| :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 bytes objects. 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. Samples are truncated in case of overflow. |
| |
| |
| .. 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. |
| |
| |
| .. 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. Samples wrap around in case of overflow. |
| |
| |
| .. 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 |
| bytes object. 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. |
| |
| |
| .. 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 |
| bytes object. 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:: 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:: minmax(fragment, width) |
| |
| Return a tuple consisting of the minimum and maximum values of all samples 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*. Samples are truncated in case of overflow. |
| |
| |
| .. 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(lsample, width, lfactor) |
| rsample = audioop.mul(rsample, 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) |
| |