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author | Guido van Rossum <guido@python.org> | 1994-01-02 01:22:07 (GMT) |
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committer | Guido van Rossum <guido@python.org> | 1994-01-02 01:22:07 (GMT) |
commit | 5fdeeeae2a12b9956cc84d62eae82f72cabc8664 (patch) | |
tree | ac0053479e10099850c8e0d06e31cb3afbf632bb /Doc/libaudioop.tex | |
parent | 0b0719866e8a32d0a787e73bca9e79df1d1a74f8 (diff) | |
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Restructured library documentation
Diffstat (limited to 'Doc/libaudioop.tex')
-rw-r--r-- | Doc/libaudioop.tex | 241 |
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diff --git a/Doc/libaudioop.tex b/Doc/libaudioop.tex new file mode 100644 index 0000000..734065a --- /dev/null +++ b/Doc/libaudioop.tex @@ -0,0 +1,241 @@ +\section{Built-in module \sectcode{audioop}} +\bimodindex{audioop} + +The audioop module contains some useful operations on sound fragments. +It operates on sound fragments consisting of signed integer samples of +8, 16 or 32 bits wide, stored in Python strings. This is the same +format as used by the \code{al} and \code{sunaudiodev} modules. All +scalar items are integers, unless specified otherwise. + +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: + +\renewcommand{\indexsubitem}{(in module audioop)} +\begin{excdesc}{error} +This exception is raised on all errors, such as unknown number of bytes +per sample, etc. +\end{excdesc} + +\begin{funcdesc}{add}{fragment1\, fragment2\, width} +This function returns a fragment that is the addition of the two samples +passed as parameters. \var{width} is the sample width in bytes, either +\code{1}, \code{2} or \code{4}. Both fragments should have the same length. +\end{funcdesc} + +\begin{funcdesc}{adpcm2lin}{adpcmfragment\, width\, state} +This routine decodes an Intel/DVI ADPCM coded fragment to a linear +fragment. See the description of \code{lin2adpcm} for details on ADPCM +coding. The routine returns a tuple +\code{(\var{sample}, \var{newstate})} +where the sample has the width specified in \var{width}. +\end{funcdesc} + +\begin{funcdesc}{adpcm32lin}{adpcmfragment\, width\, state} +This routine decodes an alternative 3-bit ADPCM code. See +\code{lin2adpcm3} for details. +\end{funcdesc} + +\begin{funcdesc}{avg}{fragment\, width} +This function returns the average over all samples in the fragment. +\end{funcdesc} + +\begin{funcdesc}{avgpp}{fragment\, width} +This function returns the average peak-peak value over all samples in +the fragment. No filtering is done, so the useability of this routine +is questionable. +\end{funcdesc} + +\begin{funcdesc}{bias}{fragment\, width\, bias} +This function returns a fragment that is the original fragment with a +bias added to each sample. +\end{funcdesc} + +\begin{funcdesc}{cross}{fragment\, width} +This function returns the number of zero crossings in the fragment +passed as an argument. +\end{funcdesc} + +\begin{funcdesc}{findfactor}{fragment\, reference} +This routine (which only accepts 2-byte sample fragments) calculates a +factor \var{F} such that \code{rms(add(fragment, mul(reference, -F)))} +is minimal, i.e. it calculates the factor with which you should +multiply \var{reference} to make it match as good as possible to +\var{fragment}. The fragments should be the same size. + +The time taken by this routine is proportional to \code{len(fragment)}. +\end{funcdesc} + +\begin{funcdesc}{findfit}{fragment\, reference} +This routine (which only accepts 2-byte sample fragments) tries to +match \var{reference} as good as possible to a portion of +\var{fragment} (which should be the longer fragment). It +(conceptually) does this by taking slices out of \var{fragment}, using +\code{findfactor} to compute the best match, and minimizing the +result. +It returns a tuple \code{(\var{offset}, \var{factor})} with offset the +(integer) offset into \var{fragment} where the optimal match started +and \var{factor} the floating-point factor as per findfactor. +\end{funcdesc} + +\begin{funcdesc}{findmax}{fragment\, length} +This routine (which only accepts 2-byte sample fragments) searches +\var{fragment} for a slice of length \var{length} samples (not bytes!) +with