's raw.

    '''

    X_best = copy.deepcopy(spectrogram)

    for i in range(n_iter):

        X_t = invert_spectrogram(X_best)

        est = librosa.stft(X_t, n_fft, hop_length, win_length=win_length)

        phase = est / np.maximum(1e-8, np.abs(est))

        X_best = spectrogram * phase

    X_t = invert_spectrogram(X_best)

    y = np.real(X_t)

    return y

def invert_spectrogram(spectrogram):

    '''

    spectrogram: [f, t]

    '''

    return librosa.istft(spectrogram, hop_length, win_length=win_length, window="hann")

 



预加重:

语音信号的平均功率谱受声门激励和口鼻辐射影响,高频端约在800HZ以上按6dB/倍频程衰落,预加重的目的是提升高频成分,使信号频谱平坦化,以便于频谱分析或声道参数分析.

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版权声明:本文为CSDN博主「明月几时有.」的原创文章,遵循CC 4.0 BY-SA版权协议,转载请附上原文出处链接及本声明。

原文链接:https://blog.csdn.net/weixin_35576881/article/details/90300799

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