On the client side, I am sending a blob audio (wav) file. On the server side, I am trying to convert the blob file to an audio wav file. I did the following:
blob = request.FILES['file'] name = "TEST.wav" audio = wave.open(name, 'wb') audio.setnchannels(1) audio.writeframes(blob.read())
I thought that converting the blob would be similar to converting a blob image to a jpeg file, but was very incorrect in that assumption. That didn’t work; I get an error – “Error: sample width not specified.” I then used setsampwidth() and tossed in an arbitrary number between 1 and 4 (after looking at the wave.py source file…I don’t know why the bytes have to be between 1 and 4). After that another error is thrown – “Error: sampling rate not specified.” How do I specify the sampling rate?
What does the setnchannels(), setsampwidth() methods do? Is there an “easy” way I generate the wav file from the blob?
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Answer
Previously, I never do it before.. but, in my test this script below is worked well for me.. (But the audio output isn’t same like original file).
>>> nchannels = 2
>>> sampwidth = 2
>>> framerate = 8000
>>> nframes = 100
>>>
>>> import wave
>>>
>>> name = 'output.wav'
>>> audio = wave.open(name, 'wb')
>>> audio.setnchannels(nchannels)
>>> audio.setsampwidth(sampwidth)
>>> audio.setframerate(framerate)
>>> audio.setnframes(nframes)
>>>
>>> blob = open("original.wav").read() # such as `blob.read()`
>>> audio.writeframes(blob)
>>>
I found this method at https://stackoverflow.com/a/3637480/6396981
Finally, by changing the value of nchannels and sampwidth with 1. and I got an audio that same with original file.
nchannels = 1 sampwidth = 1 framerate = 8000 nframes = 1
Tested under Python2, and got an error
UnicodeDecodeError: 'utf-8' codec can't decode byte 0x95 in position 4: invalid start byteon Python3.