1:mod:`audioop` --- Manipulate raw audio data 2============================================ 3 4.. module:: audioop 5 :synopsis: Manipulate raw audio data. 6 7-------------- 8 9The :mod:`audioop` module contains some useful operations on sound fragments. 10It operates on sound fragments consisting of signed integer samples 8, 16, 24 11or 32 bits wide, stored in :term:`bytes-like objects <bytes-like object>`. All scalar items are 12integers, unless specified otherwise. 13 14.. versionchanged:: 3.4 15 Support for 24-bit samples was added. 16 All functions now accept any :term:`bytes-like object`. 17 String input now results in an immediate error. 18 19.. index:: 20 single: Intel/DVI ADPCM 21 single: ADPCM, Intel/DVI 22 single: a-LAW 23 single: u-LAW 24 25This module provides support for a-LAW, u-LAW and Intel/DVI ADPCM encodings. 26 27.. This para is mostly here to provide an excuse for the index entries... 28 29A few of the more complicated operations only take 16-bit samples, otherwise the 30sample size (in bytes) is always a parameter of the operation. 31 32The module defines the following variables and functions: 33 34 35.. exception:: error 36 37 This exception is raised on all errors, such as unknown number of bytes per 38 sample, etc. 39 40 41.. function:: add(fragment1, fragment2, width) 42 43 Return a fragment which is the addition of the two samples passed as parameters. 44 *width* is the sample width in bytes, either ``1``, ``2``, ``3`` or ``4``. Both 45 fragments should have the same length. Samples are truncated in case of overflow. 46 47 48.. function:: adpcm2lin(adpcmfragment, width, state) 49 50 Decode an Intel/DVI ADPCM coded fragment to a linear fragment. See the 51 description of :func:`lin2adpcm` for details on ADPCM coding. Return a tuple 52 ``(sample, newstate)`` where the sample has the width specified in *width*. 53 54 55.. function:: alaw2lin(fragment, width) 56 57 Convert sound fragments in a-LAW encoding to linearly encoded sound fragments. 58 a-LAW encoding always uses 8 bits samples, so *width* refers only to the sample 59 width of the output fragment here. 60 61 62.. function:: avg(fragment, width) 63 64 Return the average over all samples in the fragment. 65 66 67.. function:: avgpp(fragment, width) 68 69 Return the average peak-peak value over all samples in the fragment. No 70 filtering is done, so the usefulness of this routine is questionable. 71 72 73.. function:: bias(fragment, width, bias) 74 75 Return a fragment that is the original fragment with a bias added to each 76 sample. Samples wrap around in case of overflow. 77 78 79.. function:: byteswap(fragment, width) 80 81 "Byteswap" all samples in a fragment and returns the modified fragment. 82 Converts big-endian samples to little-endian and vice versa. 83 84 .. versionadded:: 3.4 85 86 87.. function:: cross(fragment, width) 88 89 Return the number of zero crossings in the fragment passed as an argument. 90 91 92.. function:: findfactor(fragment, reference) 93 94 Return a factor *F* such that ``rms(add(fragment, mul(reference, -F)))`` is 95 minimal, i.e., return the factor with which you should multiply *reference* to 96 make it match as well as possible to *fragment*. The fragments should both 97 contain 2-byte samples. 98 99 The time taken by this routine is proportional to ``len(fragment)``. 100 101 102.. function:: findfit(fragment, reference) 103 104 Try to match *reference* as well as possible to a portion of *fragment* (which 105 should be the longer fragment). This is (conceptually) done by taking slices 106 out of *fragment*, using :func:`findfactor` to compute the best match, and 107 minimizing the result. The fragments should both contain 2-byte samples. 108 Return a tuple ``(offset, factor)`` where *offset* is the (integer) offset into 109 *fragment* where the optimal match started and *factor* is the (floating-point) 110 factor as per :func:`findfactor`. 111 112 113.. function:: findmax(fragment, length) 114 115 Search *fragment* for a slice of length *length* samples (not bytes!) with 116 maximum energy, i.e., return *i* for which ``rms(fragment[i*2:(i+length)*2])`` 117 is maximal. The fragments should both contain 2-byte samples. 118 119 The routine takes time proportional to ``len(fragment)``. 120 121 122.. function:: getsample(fragment, width, index) 123 124 Return the value of sample *index* from the fragment. 125 126 127.. function:: lin2adpcm(fragment, width, state) 128 129 Convert samples to 4 bit Intel/DVI ADPCM encoding. ADPCM coding is an adaptive 130 coding scheme, whereby each 4 bit number is the difference between one sample 131 and the next, divided by a (varying) step. The Intel/DVI ADPCM algorithm has 132 been selected for use by the IMA, so it may well become a standard. 133 134 *state* is a tuple containing the state of the coder. The coder returns a tuple 135 ``(adpcmfrag, newstate)``, and the *newstate* should be passed to the next call 136 of :func:`lin2adpcm`. In the initial call, ``None`` can be passed as the state. 137 *adpcmfrag* is the ADPCM coded fragment packed 2 4-bit values per byte. 138 139 140.. function:: lin2alaw(fragment, width) 141 142 Convert samples in the audio fragment to a-LAW encoding and return this as a 143 bytes object. a-LAW is an audio encoding format whereby you get a dynamic 144 range of about 13 bits using only 8 bit samples. It is used by the Sun audio 145 hardware, among others. 146 147 148.. function:: lin2lin(fragment, width, newwidth) 149 150 Convert samples between 1-, 2-, 3- and 4-byte formats. 