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.

    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 

   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(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 

   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 :func:`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