mirror of
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f8316638af
SourceForge doesn't choke on this batch :-)
I'm not entirely sure this is 100% correct. The patch changes an
\index{persistency} to \index{presistence}, and I don't know what \index{}
does. But it seems to do so persi--er, consistently, so I hope it isn't a
problem.
255 lines
9.9 KiB
TeX
255 lines
9.9 KiB
TeX
\section{\module{audioop} ---
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Manipulate raw audio data}
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\declaremodule{builtin}{audioop}
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\modulesynopsis{Manipulate raw audio data.}
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The \module{audioop} module contains some useful operations on sound
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fragments. It operates on sound fragments consisting of signed
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integer samples 8, 16 or 32 bits wide, stored in Python strings. This
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is the same format as used by the \refmodule{al} and \refmodule{sunaudiodev}
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modules. All scalar items are integers, unless specified otherwise.
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% This para is mostly here to provide an excuse for the index entries...
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This module provides support for u-LAW and Intel/DVI ADPCM encodings.
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\index{Intel/DVI ADPCM}
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\index{ADPCM, Intel/DVI}
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\index{u-LAW}
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A few of the more complicated operations only take 16-bit samples,
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otherwise the sample size (in bytes) is always a parameter of the
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operation.
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The module defines the following variables and functions:
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\begin{excdesc}{error}
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This exception is raised on all errors, such as unknown number of bytes
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per sample, etc.
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\end{excdesc}
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\begin{funcdesc}{add}{fragment1, fragment2, width}
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Return a fragment which is the addition of the two samples passed as
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parameters. \var{width} is the sample width in bytes, either
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\code{1}, \code{2} or \code{4}. Both fragments should have the same
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length.
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\end{funcdesc}
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\begin{funcdesc}{adpcm2lin}{adpcmfragment, width, state}
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Decode an Intel/DVI ADPCM coded fragment to a linear fragment. See
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the description of \function{lin2adpcm()} for details on ADPCM coding.
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Return a tuple \code{(\var{sample}, \var{newstate})} where the sample
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has the width specified in \var{width}.
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\end{funcdesc}
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\begin{funcdesc}{adpcm32lin}{adpcmfragment, width, state}
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Decode an alternative 3-bit ADPCM code. See \function{lin2adpcm3()}
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for details.
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\end{funcdesc}
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\begin{funcdesc}{avg}{fragment, width}
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Return the average over all samples in the fragment.
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\end{funcdesc}
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\begin{funcdesc}{avgpp}{fragment, width}
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Return the average peak-peak value over all samples in the fragment.
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No filtering is done, so the usefulness of this routine is
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questionable.
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\end{funcdesc}
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\begin{funcdesc}{bias}{fragment, width, bias}
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Return a fragment that is the original fragment with a bias added to
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each sample.
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\end{funcdesc}
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\begin{funcdesc}{cross}{fragment, width}
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Return the number of zero crossings in the fragment passed as an
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argument.
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\end{funcdesc}
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\begin{funcdesc}{findfactor}{fragment, reference}
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Return a factor \var{F} such that
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\code{rms(add(\var{fragment}, mul(\var{reference}, -\var{F})))} is
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minimal, i.e., return the factor with which you should multiply
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\var{reference} to make it match as well as possible to
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\var{fragment}. The fragments should both contain 2-byte samples.
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The time taken by this routine is proportional to
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\code{len(\var{fragment})}.
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\end{funcdesc}
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\begin{funcdesc}{findfit}{fragment, reference}
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Try to match \var{reference} as well as possible to a portion of
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\var{fragment} (which should be the longer fragment). This is
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(conceptually) done by taking slices out of \var{fragment}, using
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\function{findfactor()} to compute the best match, and minimizing the
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result. The fragments should both contain 2-byte samples. Return a
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tuple \code{(\var{offset}, \var{factor})} where \var{offset} is the
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(integer) offset into \var{fragment} where the optimal match started
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and \var{factor} is the (floating-point) factor as per
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\function{findfactor()}.
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\end{funcdesc}
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\begin{funcdesc}{findmax}{fragment, length}
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Search \var{fragment} for a slice of length \var{length} samples (not
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bytes!)\ with maximum energy, i.e., return \var{i} for which
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\code{rms(fragment[i*2:(i+length)*2])} is maximal. The fragments
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should both contain 2-byte samples.
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The routine takes time proportional to \code{len(\var{fragment})}.
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\end{funcdesc}
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\begin{funcdesc}{getsample}{fragment, width, index}
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Return the value of sample \var{index} from the fragment.
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\end{funcdesc}
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\begin{funcdesc}{lin2lin}{fragment, width, newwidth}
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Convert samples between 1-, 2- and 4-byte formats.
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\end{funcdesc}
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\begin{funcdesc}{lin2adpcm}{fragment, width, state}
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Convert samples to 4 bit Intel/DVI ADPCM encoding. ADPCM coding is an
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adaptive coding scheme, whereby each 4 bit number is the difference
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between one sample and the next, divided by a (varying) step. The
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Intel/DVI ADPCM algorithm has been selected for use by the IMA, so it
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may well become a standard.
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\var{state} is a tuple containing the state of the coder. The coder
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returns a tuple \code{(\var{adpcmfrag}, \var{newstate})}, and the
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\var{newstate} should be passed to the next call of
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\function{lin2adpcm()}. In the initial call, \code{None} can be
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passed as the state. \var{adpcmfrag} is the ADPCM coded fragment
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packed 2 4-bit values per byte.
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\end{funcdesc}
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\begin{funcdesc}{lin2adpcm3}{fragment, width, state}
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This is an alternative ADPCM coder that uses only 3 bits per sample.
