audioop— Manipulate raw audio data
audioop module contains some useful operations on sound fragments.
It operates on sound fragments consisting of signed integer samples 8, 16, 24
or 32 bits wide, stored in bytes-like objects. All scalar items are
integers, unless specified otherwise.
Changed in version 3.4: Support for 24-bit samples was added. All functions now accept any bytes-like object. String input now results in an immediate error.
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:
add(fragment1, fragment2, width)
4. Both fragments should have the same length. Samples are truncated in case of overflow.
adpcm2lin(adpcmfragment, width, state)
lin2adpcm()for details on ADPCM coding. Return a tuple
(sample, newstate)where the sample has the width specified in width.
bias(fragment, width, bias)
“Byteswap” all samples in a fragment and returns the modified fragment. Converts big-endian samples to little-endian and vice versa.
New in version 3.4.
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
findfactor()to compute the best match, and minimizing the result. The fragments should both contain 2-byte samples. Return a tuple
(offset, factor)where offset is the (integer) offset into fragment where the optimal match started and factor is the (floating-point) factor as per
Search fragment for a slice of length length samples (not bytes!) with
maximum energy, i.e., return i for which
is maximal. The fragments should both contain 2-byte samples.
The routine takes time proportional to
getsample(fragment, width, index)
lin2adpcm(fragment, width, state)
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
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.
lin2lin(fragment, width, newwidth)
Convert samples between 1-, 2-, 3- and 4-byte formats.
In some audio formats, such as .WAV files, 16, 24 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, 24 or 32 bit width samples.
mul(fragment, width, factor)
ratecv(fragment, width, nchannels, inrate, outrate, state[, weightA[, weightB]])
Convert the frame rate of the input fragment.
state is a tuple containing the state of the converter. The converter returns
(newfragment, newstate), and newstate should be passed to the next
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
Return the root-mean-square of the fragment, i.e.
This is a measure of the power in an audio signal.
tomono(fragment, width, lfactor, rfactor)
tostereo(fragment, width, lfactor, rfactor)
Note that operations such as
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(lsample, width, lfactor) rsample = audioop.mul(rsample, 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.
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)