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_sources/libalsaaudio.txt
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@@ -11,8 +11,7 @@
.. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
.. moduleauthor:: Lars Immisch <lars@ibp.de>
.. % Author of the module code;
@@ -34,37 +33,71 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
.. % should be enclosed in \var{...}.
.. function:: pcms([type=PCM_PLAYBACK])
List available PCM objects by name.
Items from this list can be used as the `device` keyword argument for the
:class:`PCM` constructor).
Arguments are:
* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default).
**Note:**
For :const:`PCM_PLAYBACK`, the list of device names should be equivalent
to the list of device names that ``aplay -L`` displays on the commandline::
$ aplay -L
For :const:`PCM_CAPTURE`, the list of device names should be equivalent
to the list of device names that ``arecord -L`` displays on the
commandline::
$ arecord -L
*New in 0.8*
.. function:: cards()
List the available cards by name (suitable for PCM objects).
List the available cards by name.
.. function:: mixers([cardindex])
.. function:: mixers(cardindex=-1, device='default')
List the available mixers. The optional *cardindex* specifies which card
should be queried. The default is 0.
List the available mixers. The arguments are:
.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, card='default')
* *cardindex* - the card index. If this argument is given, the device name
is constructed as: 'hw:*cardindex*' and
the `device` keyword argument is ignored. ``0`` is the first hardware sound
card.
This class is used to represent a PCM device (both for playback and
recording - capture). The arguments are:
* *device* - the name of the device on which the mixer resides. The default
is ``'default'``.
* *type* - can be either ``PCM_CAPTURE`` or ``PCM_PLAYBACK`` (default).
* *mode* - can be either ``PCM_NONBLOCK``, or ``PCM_NORMAL`` (default).
* *card* - specifies the name of the card that should be used.
**Note:** For a list of available controls, you can also use ``amixer`` on
the commandline::
$ amixer
.. class:: Mixer(control='Master', id=0, cardindex=0)
To elaborate the example, calling :func:`mixers` with the argument
``cardindex=0`` should give the same list of Mixer controls as::
This class is used to access a specific ALSA mixer. The arguments
are:
$ amixer -c 0
* *control* - Name of the chosen mixed (default is 'Master').
* *id* - id of mixer -- More explanation needed here
* *cardindex* specifies which card should be used.
And calling :func:`mixers` with the argument ``device='foo'`` should give
the same list of Mixer controls as::
.. exception:: ALSAAudioError
$ amixer -D foo
*Changed in 0.8*:
- The keyword argument `device` is new and can be used to
select virtual devices. As a result, the default behaviour has subtly
changed. Since 0.8, this functions returns the mixers for the default
device, not the mixers for the first card.
Exception raised when an operation fails for a ALSA specific reason. The
exception argument is a string describing the reason of the failure.
.. _pcm-objects:
@@ -75,39 +108,54 @@ PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
sound through speakers or a microphone. The PCM constructor takes the
following arguments:
.. class:: PCM(type=PCM_CAPTURE, mode=PCM_NORMAL, card='default')
.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)
*type* - can be either ``PCM_CAPTURE`` or ``PCM_PLAYBACK`` (default).
This class is used to represent a PCM device (either for playback and
recording). The arguments are:
*mode* - can be either ``PCM_NONBLOCK``, or ``PCM_NORMAL`` (the
default). In ``PCM_NONBLOCK`` mode, calls to :func:`read` will return
immediately independent of whether there is any actual data to
read. Similarly, calls to :func:`write` will return immediately without
actually writing anything to the playout buffer if the buffer is
full [#f1]_.
*card* - specifies which card should be used. This can be a string
like 'default' or a name that was returned from the :func:`cards` function.
* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default).
* *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
(default).
* *device* - the name of the PCM device that should be used (for example
a value from the output of :func:`pcms`). The default value is
``'default'``.
* *cardindex* - the card index. If this argument is given, the device name
is constructed as 'hw:*cardindex*' and
the `device` keyword argument is ignored.
``0`` is the first hardware sound card.
