1 2 :mod:`audioop` --- Manipulate raw audio data 3 ============================================ 4 5 .. module:: audioop 6 :synopsis: Manipulate raw audio data. 7 8 9 The :mod:`audioop` module contains some useful operations on sound fragments. 10 It operates on sound fragments consisting of signed integer samples 8, 16 or 32 11 bits wide, stored in Python strings. This is the same format as used by the 12 :mod:`al` and :mod:`sunaudiodev` modules. All scalar items are integers, unless 13 specified otherwise. 14 15 .. index:: 16 single: Intel/DVI ADPCM 17 single: ADPCM, Intel/DVI 18 single: a-LAW 19 single: u-LAW 20 21 This module provides support for a-LAW, u-LAW and Intel/DVI ADPCM encodings. 22 23 .. This para is mostly here to provide an excuse for the index entries... 24 25 A few of the more complicated operations only take 16-bit samples, otherwise the 26 sample size (in bytes) is always a parameter of the operation. 27 28 The module defines the following variables and functions: 29 30 31 .. exception:: error 32 33 This exception is raised on all errors, such as unknown number of bytes per 34 sample, etc. 35 36 37 .. function:: add(fragment1, fragment2, width) 38 39 Return a fragment which is the addition of the two samples passed as parameters. 40 *width* is the sample width in bytes, either ``1``, ``2`` or ``4``. Both 41 fragments should have the same length. Samples are truncated in case of overflow. 42 43 44 .. function:: adpcm2lin(adpcmfragment, width, state) 45 46 Decode an Intel/DVI ADPCM coded fragment to a linear fragment. See the 47 description of :func:`lin2adpcm` for details on ADPCM coding. Return a tuple 48 ``(sample, newstate)`` where the sample has the width specified in *width*. 49 50 51 .. function:: alaw2lin(fragment, width) 52 53 Convert sound fragments in a-LAW encoding to linearly encoded sound fragments. 54 a-LAW encoding always uses 8 bits samples, so *width* refers only to the sample 55 width of the output fragment here. 56 57 .. versionadded:: 2.5 58 59 60 .. function:: avg(fragment, width) 61 62 Return the average over all samples in the fragment. 63 64 65 .. function:: avgpp(fragment, width) 66 67 Return the average peak-peak value over all samples in the fragment. No 68 filtering is done, so the usefulness of this routine is questionable. 69 70 71 .. function:: bias(fragment, width, bias) 72 73 Return a fragment that is the original fragment with a bias added to each 74 sample. Samples wrap around in case of overflow. 75 76 77 .. function:: cross(fragment, width) 78 79 Return the number of zero crossings in the fragment passed as an argument. 80 81 82 .. function:: findfactor(fragment, reference) 83 84 Return a factor *F* such that ``rms(add(fragment, mul(reference, -F)))`` is 85 minimal, i.e., return the factor with which you should multiply *reference* to 86 make it match as well as possible to *fragment*. The fragments should both 87 contain 2-byte samples. 88 89 The time taken by this routine is proportional to ``len(fragment)``. 90 91 92 .. function:: findfit(fragment, reference) 93 94 Try to match *reference* as well as possible to a portion of *fragment* (which 95 should be the longer fragment). This is (conceptually) done by taking slices 96 out of *fragment*, using :func:`findfactor` to compute the best match, and 97 minimizing the result. The fragments should both contain 2-byte samples. 98 Return a tuple ``(offset, factor)`` where *offset* is the (integer) offset into 99 *fragment* where the optimal match started and *factor* is the (floating-point) 100 factor as per :func:`findfactor`. 101 102 103 .. function:: findmax(fragment, length) 104 105 Search *fragment* for a slice of length *length* samples (not bytes!) with 106 maximum energy, i.e., return *i* for which ``rms(fragment[i*2:(i+length)*2])`` 107 is maximal. The fragments should both contain 2-byte samples. 108 109 The routine takes time proportional to ``len(fragment)``. 110 111 112 .. function:: getsample(fragment, width, index) 113 114 Return the value of sample *index* from the fragment. 115 116 117 .. function:: lin2adpcm(fragment, width, state) 118 119 Convert samples to 4 bit Intel/DVI ADPCM encoding. ADPCM coding is an adaptive 120 coding scheme, whereby each 4 bit number is the difference between one sample 121 and the next, divided by a (varying) step. The Intel/DVI ADPCM algorithm has 122 been selected for use by the IMA, so it may well become a standard. 123 124 *state* is a tuple containing the state of the coder. The coder returns a tuple 125 ``(adpcmfrag, newstate)``, and the *newstate* should be passed to the next call 126 of :func:`lin2adpcm`. In the initial call, ``None`` can be passed as the state. 127 *adpcmfrag* is the ADPCM coded fragment packed 2 4-bit values per byte. 128 129 130 .. function:: lin2alaw(fragment, width) 131 132 Convert samples in the audio fragment to a-LAW encoding and return this as a 133 Python string. a-LAW is an audio encoding format whereby you get a dynamic 134 range of about 13 bits using only 8 bit samples. It is used by the Sun audio 135 hardware, among others. 136 137 .. versionadded:: 2.5 138 139 140 .. function:: lin2lin(fragment, width, newwidth) 141 142 Convert samples between 1-, 2- and 4-byte formats. 