maximum energy, i.e. it returns \var{i} for which +\code{rms(fragment[i*2:(i+length)*2])} is maximal. + +The routine takes time proportional to \code{len(fragment)}. +\end{funcdesc} + +\begin{funcdesc}{getsample}{fragment\, width\, index} +This function returns the value of sample \var{index} from the +fragment. +\end{funcdesc} + +\begin{funcdesc}{lin2lin}{fragment\, width\, newwidth} +This function converts samples between 1-, 2- and 4-byte formats. +\end{funcdesc} + +\begin{funcdesc}{lin2adpcm}{fragment\, width\, state} +This function converts 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 algorythm has been selected for +use by the IMA, so may well become a standard. + +\code{State} is a tuple containing the state of the coder. The coder +returns a tuple \code{(\var{adpcmfrag}, \var{newstate})}, and the +\var{newstate} should be passed to the next call of lin2adpcm. In the +initial call \code{None} can be passed as the state. \var{adpcmfrag} is +the ADPCM coded fragment packed 2 4-bit values per byte. +\end{funcdesc} + +\begin{funcdesc}{lin2adpcm3}{fragment\, width\, state} +This is an alternative ADPCM coder that uses only 3 bits per sample. +It is not compatible with the Intel/DVI ADPCM coder and its output is +not packed (due to laziness on the side of the author). Its use is +discouraged. +\end{funcdesc} + +\begin{funcdesc}{lin2ulaw}{fragment\, width} +This function converts samples in the audio fragment to U-LAW encoding +and returns 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. +\end{funcdesc} + +\begin{funcdesc}{minmax}{fragment\, width} +This function returns a tuple consisting of the minimum and maximum +values of all samples in the sound fragment. +\end{funcdesc} + +\begin{funcdesc}{max}{fragment\, width} +This function returns the maximum of the {\em absolute value} of all +samples in a fragment. +\end{funcdesc} + +\begin{funcdesc}{maxpp}{fragment\, width} +This function returns the maximum peak-peak value in the sound fragment. +\end{funcdesc} + +\begin{funcdesc}{mul}{fragment\, width\, factor} +Mul returns a fragment that has all samples in the original framgent +multiplied by the floating-point value \var{factor}. Overflow is +silently ignored. +\end{funcdesc} + +\begin{funcdesc}{reverse}{fragment\, width} +This function reverses the samples in a fragment and returns the +modified fragment. +\end{funcdesc} + +\begin{funcdesc}{tomono}{fragment\, width\, lfactor\, rfactor} +This function converts a stereo fragment to a mono fragment. The left +channel is multiplied 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} +This function generates 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 \var{lfactor} +and right channel samples by \var{rfactor}. +\end{funcdesc} + +\begin{funcdesc}{mul}{fragment\, width\, factor} +Mul returns a fragment that has all samples in the original framgent +multiplied by the floating-point value \var{factor}. Overflow is +silently ignored. +\end{funcdesc} + +\begin{funcdesc}{rms}{fragment\, width\, factor} +Returns the root-mean-square of the fragment, i.e. +\iftexi +the square root of the quotient of the sum of all squared sample value, +divided by the sumber of samples. +\else +% in eqn: sqrt { sum S sub i sup 2 over n } +\begin{displaymath} +\catcode`_=8 +\sqrt{\frac{\sum{{S_{i}}^{2}}}{n}} +\end{displaymath} +\fi +This is a measure of the power in an audio signal. +\end{funcdesc} + +\begin{funcdesc}{ulaw2lin}{fragment\, width} +This function converts sound fragments in ULAW encoding to linearly +encoded sound fragments. ULAW 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 \code{mul} or \code{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: +\bcode\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}\ecode + +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 \var{initial} state (the one you passed to +lin2adpcm) along to the decoder, not the final state (as returned by +the coder). If you want to use \code{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 \code{find...} routines might look a bit funny at first sight. +They are primarily meant for doing 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: +\bcode\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}\ecode |