The audioop module contains some useful operations on sound fragments. It operates on sound fragments consisting of signed integer samples 8, 16 or 32 bits wide, stored in Python strings. All scalar items are integers, unless specified otherwise.
This module provides support for a-LAW, u-LAW and Intel/DVI ADPCM encodings.
A few of the more complicated operations only take 16-bit samples, otherwise the sample size (in bytes) is always a parameter of the operation.
The module defines the following variables and functions:
Return a factor F such that rms(add(fragment, mul(reference, -F))) is minimal, i.e., return the factor with which you should multiply reference to make it match as well as possible to fragment. The fragments should both contain 2-byte samples.
The time taken by this routine is proportional to len(fragment).
Search fragment for a slice of length length samples (not bytes!) with maximum energy, i.e., return i for which rms(fragment[i*2:(i+length)*2]) is maximal. The fragments should both contain 2-byte samples.
The routine takes time proportional to len(fragment).
Convert samples to 4 bit Intel/DVI ADPCM encoding. ADPCM coding is an adaptive coding scheme, whereby each 4 bit number is the difference between one sample and the next, divided by a (varying) step. The Intel/DVI ADPCM algorithm has been selected for use by the IMA, so it may well become a standard.
state is a tuple containing the state of the coder. The coder returns a tuple (adpcmfrag, newstate), and the newstate should be passed to the next call of lin2adpcm(). In the initial call, None can be passed as the state. adpcmfrag is the ADPCM coded fragment packed 2 4-bit values per byte.
Convert samples between 1-, 2- and 4-byte formats.
In some audio formats, such as .WAV files, 16 and 32 bit samples are signed, but 8 bit samples are unsigned. So when converting to 8 bit wide samples for these formats, you need to also add 128 to the result:
new_frames = audioop.lin2lin(frames, old_width, 1) new_frames = audioop.bias(new_frames, 1, 128)
The same, in reverse, has to be applied when converting from 8 to 16 or 32 bit width samples.
Convert the frame rate of the input fragment.
state is a tuple containing the state of the converter. The converter returns a tuple (newfragment, newstate), and newstate should be passed to the next call of ratecv(). The initial call should pass None as the state.
The weightA and weightB arguments are parameters for a simple digital filter and default to 1 and 0 respectively.
Return the root-mean-square of the fragment, i.e. sqrt(sum(S_i^2)/n).
This is a measure of the power in an audio signal.
Note that operations such as mul() or max() make no distinction between mono and stereo fragments, i.e. all samples are treated equal. If this is a problem the stereo fragment should be split into two mono fragments first and recombined later. Here is an example of how to do that:
def mul_stereo(sample, width, lfactor, rfactor): lsample = audioop.tomono(sample, width, 1, 0) rsample = audioop.tomono(sample, width, 0, 1) lsample = audioop.mul(sample, width, lfactor) rsample = audioop.mul(sample, width, rfactor) lsample = audioop.tostereo(lsample, width, 1, 0) rsample = audioop.tostereo(rsample, width, 0, 1) return audioop.add(lsample, rsample, width)
If you use the ADPCM coder to build network packets and you want your protocol to be stateless (i.e. to be able to tolerate packet loss) you should not only transmit the data but also the state. Note that you should send the initial state (the one you passed to lin2adpcm()) along to the decoder, not the final state (as returned by the coder). If you want to use struct.struct() to store the state in binary you can code the first element (the predicted value) in 16 bits and the second (the delta index) in 8.
The ADPCM coders have never been tried against other ADPCM coders, only against themselves. It could well be that I misinterpreted the standards in which case they will not be interoperable with the respective standards.
The find*() routines might look a bit funny at first sight. They are primarily meant to do echo cancellation. A reasonably fast way to do this is to pick the most energetic piece of the output sample, locate that in the input sample and subtract the whole output sample from the input sample:
def echocancel(outputdata, inputdata): pos = audioop.findmax(outputdata, 800) # one tenth second out_test = outputdata[pos*2:] in_test = inputdata[pos*2:] ipos, factor = audioop.findfit(in_test, out_test) # Optional (for better cancellation): # factor = audioop.findfactor(in_test[ipos*2:ipos*2+len(out_test)], # out_test) prefill = '\0'*(pos+ipos)*2 postfill = '\0'*(len(inputdata)-len(prefill)-len(outputdata)) outputdata = prefill + audioop.mul(outputdata,2,-factor) + postfill return audioop.add(inputdata, outputdata, 2)