audioop
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. This is the same
format as used by the al
and sunaudiodev
modules. All
scalar items are integers, unless specified otherwise.
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:
1
, 2
or 4
. Both fragments should have the same
length.
lin2adpcm
for details on ADPCM coding.
Return a tuple (sample, newstate)
where the sample
has the width specified in width.
lin2adpcm3
for
details.
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)
.
Try to match reference as well as possible to a portion of
fragment (which should be the longer fragment). This is
(conceptually) done by taking slices out of fragment, using
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
findfactor
.
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)
.
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.
, 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
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)