[英]How to convert a 24-bit wav file to 16 or 32 bit files in python3
[英]How to convert a .wav file to a spectrogram in python3
我正在嘗試從 python3 中的 .wav 文件創建頻譜圖。
我希望最終保存的圖像看起來類似於此圖像:
我嘗試了以下方法:
此堆棧溢出帖子: 波形文件的頻譜圖
這篇文章有點奏效。 運行后,我得到了
但是,此圖不包含我需要的顏色。 我需要一個有顏色的光譜圖。 我嘗試修改此代碼以嘗試添加顏色,但是在為此花費了大量時間和精力之后,我無法弄清楚!
然后我嘗試了這個教程。
當我嘗試運行它時,此代碼崩潰(在第 17 行),並出現錯誤 TypeError: 'numpy.float64' object cannot be interpret as an integer。
第 17 行:
samples = np.append(np.zeros(np.floor(frameSize/2.0)), sig)
我試圖通過鑄造來修復它
samples = int(np.append(np.zeros(np.floor(frameSize/2.0)), sig))
我也試過
samples = np.append(np.zeros(int(np.floor(frameSize/2.0)), sig))
然而,這些最終都沒有奏效。
我真的很想知道如何將我的 .wav 文件轉換為帶有顏色的頻譜圖,以便我可以分析它們! 任何幫助,將不勝感激!!!!!
如果您希望我提供有關我的 Python 版本、我嘗試過的內容或我想要實現的內容的更多信息,請告訴我。
使用scipy.signal.spectrogram
。
import matplotlib.pyplot as plt
from scipy import signal
from scipy.io import wavfile
sample_rate, samples = wavfile.read('path-to-mono-audio-file.wav')
frequencies, times, spectrogram = signal.spectrogram(samples, sample_rate)
plt.pcolormesh(times, frequencies, spectrogram)
plt.imshow(spectrogram)
plt.ylabel('Frequency [Hz]')
plt.xlabel('Time [sec]')
plt.show()
在嘗試執行此操作之前,請確保您的 wav 文件是單聲道(單聲道)而不是立體聲(雙聲道)。 我強烈建議閱讀https://docs.scipy.org/doc/scipy-0.19.0/reference/generated/scipy.signal.spectrogram.html 上的 scipy 文檔。
plt.pcolormesh
plt.imshow
所指出的,將plt.pcolormesh
放在plt.pcolormesh
之前似乎可以解決一些問題,如果發生解包錯誤,請按照下面@cgnorthcutt 的步驟進行操作。
我已經修復了您在http://www.frank-zalkow.de/en/code-snippets/create-audio-spectrograms-with-python.html 中遇到的錯誤
這種實現更好,因為您可以更改binsize
(例如binsize=2**8
)
import numpy as np
from matplotlib import pyplot as plt
import scipy.io.wavfile as wav
from numpy.lib import stride_tricks
""" short time fourier transform of audio signal """
def stft(sig, frameSize, overlapFac=0.5, window=np.hanning):
win = window(frameSize)
hopSize = int(frameSize - np.floor(overlapFac * frameSize))
# zeros at beginning (thus center of 1st window should be for sample nr. 0)
samples = np.append(np.zeros(int(np.floor(frameSize/2.0))), sig)
# cols for windowing
cols = np.ceil( (len(samples) - frameSize) / float(hopSize)) + 1
# zeros at end (thus samples can be fully covered by frames)
samples = np.append(samples, np.zeros(frameSize))
frames = stride_tricks.as_strided(samples, shape=(int(cols), frameSize), strides=(samples.strides[0]*hopSize, samples.strides[0])).copy()
frames *= win
return np.fft.rfft(frames)
""" scale frequency axis logarithmically """
def logscale_spec(spec, sr=44100, factor=20.):
timebins, freqbins = np.shape(spec)
scale = np.linspace(0, 1, freqbins) ** factor
scale *= (freqbins-1)/max(scale)
scale = np.unique(np.round(scale))
# create spectrogram with new freq bins
newspec = np.complex128(np.zeros([timebins, len(scale)]))
for i in range(0, len(scale)):
if i == len(scale)-1:
newspec[:,i] = np.sum(spec[:,int(scale[i]):], axis=1)
else:
newspec[:,i] = np.sum(spec[:,int(scale[i]):int(scale[i+1])], axis=1)
# list center freq of bins
allfreqs = np.abs(np.fft.fftfreq(freqbins*2, 1./sr)[:freqbins+1])
freqs = []
for i in range(0, len(scale)):
if i == len(scale)-1:
freqs += [np.mean(allfreqs[int(scale[i]):])]
else:
freqs += [np.mean(allfreqs[int(scale[i]):int(scale[i+1])])]
return newspec, freqs
""" plot spectrogram"""
def plotstft(audiopath, binsize=2**10, plotpath=None, colormap="jet"):
samplerate, samples = wav.read(audiopath)
s = stft(samples, binsize)
sshow, freq = logscale_spec(s, factor=1.0, sr=samplerate)
ims = 20.*np.log10(np.abs(sshow)/10e-6) # amplitude to decibel
timebins, freqbins = np.shape(ims)
print("timebins: ", timebins)
print("freqbins: ", freqbins)
plt.figure(figsize=(15, 7.5))
plt.imshow(np.transpose(ims), origin="lower", aspect="auto", cmap=colormap, interpolation="none")
plt.colorbar()
plt.xlabel("time (s)")
plt.ylabel("frequency (hz)")
plt.xlim([0, timebins-1])
plt.ylim([0, freqbins])
