12.1. Built-in Module audioop

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 -- exception of module audioop

This exception is raised on all errors, such as unknown number of bytes per sample, etc.

add (fragment1, fragment2, width) -- function of module audioop

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) -- function of module audioop

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) -- function of module audioop

Decode an alternative 3-bit ADPCM code. See lin2adpcm3 for details.

avg (fragment, width) -- function of module audioop

Return the average over all samples in the fragment.

avgpp (fragment, width) -- function of module audioop

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) -- function of module audioop

Return a fragment that is the original fragment with a bias added to each sample.

cross (fragment, width) -- function of module audioop

Return the number of zero crossings in the fragment passed as an argument.

findfactor (fragment, reference) -- function of module audioop

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) -- function of module audioop

This routine (which only accepts 2-byte sample fragments)

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) -- function of module audioop

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) -- function of module audioop

Return the value of sample index from the fragment.

lin2lin (fragment, width, newwidth) -- function of module audioop

Convert samples between 1-, 2- and 4-byte formats.

lin2adpcm (fragment, width, state) -- function of module audioop

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) -- function of module audioop

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) -- function of module audioop

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) -- function of module audioop

Return a tuple consisting of the minimum and maximum values of all samples in the sound fragment.

max (fragment, width) -- function of module audioop

Return the maximum of the absolute value of all samples in a fragment.

maxpp (fragment, width) -- function of module audioop

Return the maximum peak-peak value in the sound fragment.

mul (fragment, width, factor) -- function of module audioop

Return a fragment that has all samples in the original framgent multiplied by the floating-point value factor. Overflow is silently ignored.

reverse (fragment, width) -- function of module audioop

Reverse the samples in a fragment and returns the modified fragment.

rms (fragment, width) -- function of module audioop

Return the root-mean-square of the fragment, i.e. the square root of the quotient of the sum of all squared sample value, divided by the sumber of samples. This is a measure of the power in an audio signal.

tomono (fragment, width, lfactor, rfactor) -- function of module audioop

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) -- function of module audioop

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) -- function of module audioop

Convert sound fragments in ULAW encoding to linearly encoded sound fragments. ULAW encoding always uses 8 bits samples, so width refers only to the sample width of the output fragment here.

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)

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)