:mod:`audioop` --- Manipulate raw audio data

============================================

.. module:: audioop

   :synopsis: Manipulate raw audio data.

The :mod:`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

:mod:`al` and :mod:`sunaudiodev` modules.  All scalar items are integers, unless

specified otherwise.

.. index::

   single: Intel/DVI ADPCM

   single: ADPCM, Intel/DVI

   single: a-LAW

   single: u-LAW

This module provides support for a-LAW, u-LAW and Intel/DVI ADPCM encodings.

.. This para is mostly here to provide an excuse for the index entries...

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:

.. exception:: error

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

   sample, etc.

.. function:: 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.

.. function:: adpcm2lin(adpcmfragment, width, state)

   Decode an Intel/DVI ADPCM coded fragment to a linear fragment.  See the

   description of :func:`lin2adpcm` for details on ADPCM coding. Return a tuple

   ``(sample, newstate)`` where the sample has the width specified in *width*.

.. function:: alaw2lin(fragment, width)

   Convert sound fragments in a-LAW encoding to linearly encoded sound fragments.

   a-LAW encoding always uses 8 bits samples, so *width* refers only to the sample

   width of the output fragment here.

   .. versionadded:: 2.5

.. function:: avg(fragment, width)

   Return the average over all samples in the fragment.

.. function:: 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.

.. function:: bias(fragment, width, bias)

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

   sample.

.. function:: cross(fragment, width)

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

.. function:: 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)``.

.. function:: 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 :func:`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 :func:`findfactor`.

.. function:: 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)``.

.. function:: getsample(fragment, width, index)

   Return the value of sample *index* from the fragment.

.. function:: 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 :func:`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.

.. function:: lin2alaw(fragment, width)

   Convert samples in the audio fragment to a-LAW encoding and return this as a

   Python string.  a-LAW is an audio encoding format whereby you get a dynamic

   range of about 13 bits using only 8 bit samples.  It is used by the Sun audio

   hardware, among others.

   .. versionadded:: 2.5

.. function:: lin2lin(fragment, width, newwidth)

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

   .. note::

      In some audio formats, such as .WAV files, 16 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 or 32

      bit width samples.

.. function:: 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.

.. function:: minmax(fragment, width)

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

   the sound fragment.

.. function:: max(fragment, width)

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

.. function:: maxpp(fragment, width)

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

.. function:: 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.

.. function:: 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 :func:`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 ``1`` and ``0`` respectively.

.. function:: reverse(fragment, width)

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

.. function:: rms(fragment, width)

   Return the root-mean-square of the fragment, i.e. ``sqrt(sum(S_i^2)/n)``.

   This is a measure of the power in an audio signal.

.. function:: 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.

.. function:: 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*.

.. function:: 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 :func:`mul` or :func:`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 :func:`lin2adpcm`) along to the decoder, not the

final state (as returned by the coder).  If you want to use

:func:`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 :func:`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)