143 144 .. note:: 145 146 In some audio formats, such as .WAV files, 16 and 32 bit samples are 147 signed, but 8 bit samples are unsigned. So when converting to 8 bit wide 148 samples for these formats, you need to also add 128 to the result:: 149 150 new_frames = audioop.lin2lin(frames, old_width, 1) 151 new_frames = audioop.bias(new_frames, 1, 128) 152 153 The same, in reverse, has to be applied when converting from 8 to 16 or 32 154 bit width samples. 155 156 157 .. function:: lin2ulaw(fragment, width) 158 159 Convert samples in the audio fragment to u-LAW encoding and return this as a 160 Python string. u-LAW is an audio encoding format whereby you get a dynamic 161 range of about 14 bits using only 8 bit samples. It is used by the Sun audio 162 hardware, among others. 163 164 165 .. function:: max(fragment, width) 166 167 Return the maximum of the *absolute value* of all samples in a fragment. 168 169 170 .. function:: maxpp(fragment, width) 171 172 Return the maximum peak-peak value in the sound fragment. 173 174 175 .. function:: minmax(fragment, width) 176 177 Return a tuple consisting of the minimum and maximum values of all samples in 178 the sound fragment. 179 180 181 .. function:: mul(fragment, width, factor) 182 183 Return a fragment that has all samples in the original fragment multiplied by 184 the floating-point value *factor*. Samples are truncated in case of overflow. 185 186 187 .. function:: ratecv(fragment, width, nchannels, inrate, outrate, state[, weightA[, weightB]]) 188 189 Convert the frame rate of the input fragment. 190 191 *state* is a tuple containing the state of the converter. The converter returns 192 a tuple ``(newfragment, newstate)``, and *newstate* should be passed to the next 193 call of :func:`ratecv`. The initial call should pass ``None`` as the state. 194 195 The *weightA* and *weightB* arguments are parameters for a simple digital filter 196 and default to ``1`` and ``0`` respectively. 197 198 199 .. function:: reverse(fragment, width) 200 201 Reverse the samples in a fragment and returns the modified fragment. 202 203 204 .. function:: rms(fragment, width) 205 206 Return the root-mean-square of the fragment, i.e. ``sqrt(sum(S_i^2)/n)``. 207 208 This is a measure of the power in an audio signal. 209 210 211 .. function:: tomono(fragment, width, lfactor, rfactor) 212 213 Convert a stereo fragment to a mono fragment. The left channel is multiplied by 214 *lfactor* and the right channel by *rfactor* before adding the two channels to 215 give a mono signal. 216 217 218 .. function:: tostereo(fragment, width, lfactor, rfactor) 219 220 Generate a stereo fragment from a mono fragment. Each pair of samples in the 221 stereo fragment are computed from the mono sample, whereby left channel samples 222 are multiplied by *lfactor* and right channel samples by *rfactor*. 223 224 225 .. function:: ulaw2lin(fragment, width) 226 227 Convert sound fragments in u-LAW encoding to linearly encoded sound fragments. 228 u-LAW encoding always uses 8 bits samples, so *width* refers only to the sample 229 width of the output fragment here. 230 231 Note that operations such as :func:`.mul` or :func:`.max` make no distinction 232 between mono and stereo fragments, i.e. all samples are treated equal. If this 233 is a problem the stereo fragment should be split into two mono fragments first 234 and recombined later. Here is an example of how to do that:: 235 236 def mul_stereo(sample, width, lfactor, rfactor): 237 lsample = audioop.tomono(sample, width, 1, 0) 238 rsample = audioop.tomono(sample, width, 0, 1) 239 lsample = audioop.mul(lsample, width, lfactor) 240 rsample = audioop.mul(rsample, width, rfactor) 241 lsample = audioop.tostereo(lsample, width, 1, 0) 242 rsample = audioop.tostereo(rsample, width, 0, 1) 243 return audioop.add(lsample, rsample, width) 244 245 If you use the ADPCM coder to build network packets and you want your protocol 246 to be stateless (i.e. to be able to tolerate packet loss) you should not only 247 transmit the data but also the state. Note that you should send the *initial* 248 state (the one you passed to :func:`lin2adpcm`) along to the decoder, not the 249 final state (as returned by the coder). If you want to use 250 :class:`struct.Struct` to store the state in binary you can code the first 251 element (the predicted value) in 16 bits and the second (the delta index) in 8. 252 253 The ADPCM coders have never been tried against other ADPCM coders, only against 254 themselves. It could well be that I misinterpreted the standards in which case 255 they will not be interoperable with the respective standards. 256 257 The :func:`find\*` routines might look a bit funny at first sight. They are 258 primarily meant to do echo cancellation. A reasonably fast way to do this is to 259 pick the most energetic piece of the output sample, locate that in the input 260 sample and subtract the whole output sample from the input sample:: 261 262 def echocancel(outputdata, inputdata): 263 pos = audioop.findmax(outputdata, 800) # one tenth second 264 out_test = outputdata[pos*2:] 265 in_test = inputdata[pos*2:] 266 ipos, factor = audioop.findfit(in_test, out_test) 267 # Optional (for better cancellation): 268 # factor = audioop.findfactor(in_test[ipos*2:ipos*2+len(out_test)], 269 # out_test) 270 prefill = '\0'*(pos+ipos)*2 271 postfill = '\0'*(len(inputdata)-len(prefill)-len(outputdata)) 272 outputdata = prefill + audioop.mul(outputdata, 2, -factor) + postfill 273 return audioop.add(inputdata, outputdata, 2) 274 275