151 152 .. note:: 153 154 In some audio formats, such as .WAV files, 16, 24 and 32 bit samples are 155 signed, but 8 bit samples are unsigned. So when converting to 8 bit wide 156 samples for these formats, you need to also add 128 to the result:: 157 158 new_frames = audioop.lin2lin(frames, old_width, 1) 159 new_frames = audioop.bias(new_frames, 1, 128) 160 161 The same, in reverse, has to be applied when converting from 8 to 16, 24 162 or 32 bit width samples. 163 164 165.. function:: lin2ulaw(fragment, width) 166 167 Convert samples in the audio fragment to u-LAW encoding and return this as a 168 bytes object. u-LAW is an audio encoding format whereby you get a dynamic 169 range of about 14 bits using only 8 bit samples. It is used by the Sun audio 170 hardware, among others. 171 172 173.. function:: max(fragment, width) 174 175 Return the maximum of the *absolute value* of all samples in a fragment. 176 177 178.. function:: maxpp(fragment, width) 179 180 Return the maximum peak-peak value in the sound fragment. 181 182 183.. function:: minmax(fragment, width) 184 185 Return a tuple consisting of the minimum and maximum values of all samples in 186 the sound fragment. 187 188 189.. function:: mul(fragment, width, factor) 190 191 Return a fragment that has all samples in the original fragment multiplied by 192 the floating-point value *factor*. Samples are truncated in case of overflow. 193 194 195.. function:: ratecv(fragment, width, nchannels, inrate, outrate, state[, weightA[, weightB]]) 196 197 Convert the frame rate of the input fragment. 198 199 *state* is a tuple containing the state of the converter. The converter returns 200 a tuple ``(newfragment, newstate)``, and *newstate* should be passed to the next 201 call of :func:`ratecv`. The initial call should pass ``None`` as the state. 202 203 The *weightA* and *weightB* arguments are parameters for a simple digital filter 204 and default to ``1`` and ``0`` respectively. 205 206 207.. function:: reverse(fragment, width) 208 209 Reverse the samples in a fragment and returns the modified fragment. 210 211 212.. function:: rms(fragment, width) 213 214 Return the root-mean-square of the fragment, i.e. ``sqrt(sum(S_i^2)/n)``. 215 216 This is a measure of the power in an audio signal. 217 218 219.. function:: tomono(fragment, width, lfactor, rfactor) 220 221 Convert a stereo fragment to a mono fragment. The left channel is multiplied by 222 *lfactor* and the right channel by *rfactor* before adding the two channels to 223 give a mono signal. 224 225 226.. function:: tostereo(fragment, width, lfactor, rfactor) 227 228 Generate a stereo fragment from a mono fragment. Each pair of samples in the 229 stereo fragment are computed from the mono sample, whereby left channel samples 230 are multiplied by *lfactor* and right channel samples by *rfactor*. 231 232 233.. function:: ulaw2lin(fragment, width) 234 235 Convert sound fragments in u-LAW encoding to linearly encoded sound fragments. 236 u-LAW encoding always uses 8 bits samples, so *width* refers only to the sample 237 width of the output fragment here. 238 239Note that operations such as :func:`.mul` or :func:`.max` make no distinction 240between mono and stereo fragments, i.e. all samples are treated equal. If this 241is a problem the stereo fragment should be split into two mono fragments first 242and recombined later. Here is an example of how to do that:: 243 244 def mul_stereo(sample, width, lfactor, rfactor): 245 lsample = audioop.tomono(sample, width, 1, 0) 246 rsample = audioop.tomono(sample, width, 0, 1) 247 lsample = audioop.mul(lsample, width, lfactor) 248 rsample = audioop.mul(rsample, width, rfactor) 249 lsample = audioop.tostereo(lsample, width, 1, 0) 250 rsample = audioop.tostereo(rsample, width, 0, 1) 251 return audioop.add(lsample, rsample, width) 252 253If you use the ADPCM coder to build network packets and you want your protocol 254to be stateless (i.e. to be able to tolerate packet loss) you should not only 255transmit the data but also the state. Note that you should send the *initial* 256state (the one you passed to :func:`lin2adpcm`) along to the decoder, not the 257final state (as returned by the coder). If you want to use 258:class:`struct.Struct` to store the state in binary you can code the first 259element (the predicted value) in 16 bits and the second (the delta index) in 8. 260 261The ADPCM coders have never been tried against other ADPCM coders, only against 262themselves. It could well be that I misinterpreted the standards in which case 263they will not be interoperable with the respective standards. 264 265The :func:`find\*` routines might look a bit funny at first sight. They are 266primarily meant to do echo cancellation. A reasonably fast way to do this is to 267pick the most energetic piece of the output sample, locate that in the input 268sample and subtract the whole output sample from the input sample:: 269 270 def echocancel(outputdata, inputdata): 271 pos = audioop.findmax(outputdata, 800) # one tenth second 272 out_test = outputdata[pos*2:] 273 in_test = inputdata[pos*2:] 274 ipos, factor = audioop.findfit(in_test, out_test) 275 # Optional (for better cancellation): 276 # factor = audioop.findfactor(in_test[ipos*2:ipos*2+len(out_test)], 277 # out_test) 278 prefill = '\0'*(pos+ipos)*2 279 postfill = '\0'*(len(inputdata)-len(prefill)-len(outputdata)) 280 outputdata = prefill + audioop.mul(outputdata, 2, -factor) + postfill 281 return audioop.add(inputdata, outputdata, 2) 282 283