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6.1. Built-in module `audioop`

The audioop module contains some useful operations on sound fragments. It operates on sound fragments consisting of signed integer samples of 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:

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: This function returns a fragment that 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: This routine decodes an Intel/DVI ADPCM coded fragment to a linear fragment. See the description of lin2adpcm for details on ADPCM coding. The routine returns a tuple (sample, newstate) where the sample has the width specified in width.

adpcm32lin (adpcmfragment, width, state) -- function of module audioop: This routine decodes an alternative 3-bit ADPCM code. See lin2adpcm3 for details.

avg (fragment, width) -- function of module audioop: This function returns the average over all samples in the fragment.

avgpp (fragment, width) -- function of module audioop: This function returns the average peak-peak value over all samples in the fragment. No filtering is done, so the useability of this routine is questionable.

bias (fragment, width, bias) -- function of module audioop: This function returns a fragment that is the original fragment with a bias added to each sample.

cross (fragment, width) -- function of module audioop: This function returns the number of zero crossings in the fragment passed as an argument.

findfactor (fragment, reference) -- function of module audioop: This routine (which only accepts 2-byte sample fragments) calculates a factor F such that rms(add(fragment, mul(reference, -F))) is minimal, i.e. it calculates the factor with which you should multiply reference to make it match as good as possible to fragment. The fragments should be the same size.
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) tries to match reference as good as possible to a portion of fragment (which should be the longer fragment). It (conceptually) does this by taking slices out of fragment, using findfactor to compute the best match, and minimizing the result. It returns a tuple (offset, factor) with offset the (integer) offset into fragment where the optimal match started and factor the floating-point factor as per findfactor.

findmax (fragment, length) -- function of module audioop: This routine (which only accepts 2-byte sample fragments) searches fragment for a slice of length length samples (not bytes!) with maximum energy, i.e. it returns i for which rms(fragment[i*2:(i+length)*2]) is maximal.
The routine takes time proportional to len(fragment).

getsample (fragment, width, index) -- function of module audioop: This function returns the value of sample index from the fragment.

lin2lin (fragment, width, newwidth) -- function of module audioop: This function converts samples between 1-, 2- and 4-byte formats.

lin2adpcm (fragment, width, state) -- function of module audioop: This function converts 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 algorythm has been selected for use by the IMA, so 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: This function converts samples in the audio fragment to U-LAW encoding and returns 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: This function returns a tuple consisting of the minimum and maximum values of all samples in the sound fragment.

max (fragment, width) -- function of module audioop: This function returns the maximum of the absolute value of all samples in a fragment.

maxpp (fragment, width) -- function of module audioop: This function returns the maximum peak-peak value in the sound fragment.

mul (fragment, width, factor) -- function of module audioop: Mul returns 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: This function reverses the samples in a fragment and returns the modified fragment.

tomono (fragment, width, lfactor, rfactor) -- function of module audioop: This function converts 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: This function generates 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.

mul (fragment, width, factor) -- function of module audioop: Mul returns a fragment that has all samples in the original framgent multiplied by the floating-point value factor. Overflow is silently ignored.

rms (fragment, width, factor) -- function of module audioop: Returns 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.

ulaw2lin (fragment, width) -- function of module audioop: This function converts 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.

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 for doing 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)

6.1. Built-in module audioop

6.1. Built-in module `audioop`