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It is not compatible with the Intel/DVI ADPCM coder and its output is
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not packed (due to laziness on the side of the author). Its use is
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discouraged.
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\end{funcdesc}
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\begin{funcdesc}{lin2ulaw}{fragment, width}
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Convert samples in the audio fragment to u-LAW encoding and return
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this as a Python string. u-LAW is an audio encoding format whereby
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you get a dynamic range of about 14 bits using only 8 bit samples. It
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is used by the Sun audio hardware, among others.
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\end{funcdesc}
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\begin{funcdesc}{minmax}{fragment, width}
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Return a tuple consisting of the minimum and maximum values of all
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samples in the sound fragment.
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\end{funcdesc}
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\begin{funcdesc}{max}{fragment, width}
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Return the maximum of the \emph{absolute value} of all samples in a
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fragment.
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\end{funcdesc}
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\begin{funcdesc}{maxpp}{fragment, width}
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Return the maximum peak-peak value in the sound fragment.
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\end{funcdesc}
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\begin{funcdesc}{mul}{fragment, width, factor}
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Return a fragment that has all samples in the original fragment
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multiplied by the floating-point value \var{factor}. Overflow is
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silently ignored.
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\end{funcdesc}
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\begin{funcdesc}{ratecv}{fragment, width, nchannels, inrate, outrate,
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state\optional{, weightA\optional{, weightB}}}
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Convert the frame rate of the input fragment.
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\var{state} is a tuple containing the state of the converter. The
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converter returns a tuple \code{(\var{newfragment}, \var{newstate})},
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and \var{newstate} should be passed to the next call of
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\function{ratecv()}.
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The \var{weightA} and \var{weightB} arguments are parameters for a
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simple digital filter and default to \code{1} and \code{0} respectively.
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\end{funcdesc}
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\begin{funcdesc}{reverse}{fragment, width}
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Reverse the samples in a fragment and returns the modified fragment.
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\end{funcdesc}
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\begin{funcdesc}{rms}{fragment, width}
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Return the root-mean-square of the fragment, i.e.
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\begin{displaymath}
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\catcode`_=8
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\sqrt{\frac{\sum{{S_{i}}^{2}}}{n}}
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\end{displaymath}
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This is a measure of the power in an audio signal.
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\end{funcdesc}
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\begin{funcdesc}{tomono}{fragment, width, lfactor, rfactor}
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Convert a stereo fragment to a mono fragment. The left channel is
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multiplied by \var{lfactor} and the right channel by \var{rfactor}
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before adding the two channels to give a mono signal.
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\end{funcdesc}
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\begin{funcdesc}{tostereo}{fragment, width, lfactor, rfactor}
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Generate a stereo fragment from a mono fragment. Each pair of samples
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in the stereo fragment are computed from the mono sample, whereby left
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channel samples are multiplied by \var{lfactor} and right channel
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samples by \var{rfactor}.
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\end{funcdesc}
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\begin{funcdesc}{ulaw2lin}{fragment, width}
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Convert sound fragments in u-LAW encoding to linearly encoded sound
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fragments. u-LAW encoding always uses 8 bits samples, so \var{width}
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refers only to the sample width of the output fragment here.
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\end{funcdesc}
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Note that operations such as \function{mul()} or \function{max()} make
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no distinction between mono and stereo fragments, i.e.\ all samples
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are treated equal. If this is a problem the stereo fragment should be
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split into two mono fragments first and recombined later. Here is an
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example of how to do that:
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\begin{verbatim}
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def mul_stereo(sample, width, lfactor, rfactor):
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lsample = audioop.tomono(sample, width, 1, 0)
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rsample = audioop.tomono(sample, width, 0, 1)
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lsample = audioop.mul(sample, width, lfactor)
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rsample = audioop.mul(sample, width, rfactor)
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lsample = audioop.tostereo(lsample, width, 1, 0)
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rsample = audioop.tostereo(rsample, width, 0, 1)
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return audioop.add(lsample, rsample, width)
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\end{verbatim}
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If you use the ADPCM coder to build network packets and you want your
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protocol to be stateless (i.e.\ to be able to tolerate packet loss)
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you should not only transmit the data but also the state. Note that
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you should send the \var{initial} state (the one you passed to
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\function{lin2adpcm()}) along to the decoder, not the final state (as
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returned by the coder). If you want to use \function{struct.struct()}
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to store the state in binary you can code the first element (the
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predicted value) in 16 bits and the second (the delta index) in 8.
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The ADPCM coders have never been tried against other ADPCM coders,
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only against themselves. It could well be that I misinterpreted the
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standards in which case they will not be interoperable with the
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respective standards.
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The \function{find*()} routines might look a bit funny at first sight.
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They are primarily meant to do echo cancellation. A reasonably
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fast way to do this is to pick the most energetic piece of the output
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sample, locate that in the input sample and subtract the whole output
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sample from the input sample:
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\begin{verbatim}
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def echocancel(outputdata, inputdata):
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pos = audioop.findmax(outputdata, 800) # one tenth second
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out_test = outputdata[pos*2:]
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in_test = inputdata[pos*2:]
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ipos, factor = audioop.findfit(in_test, out_test)
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# Optional (for better cancellation):
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# factor = audioop.findfactor(in_test[ipos*2:ipos*2+len(out_test)],
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# out_test)
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prefill = '\0'*(pos+ipos)*2
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postfill = '\0'*(len(inputdata)-len(prefill)-len(outputdata))
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outputdata = prefill + audioop.mul(outputdata,2,-factor) + postfill
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return audioop.add(inputdata, outputdata, 2)
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\end{verbatim}
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