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(python2.1-lib.info)audioop
Manipulate raw audio data
=========================
Manipulate raw audio data.
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.
This module provides support for 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:
`error'
This exception is raised on all errors, such as unknown number of
bytes per sample, etc.
`add(fragment1, fragment2, width)'
Return a fragment which is the addition of the two samples passed
as parameters. WIDTH is the sample width in bytes, either `1',
`2' or `4'. Both fragments should have the same length.
`adpcm2lin(adpcmfragment, width, state)'
Decode an Intel/DVI ADPCM coded fragment to a linear fragment. See
the description of `lin2adpcm()' for details on ADPCM coding.
Return a tuple `(SAMPLE, NEWSTATE)' where the sample has the width
specified in WIDTH.
`adpcm32lin(adpcmfragment, width, state)'
Decode an alternative 3-bit ADPCM code. See `lin2adpcm3()' for
details.
`avg(fragment, width)'
Return the average over all samples in the fragment.
`avgpp(fragment, width)'
Return the average peak-peak value over all samples in the
fragment. No filtering is done, so the usefulness of this routine
is questionable.
`bias(fragment, width, bias)'
Return a fragment that is the original fragment with a bias added
to each sample.
`cross(fragment, width)'
Return the number of zero crossings in the fragment passed as an
argument.
`findfactor(fragment, reference)'
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)'.
`findfit(fragment, reference)'
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()'.
`findmax(fragment, length)'
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)'.
`getsample(fragment, width, index)'
Return the value of sample INDEX from the fragment.
`lin2lin(fragment, width, newwidth)'
Convert samples between 1-, 2- and 4-byte formats.
`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 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.
`lin2adpcm3(fragment, width, state)'
This is an alternative ADPCM coder that uses only 3 bits per
sample. It is not compatible with the Intel/DVI ADPCM coder and
its output is not packed (due to laziness on the side of the
author). Its use is discouraged.
`lin2ulaw(fragment, width)'
Convert samples in the audio fragment to u-LAW encoding and return
this as a Python string. u-LAW is an audio encoding format whereby
you get a dynamic range of about 14 bits using only 8 bit samples.
It is used by the Sun audio hardware, among others.
`minmax(fragment, width)'
Return a tuple consisting of the minimum and maximum values of all
samples in the sound fragment.
`max(fragment, width)'
Return the maximum of the _absolute value_ of all samples in a
fragment.
`maxpp(fragment, width)'
Return the maximum peak-peak value in the sound fragment.
`mul(fragment, width, factor)'
Return a fragment that has all samples in the original fragment
multiplied by the floating-point value FACTOR. Overflow is
silently ignored.
`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 a tuple `(NEWFRAGMENT, NEWSTATE)', and NEWSTATE
should be passed to the next call of `ratecv()'.
The WEIGHTA and WEIGHTB arguments are parameters for a simple
digital filter and default to `1' and `0' respectively.
`reverse(fragment, width)'
Reverse the samples in a fragment and returns the modified
fragment.
`rms(fragment, width)'
Return the root-mean-square of the fragment, i.e.
\catcode`_=8
\sqrt{\frac{\sum{{S_{i}}^{2}}}{n}}
This is a measure of the power in an audio signal.
`tomono(fragment, width, lfactor, rfactor)'
Convert a stereo fragment to a mono fragment. The left channel is
multiplied by LFACTOR and the right channel by RFACTOR before
adding the two channels to give a mono signal.
`tostereo(fragment, width, lfactor, rfactor)'
Generate a stereo fragment from a mono fragment. Each pair of
samples in the stereo fragment are computed from the mono sample,
whereby left channel samples are multiplied by LFACTOR and right
channel samples by RFACTOR.
`ulaw2lin(fragment, width)'
Convert sound fragments in u-LAW encoding to linearly encoded sound
fragments. u-LAW encoding always uses 8 bits samples, so WIDTH
refers only to the sample width of the output fragment here.
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)
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