\section{\module{audioop} ---         Manipulate raw audio data} \declaremodule{builtin}{audioop}\modulesynopsis{Manipulate raw audio data.}  The \module{audioop} module contains some useful operations on soundfragments.  It operates on sound fragments consisting of signedinteger samples 8, 16 or 32 bits wide, stored in Python strings.  Thisis the same format as used by the \refmodule{al} and \refmodule{sunaudiodev}modules.  All scalar items are integers, unless specified otherwise. % This para is mostly here to provide an excuse for the index entries...This module provides support for a-LAW, u-LAW and Intel/DVI ADPCM encodings.\index{Intel/DVI ADPCM}\index{ADPCM, Intel/DVI}\index{a-LAW}\index{u-LAW} A few of the more complicated operations only take 16-bit samples,otherwise the sample size (in bytes) is always a parameter of theoperation. The module defines the following variables and functions: \begin{excdesc}{error}This exception is raised on all errors, such as unknown number of bytesper sample, etc.\end{excdesc} \begin{funcdesc}{add}{fragment1, fragment2, width}Return a fragment which is the addition of the two samples passed asparameters.  \var{width} is the sample width in bytes, either\code{1}, \code{2} or \code{4}.  Both fragments should have the samelength.\end{funcdesc} \begin{funcdesc}{adpcm2lin}{adpcmfragment, width, state}Decode an Intel/DVI ADPCM coded fragment to a linear fragment.  Seethe description of \function{lin2adpcm()} for details on ADPCM coding.Return a tuple \code{(\var{sample}, \var{newstate})} where the samplehas the width specified in \var{width}.\end{funcdesc} \begin{funcdesc}{alaw2lin}{fragment, width}Convert sound fragments in a-LAW encoding to linearly encoded soundfragments.  a-LAW encoding always uses 8 bits samples, so \var{width}refers only to the sample width of the output fragment here.\versionadded{2.5}\end{funcdesc} \begin{funcdesc}{avg}{fragment, width}Return the average over all samples in the fragment.\end{funcdesc} \begin{funcdesc}{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 isquestionable.\end{funcdesc} \begin{funcdesc}{bias}{fragment, width, bias}Return a fragment that is the original fragment with a bias added toeach sample.\end{funcdesc} \begin{funcdesc}{cross}{fragment, width}Return the number of zero crossings in the fragment passed as anargument.\end{funcdesc} \begin{funcdesc}{findfactor}{fragment, reference}Return a factor \var{F} such that\code{rms(add(\var{fragment}, mul(\var{reference}, -\var{F})))} isminimal, i.e., return the factor with which you should multiply\var{reference} to make it match as well as possible to\var{fragment}.  The fragments should both contain 2-byte samples. The time taken by this routine is proportional to\code{len(\var{fragment})}.\end{funcdesc} \begin{funcdesc}{findfit}{fragment, reference}Try to match \var{reference} as well as possible to a portion of\var{fragment} (which should be the longer fragment).  This is(conceptually) done by taking slices out of \var{fragment}, using\function{findfactor()} to compute the best match, and minimizing theresult.  The fragments should both contain 2-byte samples.  Return atuple \code{(\var{offset}, \var{factor})} where \var{offset} is the(integer) offset into \var{fragment} where the optimal match startedand \var{factor} is the (floating-point) factor as per\function{findfactor()}.\end{funcdesc} \begin{funcdesc}{findmax}{fragment, length}Search \var{fragment} for a slice of length \var{length} samples (notbytes!)\ with maximum energy, i.e., return \var{i} for which\code{rms(fragment[i*2:(i+length)*2])} is maximal.  The fragmentsshould both contain 2-byte samples. The routine takes time proportional to \code{len(\var{fragment})}.\end{funcdesc} \begin{funcdesc}{getsample}{fragment, width, index}Return the value of sample \var{index} from the fragment.\end{funcdesc} \begin{funcdesc}{lin2adpcm}{fragment, width, state}Convert samples to 4 bit Intel/DVI ADPCM encoding.  ADPCM coding is anadaptive coding scheme, whereby each 4 bit number is the differencebetween one sample and the next, divided by a (varying) step.  TheIntel/DVI ADPCM algorithm has been selected for use by the IMA, so itmay well become a standard. \var{state} is a tuple containing the state of the coder.  The coderreturns a tuple \code{(\var{adpcmfrag}, \var{newstate})}, and the\var{newstate} should be passed to the next call of\function{lin2adpcm()}.  In the initial call, \code{None} can bepassed as the state.  \var{adpcmfrag} is the ADPCM coded fragmentpacked 2 4-bit values per byte.