/Doc/library/audioop.rst
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- :mod:`audioop` --- Manipulate raw audio data
- ============================================
- .. module:: audioop
- :synopsis: Manipulate raw audio data.
- The :mod:`audioop` module contains some useful operations on sound fragments.
- It operates on sound fragments consisting of signed integer samples 8, 16 or 32
- bits wide, stored in Python strings. This is the same format as used by the
- :mod:`al` and :mod:`sunaudiodev` modules. All scalar items are integers, unless
- specified otherwise.
- .. index::
- single: Intel/DVI ADPCM
- single: ADPCM, Intel/DVI
- single: a-LAW
- single: u-LAW
- This module provides support for a-LAW, u-LAW and Intel/DVI ADPCM encodings.
- .. This para is mostly here to provide an excuse for the index entries...
- 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:
- .. exception:: error
- This exception is raised on all errors, such as unknown number of bytes per
- sample, etc.
- .. function:: add(fragment1, fragment2, width)
- Return a fragment which is the addition of the two samples passed as parameters.
- *width* is the sample width in bytes, either ``1``, ``2`` or ``4``. Both
- fragments should have the same length.
- .. function:: adpcm2lin(adpcmfragment, width, state)
- Decode an Intel/DVI ADPCM coded fragment to a linear fragment. See the
- description of :func:`lin2adpcm` for details on ADPCM coding. Return a tuple
- ``(sample, newstate)`` where the sample has the width specified in *width*.
- .. function:: alaw2lin(fragment, width)
- Convert sound fragments in a-LAW encoding to linearly encoded sound fragments.
- a-LAW encoding always uses 8 bits samples, so *width* refers only to the sample
- width of the output fragment here.
- .. versionadded:: 2.5
- .. function:: avg(fragment, width)
- Return the average over all samples in the fragment.
- .. function:: avgpp(fragment, width)
- Return the average peak-peak value over all samples in the fragment. No
- filtering is done, so the usefulness of this routine is questionable.
- .. function:: bias(fragment, width, bias)
- Return a fragment that is the original fragment with a bias added to each
- sample.
- .. function:: cross(fragment, width)
- Return the number of zero crossings in the fragment passed as an argument.
- .. function:: findfactor(fragment, reference)
- Return a factor *F* such that ``rms(add(fragment, mul(reference, -F)))`` is
- minimal, i.e., return the factor with which you should multiply *reference* to
- make it match as well as possible to *fragment*. The fragments should both
- contain 2-byte samples.
- The time taken by this routine is proportional to ``len(fragment)``.
- .. function:: findfit(fragment, reference)
- Try to match *reference* as well as possible to a portion of *fragment* (which
- should be the longer fragment). This is (conceptually) done by taking slices
- out of *fragment*, using :func:`findfactor` to compute the best match, and
- minimizing the result. The fragments should both contain 2-byte samples.
- Return a tuple ``(offset, factor)`` where *offset* is the (integer) offset into
- *fragment* where the optimal match started and *factor* is the (floating-point)
- factor as per :func:`findfactor`.
- .. function:: findmax(fragment, length)
- Search *fragment* for a slice of length *length* samples (not bytes!) with
- maximum energy, i.e., return *i* for which ``rms(fragment[i*2:(i+length)*2])``
- is maximal. The fragments should both contain 2-byte samples.
- The routine takes time proportional to ``len(fragment)``.
- .. function:: getsample(fragment, width, index)
- Return the value of sample *index* from the fragment.
- .. function:: lin2adpcm(fragment, width, state)
- Convert 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.
- *state* is a tuple containing the state of the coder. The coder returns a tuple
- ``(adpcmfrag, newstate)``, and the *newstate* should be passed to the next call
- of :func:`lin2adpcm`. In the initial call, ``None`` can be passed as the state.
- *adpcmfrag* is the ADPCM coded fragment packed 2 4-bit values per byte.
- .. function:: lin2alaw(fragment, width)
- Convert samples in the audio fragment to a-LAW encoding and return this as a
- Python string. a-LAW is an audio encoding format whereby you get a dynamic
- range of about 13 bits using only 8 bit samples. It is used by the Sun audio
- hardware, among others.
- .. versionadded:: 2.5
- .. function:: lin2lin(fragment, width, newwidth)
- Convert samples between 1-, 2- and 4-byte formats.
- .. note::
- In some audio formats, such as .WAV files, 16 and 32 bit samples are
- signed, but 8 bit samples are unsigned. So when converting to 8 bit wide
- samples for these formats, you need to also add 128 to the result::
- new_frames = audioop.lin2lin(frames, old_width, 1)
- new_frames = audioop.bias(new_frames, 1, 128)
- The same, in reverse, has to be applied when converting from 8 to 16 or 32
- bit width samples.
- .. function:: lin2ulaw(fragment, width)
- Convert samples in the audio fragment to u-LAW encoding and return 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.
- .. function:: minmax(fragment, width)
- Return a tuple consisting of the minimum and maximum values of all samples in
- the sound fragment.
- .. function:: max(fragment, width)
- Return the maximum of the *absolute value* of all samples in a fragment.
- .. function:: maxpp(fragment, width)
- Return the maximum peak-peak value in the sound fragment.
- .. function:: mul(fragment, width, factor)
- Return a fragment that has all samples in the original fragment multiplied by
- the floating-point value *factor*. Overflow is silently ignored.
- .. function:: ratecv(fragment, width, nchannels, inrate, outrate, state[, weightA[, weightB]])
- Convert the frame rate of the input fragment.
- *state* is a tuple containing the state of the converter. The converter returns
- a tuple ``(newfragment, newstate)``, and *newstate* should be passed to the next
- call of :func:`ratecv`. The initial call should pass ``None`` as the state.
- The *weightA* and *weightB* arguments are parameters for a simple digital filter
- and default to ``1`` and ``0`` respectively.
- .. function:: reverse(fragment, width)
- Reverse the samples in a fragment and returns the modified fragment.
- .. function:: rms(fragment, width)
- Return the root-mean-square of the fragment, i.e. ``sqrt(sum(S_i^2)/n)``.
- This is a measure of the power in an audio signal.
- .. function:: tomono(fragment, width, lfactor, rfactor)
- Convert a stereo fragment to a mono fragment. The left channel is multiplied by
- *lfactor* and the right channel by *rfactor* before adding the two channels to
- give a mono signal.
- .. function:: tostereo(fragment, width, lfactor, rfactor)
- Generate 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 *lfactor* and right channel samples by *rfactor*.
- .. function:: ulaw2lin(fragment, width)
- Convert sound fragments in u-LAW encoding to linearly encoded sound fragments.
- u-LAW encoding always uses 8 bits samples, so *width* refers only to the sample
- width of the output fragment here.
- Note that operations such as :func:`mul` or :func:`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::
- 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)
- 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 *initial*
- state (the one you passed to :func:`lin2adpcm`) along to the decoder, not the
- final state (as returned by the coder). If you want to use
- :func:`struct.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 :func:`find\*` routines might look a bit funny at first sight. They are
- primarily meant to do 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::
- 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)