# cpython-withatomic / Doc / libaudioop.tex

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  1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 \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 usefulness 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 \var{offset} the (integer) offset into \var{fragment} where the optimal match started and \var{factor} the floating-point factor as per \code{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 algorithm has been selected for use by the IMA, so it 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 `