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/Doc/library/audioop.rst

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  1
  2:mod:`audioop` --- Manipulate raw audio data
  3============================================
  4
  5.. module:: audioop
  6   :synopsis: Manipulate raw audio data.
  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 or 32
 11bits 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
 13specified otherwise.
 14
 15.. index::
 16   single: Intel/DVI ADPCM
 17   single: ADPCM, Intel/DVI
 18   single: a-LAW
 19   single: u-LAW
 20
 21This 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
 25A few of the more complicated operations only take 16-bit samples, otherwise the
 26sample size (in bytes) is always a parameter of the operation.
 27
 28The 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.
 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.
 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:: minmax(fragment, width)
166
167   Return a tuple consisting of the minimum and maximum values of all samples in
168   the sound fragment.
169
170
171.. function:: max(fragment, width)
172
173   Return the maximum of the *absolute value* of all samples in a fragment.
174
175
176.. function:: maxpp(fragment, width)
177
178   Return the maximum peak-peak value in 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*.  Overflow is silently ignored.
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
231Note that operations such as :func:`mul` or :func:`max` make no distinction
232between mono and stereo fragments, i.e. all samples are treated equal.  If this
233is a problem the stereo fragment should be split into two mono fragments first
234and 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(sample, width, lfactor)
240       rsample = audioop.mul(sample, 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
245If you use the ADPCM coder to build network packets and you want your protocol
246to be stateless (i.e. to be able to tolerate packet loss) you should not only
247transmit the data but also the state.  Note that you should send the *initial*
248state (the one you passed to :func:`lin2adpcm`) along to the decoder, not the
249final state (as returned by the coder).  If you want to use
250:func:`struct.struct` to store the state in binary you can code the first
251element (the predicted value) in 16 bits and the second (the delta index) in 8.
252
253The ADPCM coders have never been tried against other ADPCM coders, only against
254themselves.  It could well be that I misinterpreted the standards in which case
255they will not be interoperable with the respective standards.
256
257The :func:`find\*` routines might look a bit funny at first sight. They are
258primarily meant to do echo cancellation.  A reasonably fast way to do this is to
259pick the most energetic piece of the output sample, locate that in the input
260sample 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