如何使用Python将scipy.signal.spectrogram转换为音频？

Question

我有：

import librosa
from scipy import signal 
import scipy.io.wavfile as sf    

samples, sample_rate = sf.read(args.file)
nperseg = int(sample_rate * 0.001 * 20)
frequencies, times, spectrogram = signal.spectrogram(samples, 
                                                     sample_rate, 
                                                     nperseg=nperseg, 
                                                     window=signal.hann(nperseg))

audio_signal = librosa.griffinlim(spectrogram)
print(audio_signal, audio_signal.shape)

sf.write('test.wav', audio_signal, sample_rate)

但是，这会产生(接近)空声音文件。

Answer 1

正如@DrSpill所提到的，scipy.io.wav.read和scipy.io.wav.write订单是错误的，来自利布罗萨的导入也是不正确的。这应该可以做到：

import librosa
import numpy as np
import scipy.signal
import scipy.io.wavfile

# read file
file    = "temp/processed_file.wav"
fs, sig = scipy.io.wavfile.read(file)
nperseg = int(fs * 0.001 * 20)

# process
frequencies, times, spectrogram = scipy.signal.spectrogram(sig, 
                                                           fs, 
                                                           nperseg=nperseg, 
                                                           window=scipy.signal.hann(nperseg))
audio_signal = librosa.core.spectrum.griffinlim(spectrogram)
print(audio_signal, audio_signal.shape)

# write output
scipy.io.wavfile.write('test.wav', fs, np.array(audio_signal, dtype=np.int16))

备注:当我听到结果文件时，的速度加快了，我认为这是由于您的处理，但是经过一些调整，它应该能工作。

一个很好的选择是只使用利布罗萨，如下所示：

import librosa
import numpy as np

# read file
file    = "temp/processed_file.wav"
sig, fs = librosa.core.load(file, sr=8000)

# process
abs_spectrogram = np.abs(librosa.core.spectrum.stft(sig))
audio_signal = librosa.core.spectrum.griffinlim(abs_spectrogram)

print(audio_signal, audio_signal.shape)

# write output
librosa.output.write_wav('test2.wav', audio_signal, fs)

Answer 2

librosa.output为删除。它不再提供不推荐的输出模块。相反，尝试soundfile.write

import numpy as np
import soundfile as sf
sf.write('stereo_file.wav', np.random.randn(10, 2), 44100, 'PCM_24')

#Per your code you could try:
sf.write('test.wav', audio_signal, sample_rate, 'PCM_24')