\end{funcdesc} \begin{funcdesc}{lin2alaw}{fragment, width}Convert samples in the audio fragment to a-LAW encoding and returnthis as a Python string.  a-LAW is an audio encoding format wherebyyou get a dynamic range of about 13 bits using only 8 bit samples.  Itis used by the Sun audio hardware, among others.\versionadded{2.5}\end{funcdesc} \begin{funcdesc}{lin2lin}{fragment, width, newwidth}Convert samples between 1-, 2- and 4-byte formats.\end{funcdesc} \begin{funcdesc}{lin2ulaw}{fragment, width}Convert samples in the audio fragment to u-LAW encoding and returnthis as a Python string.  u-LAW is an audio encoding format wherebyyou get a dynamic range of about 14 bits using only 8 bit samples.  Itis used by the Sun audio hardware, among others.\end{funcdesc} \begin{funcdesc}{minmax}{fragment, width}Return a tuple consisting of the minimum and maximum values of allsamples in the sound fragment.\end{funcdesc} \begin{funcdesc}{max}{fragment, width}Return the maximum of the \emph{absolute value} of all samples in afragment.\end{funcdesc} \begin{funcdesc}{maxpp}{fragment, width}Return the maximum peak-peak value in the sound fragment.\end{funcdesc} \begin{funcdesc}{mul}{fragment, width, factor}Return a fragment that has all samples in the original fragmentmultiplied by the floating-point value \var{factor}.  Overflow issilently ignored.\end{funcdesc} \begin{funcdesc}{ratecv}{fragment, width, nchannels, inrate, outrate,                         state\optional{, weightA\optional{, weightB}}}Convert the frame rate of the input fragment. \var{state} is a tuple containing the state of the converter.  Theconverter returns a tuple \code{(\var{newfragment}, \var{newstate})},and \var{newstate} should be passed to the next call of\function{ratecv()}.  The initial call should pass \code{None}as the state. The \var{weightA} and \var{weightB} arguments are parameters for asimple digital filter and default to \code{1} and \code{0} respectively.\end{funcdesc} \begin{funcdesc}{reverse}{fragment, width}Reverse the samples in a fragment and returns the modified fragment.\end{funcdesc} \begin{funcdesc}{rms}{fragment, width}Return the root-mean-square of the fragment, i.e.\begin{displaymath}\catcode`_=8\sqrt{\frac{\sum{{S_{i}}^{2}}}{n}}\end{displaymath}This is a measure of the power in an audio signal.\end{funcdesc} \begin{funcdesc}{tomono}{fragment, width, lfactor, rfactor} Convert a stereo fragment to a mono fragment.  The left channel ismultiplied by \var{lfactor} and the right channel by \var{rfactor}before adding the two channels to give a mono signal.\end{funcdesc} \begin{funcdesc}{tostereo}{fragment, width, lfactor, rfactor}Generate a stereo fragment from a mono fragment.  Each pair of samplesin the stereo fragment are computed from the mono sample, whereby leftchannel samples are multiplied by \var{lfactor} and right channelsamples by \var{rfactor}.\end{funcdesc} \begin{funcdesc}{ulaw2lin}{fragment, width}Convert sound fragments in u-LAW encoding to linearly encoded soundfragments.  u-LAW encoding always uses 8 bits samples, so \var{width}refers only to the sample width of the output fragment here.\end{funcdesc} Note that operations such as \function{mul()} or \function{max()} makeno distinction between mono and stereo fragments, i.e.\ all samplesare treated equal.  If this is a problem the stereo fragment should besplit into two mono fragments first and recombined later.  Here is anexample of how to do that: \begin{verbatim}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)\end{verbatim} If you use the ADPCM coder to build network packets and you want yourprotocol to be stateless (i.e.\ to be able to tolerate packet loss)you should not only transmit the data but also the state.  Note thatyou should send the \var{initial} state (the one you passed to\function{lin2adpcm()}) along to the decoder, not the final state (asreturned by the coder).  If you want to use \function{struct.struct()}to store the state in binary you can code the first element (thepredicted 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 thestandards in which case they will not be interoperable with therespective standards. The \function{find*()} routines might look a bit funny at first sight.They are primarily meant to do echo cancellation.  A reasonablyfast way to do this is to pick the most energetic piece of the outputsample, locate that in the input sample and subtract the whole outputsample from the input sample: \begin{verbatim}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)\end{verbatim}