问题
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?
回答1:
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 byte
on Python3.
回答2:
I have encountered the same problem as well. My problem was having a low pitched output compared to the original. I manage to reverse engineer the original audio to get the nframes, samplerate, and sampwidth using getnframes(),getframerate(), and getsampwidth() respectively. At last, I managed to tweak the sample frequency/ frame rate to somehow bring the perfect tone.
The tweaking became perfect at a certain offset frequency than the original. Mine worked fine at an offset sum of the sixteenth of the original samplerate.
i.e.
OffsetFrequency = OriginalFrequency/16
Frequency = OriginalFrequency + OffsetFrequency
来源:https://stackoverflow.com/questions/42288607/create-a-wav-file-from-blob-audio-django