xlocs = np.float32(np.linspace(0, timebins-1, 5))
plt.xticks(xlocs, ["%.02f" % l for l in ((xlocs*len(samples)/timebins)+(0.5*binsize))/samplerate])
ylocs = np.int16(np.round(np.linspace(0, freqbins-1, 10)))
plt.yticks(ylocs, ["%.02f" % freq[i] for i in ylocs])
if plotpath:
plt.savefig(plotpath, bbox_inches="tight")
else:
plt.show()
plt.clf()
return ims
ims = plotstft(filepath)
import os
import wave
import pylab
def graph_spectrogram(wav_file):
sound_info, frame_rate = get_wav_info(wav_file)
pylab.figure(num=None, figsize=(19, 12))
pylab.subplot(111)
pylab.title('spectrogram of %r' % wav_file)
pylab.specgram(sound_info, Fs=frame_rate)
pylab.savefig('spectrogram.png')
def get_wav_info(wav_file):
wav = wave.open(wav_file, 'r')
frames = wav.readframes(-1)
sound_info = pylab.fromstring(frames, 'int16')
frame_rate = wav.getframerate()
wav.close()
return sound_info, frame_rate
對於卡佩拉科學 - 波西米亞重力! 這給出:
使用graph_spectrogram(path_to_your_wav_file)
。 我不記得我從哪里獲取這個片段的博客。 每當我再次看到它時,我都會添加鏈接。
您可以使用librosa
來滿足您的 mp3 librosa
需求。 這是我找到的一些代碼,感謝來自 medium 的 Parul Pandey 。 我使用的代碼是這樣的
# Method described here https://stackoverflow.com/questions/15311853/plot-spectogram-from-mp3
import librosa
import librosa.display
from pydub import AudioSegment
import matplotlib.pyplot as plt
from scipy.io import wavfile
from tempfile import mktemp
def plot_mp3_matplot(filename):
"""
plot_mp3_matplot -- using matplotlib to simply plot time vs amplitude waveplot
Arguments:
filename -- filepath to the file that you want to see the waveplot for
Returns -- None
"""
# sr is for 'sampling rate'
# Feel free to adjust it
x, sr = librosa.load(filename, sr=44100)
plt.figure(figsize=(14, 5))
librosa.display.waveplot(x, sr=sr)
def convert_audio_to_spectogram(filename):
"""
convert_audio_to_spectogram -- using librosa to simply plot a spectogram
Arguments:
filename -- filepath to the file that you want to see the waveplot for
Returns -- None
"""
# sr == sampling rate
x, sr = librosa.load(filename, sr=44100)
# stft is short time fourier transform
X = librosa.stft(x)
# convert the slices to amplitude
Xdb = librosa.amplitude_to_db(abs(X))
# ... and plot, magic!
plt.figure(figsize=(14, 5))
librosa.display.specshow(Xdb, sr = sr, x_axis = 'time', y_axis = 'hz')
plt.colorbar()
# same as above, just changed the y_axis from hz to log in the display func
def convert_audio_to_spectogram_log(filename):
x, sr = librosa.load(filename, sr=44100)
X = librosa.stft(x)
Xdb = librosa.amplitude_to_db(abs(X))
plt.figure(figsize=(14, 5))
librosa.display.specshow(Xdb, sr = sr, x_axis = 'time', y_axis = 'log')
plt.colorbar()
干杯!
上面初學者的回答非常好。 我沒有 50 rep,所以我不能評論它,但如果你想要頻域中的正確幅度,stft 函數應該是這樣的:
import numpy as np
from matplotlib import pyplot as plt
import scipy.io.wavfile as wav
from numpy.lib import stride_tricks
""" short time fourier transform of audio signal """
def stft(sig, frameSize, overlapFac=0, window=np.hanning):
win = window(frameSize)
hopSize = int(frameSize - np.floor(overlapFac * frameSize))
# zeros at beginning (thus center of 1st window should be for sample nr. 0)
samples = np.append(np.zeros(int(np.floor(frameSize/2.0))), sig)
# cols for windowing
cols = np.ceil( (len(samples) - frameSize) / float(hopSize)) + 1
# zeros at end (thus samples can be fully covered by frames)
samples = np.append(samples, np.zeros(frameSize))
frames = stride_tricks.as_strided(samples, shape=(int(cols), frameSize), strides=(samples.strides[0]*hopSize, samples.strides[0])).copy()
frames *= win
fftResults = np.fft.rfft(frames)
windowCorrection = 1/(np.sum(np.hanning(frameSize))/frameSize) #This is amplitude correct (1/mean(window)). Energy correction is 1/rms(window)
FFTcorrection = 2/frameSize
scaledFftResults = fftResults*windowCorrection*FFTcorrection
return scaledFftResults
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