236 lines
9.6 KiB
TeX
236 lines
9.6 KiB
TeX
\section{Built-in module \sectcode{audioop}}
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\bimodindex{audioop}
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The \code{audioop} module contains some useful operations on sound fragments.
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It operates on sound fragments consisting of signed integer samples
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8, 16 or 32 bits wide, stored in Python strings. This is the same
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format as used by the \code{al} and \code{sunaudiodev} modules. All
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scalar items are integers, unless specified otherwise.
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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 operation.
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The module defines the following variables and functions:
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\renewcommand{\indexsubitem}{(in module audioop)}
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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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This function returns a fragment which is the addition of the two samples
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passed as 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 length.
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\end{funcdesc}
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\begin{funcdesc}{adpcm2lin}{adpcmfragment\, width\, state}
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This routine decodes an Intel/DVI ADPCM coded fragment to a linear
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fragment. See the description of \code{lin2adpcm} for details on ADPCM
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coding. The routine returns a tuple
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\code{(\var{sample}, \var{newstate})}
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where the sample 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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This routine decodes an alternative 3-bit ADPCM code. See
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\code{lin2adpcm3} for details.
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\end{funcdesc}
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\begin{funcdesc}{avg}{fragment\, width}
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This function returns 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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This function returns the average peak-peak value over all samples in
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the fragment. No filtering is done, so the usefulness of this routine
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is questionable.
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\end{funcdesc}
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\begin{funcdesc}{bias}{fragment\, width\, bias}
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This function returns a fragment that is the original fragment with a
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bias added to each sample.
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\end{funcdesc}
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\begin{funcdesc}{cross}{fragment\, width}
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This function returns the number of zero crossings in the fragment
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passed as an argument.
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\end{funcdesc}
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\begin{funcdesc}{findfactor}{fragment\, reference}
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This routine (which only accepts 2-byte sample fragments) calculates a
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factor \var{F} such that \code{rms(add(fragment, mul(reference, -F)))}
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is minimal, i.e.\ it calculates the factor with which you should
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multiply \var{reference} to make it match as well as possible to
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\var{fragment}. The fragments should be the same size.
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The time taken by this routine is proportional to \code{len(fragment)}.
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\end{funcdesc}
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\begin{funcdesc}{findfit}{fragment\, reference}
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This routine (which only accepts 2-byte sample fragments) tries to
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match \var{reference} as well as possible to a portion of
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\var{fragment} (which should be the longer fragment). It
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(conceptually) does this by taking slices out of \var{fragment}, using
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\code{findfactor} to compute the best match, and minimizing the
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result.
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It returns a tuple \code{(\var{offset}, \var{factor})} with \var{offset} the
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(integer) offset into \var{fragment} where the optimal match started
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and \var{factor} the floating-point factor as per \code{findfactor}.
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\end{funcdesc}
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\begin{funcdesc}{findmax}{fragment\, length}
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This routine (which only accepts 2-byte sample fragments) searches
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\var{fragment} for a slice of length \var{length} samples (not bytes!)\
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with maximum energy, i.e.\ it returns \var{i} for which
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\code{rms(fragment[i*2:(i+length)*2])} is maximal.
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The routine takes time proportional to \code{len(fragment)}.
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\end{funcdesc}
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\begin{funcdesc}{getsample}{fragment\, width\, index}
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This function returns the value of sample \var{index} from the
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fragment.
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\end{funcdesc}
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\begin{funcdesc}{lin2lin}{fragment\, width\, newwidth}
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This function converts 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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This function converts samples to 4 bit Intel/DVI ADPCM encoding.
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ADPCM coding is an adaptive coding scheme, whereby each 4 bit number
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is the difference between one sample and the next, divided by a
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(varying) step. The Intel/DVI ADPCM algorithm has been selected for
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use by the IMA, so it may well become a standard.
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\code{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 lin2adpcm. In the
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initial call \code{None} can be passed as the state. \var{adpcmfrag} is
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the ADPCM coded fragment 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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This function converts samples in the audio fragment to U-LAW encoding
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and returns this as a Python string. U-LAW is an audio encoding format
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whereby you get a dynamic range of about 14 bits using only 8 bit
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samples. It 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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This function returns a tuple consisting of the minimum and maximum
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values of all samples in the sound fragment.
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\end{funcdesc}
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\begin{funcdesc}{max}{fragment\, width}
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This function returns the maximum of the {\em absolute value} of all
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samples in a fragment.
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\end{funcdesc}
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\begin{funcdesc}{maxpp}{fragment\, width}
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This function returns 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 framgent
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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}{reverse}{fragment\, width}
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This function reverses the samples in a fragment and returns the
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modified fragment.
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\end{funcdesc}
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\begin{funcdesc}{rms}{fragment\, width\, factor}
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Returns the root-mean-square of the fragment, i.e.
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\iftexi
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the square root of the quotient of the sum of all squared sample value,
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divided by the sumber of samples.
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\else
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% in eqn: sqrt { sum S sub i sup 2 over n }
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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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\fi
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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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This function converts a stereo fragment to a mono fragment. The left
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channel is multiplied by \var{lfactor} and the right channel by
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\var{rfactor} 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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This function generates a stereo fragment from a mono fragment. Each
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pair of samples in the stereo fragment are computed from the mono
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sample, whereby left channel samples are multiplied by \var{lfactor}
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and right channel samples by \var{rfactor}.
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\end{funcdesc}
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\begin{funcdesc}{ulaw2lin}{fragment\, width}
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This function converts sound fragments in ULAW encoding to linearly
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encoded sound fragments. ULAW encoding always uses 8 bits samples, so
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\var{width} 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 \code{mul} or \code{max} make no
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distinction between mono and stereo fragments, i.e.\ all samples are
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treated equal. If this is a problem the stereo fragment should be split
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into two mono fragments first and recombined later. Here is an example
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of how to do that:
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\bcode\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}\ecode
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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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\code{lin2adpcm}) along to the decoder, not the final state (as returned by
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the coder). If you want to use \code{struct} to store the state in
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binary you can code the first element (the predicted value) in 16 bits
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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 \code{find...} routines might look a bit funny at first sight.
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They are primarily meant for doing 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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\bcode\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}\ecode
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