[SciPy-User] parsing a wave file

David Cournapeau cournape@gmail....
Fri Feb 24 09:15:59 CST 2012


On Thu, Feb 23, 2012 at 9:08 AM, Friedrich Romstedt
<friedrichromstedt@gmail.com> wrote:
> Am 23.02.2012 um 12:04 schrieb David Hutto <smokefloat@gmail.com>:
>> ####################
>>            f = Sndfile(r'c:\Users\david\test01.wav', 'r')
>
> Sorry for the probably dumb question, but where does "Sndfile" originate from?  A quick search in the Python 2.6.5 docs and in the online scipy docs yields nothing.

Most likely coming from the scikits.audiolab.

>>        fs = f.samplerate
>>        nc = f.channels
>>        enc = f.encoding
>>        data = f.read_frames(1000)
>>        frame_amount = 1000
>>        data_float = f.read_frames(frame_amount, dtype=np.float32)
>>        for i in range(0,frame_amount,1):
>>            print data_float[i]
>>    ##############
>> returns data_float[i] in the form:
>> [-1,0.990988]
>> [.08545,-0.009988]
>
> It looks to me as if the "Sndfile" instance already decomposes into channels.  In the wave file, the channels are intermingled, one frame per channel for each time instance (I'm not fully sure if "frame" denotes the full chunk for one time instance or only one datum for one channel, part of the chunk).

A frame contains one time point, and each time point contains up to M
values, where M is the number of channels.

>
>> Any suggestions as to what I'm parsing in the wrong way, or better
>> solutions than the above?
>
> I could imagine that the Sndfile class you're using takes the sampling width (i.e., the number of bytes per sample) from the dtype you're handing over, interpreting the originally possible integer-valued frames as float32's. This could also alter the alignment, so that full garbage would result, possibly explaining the apparently highly variable output (as far as I can judge from the two chunks you provided).  But notice, this is only guessing, speculation so to speak.

That's not how it works. The dtype in read_frames only affect the
dtype of the output, but will work independently of the type used in
the wavfile. It goes through libsndfile to read the wav file, and
libsndfile is known to be extremely reliable (used in many profesional
audio softwares, open source and proprietary).

>
> I've made positive experience with using the wave module from the standard library. I wrote a module (part of a larger sound analysis suite) to read arbitrary wave files as long as they are integer-valued. The module reads the file into a numpy array with sufficient speed (a few seconds for a 3' piece), and separates the channels. I guess it could be easily adapted to your use case (currently the channels are averaged in the end; you would just have to leave that step out). I would give it to you as a public domain module.

While scipy.io/wave modules are fine, using scikits.audiolab will give
access to many different formats, even broken ones generated by some
softwares. I would say the only siginficant issue with
scikits.audiolab is the dependency on libsndfile, and some people may
not like the LGPL licensing for both libsndfile and scikits.audiolab.

David


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