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+
+: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)
+