This will construct a PCM object with these default settings:
* Sample format: ``PCM_FORMAT_S16_LE``
* Sample format: :const:`PCM_FORMAT_S16_LE`
* Rate: 44100 Hz
* Channels: 2
* Period size: 32 frames
*Changed in 0.8:*
- The `card` keyword argument is still supported,
but deprecated. Please use `device` instead.
- The keyword argument `cardindex` was added.
The `card` keyword is deprecated because it guesses the real ALSA
name of the card. This was always fragile and broke some legitimate usecases.
PCM objects have the following methods:
.. method:: PCM.pcmtype()
Returns the type of PCM object. Either ``PCM_CAPTURE`` or ``PCM_PLAYBACK``.
Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
:const:`PCM_PLAYBACK`.
.. method:: PCM.pcmmode()
Return the mode of the PCM object. One of ``PCM_NONBLOCK``, ``PCM_ASYNC``,
or ``PCM_NORMAL``
Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
:const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
.. method:: PCM.cardname()
@@ -118,14 +166,15 @@ PCM objects have the following methods:
.. method:: PCM.setchannels(nchannels)
Used to set the number of capture or playback channels. Common
values are: 1 = mono, 2 = stereo, and 6 = full 6 channel audio. Few
sound cards support more than 2 channels
values are: ``1`` = mono, ``2`` = stereo, and ``6`` = full 6 channel audio.
Few sound cards support more than 2 channels
.. method:: PCM.setrate(rate)
Set the sample rate in Hz for the device. Typical values are 8000
(mainly used for telephony), 16000, 44100 (CD quality), and 96000.
Set the sample rate in Hz for the device. Typical values are ``8000``
(mainly used for telephony), ``16000``, ``44100`` (CD quality),
``48000`` and ``96000``.
.. method:: PCM.setformat(format)
@@ -135,52 +184,52 @@ PCM objects have the following methods:
The following formats are provided by ALSA:
===================== ===============
Format Description
===================== ===============
PCM_FORMAT_S8 Signed 8 bit samples for each channel
PCM_FORMAT_U8 Signed 8 bit samples for each channel
PCM_FORMAT_S16_LE Signed 16 bit samples for each channel Little Endian byte order)
PCM_FORMAT_S16_BE Signed 16 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U16_LE Unsigned 16 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U16_BE Unsigned 16 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_S24_LE Signed 24 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S24_BE Signed 24 bit samples for each channel (Big Endian byte order)}
PCM_FORMAT_U24_LE Unsigned 24 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U24_BE Unsigned 24 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_S32_LE Signed 32 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_S32_BE Signed 32 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_U32_LE Unsigned 32 bit samples for each channel (Little Endian byte order)
PCM_FORMAT_U32_BE Unsigned 32 bit samples for each channel (Big Endian byte order)
PCM_FORMAT_FLOAT_LE 32 bit samples encoded as float (Little Endian byte order)
PCM_FORMAT_FLOAT_BE 32 bit samples encoded as float (Big Endian byte order)
PCM_FORMAT_FLOAT64_LE 64 bit samples encoded as float (Little Endian byte order)
PCM_FORMAT_FLOAT64_BE 64 bit samples encoded as float (Big Endian byte order)
PCM_FORMAT_MU_LAW A logarithmic encoding (used by Sun .au files and telephony)
PCM_FORMAT_A_LAW Another logarithmic encoding
PCM_FORMAT_IMA_ADPCM A 4:1 compressed format defined by the Interactive Multimedia Association.
PCM_FORMAT_MPEG MPEG encoded audio?
PCM_FORMAT_GSM 9600 bits/s constant rate encoding for speech
===================== ===============
========================= ===============
Format Description
========================= ===============
``PCM_FORMAT_S8`` Signed 8 bit samples for each channel
``PCM_FORMAT_U8`` Signed 8 bit samples for each channel
``PCM_FORMAT_S16_LE`` Signed 16 bit samples for each channel Little Endian byte order)
``PCM_FORMAT_S16_BE`` Signed 16 bit samples for each channel (Big Endian byte order)
``PCM_FORMAT_U16_LE`` Unsigned 16 bit samples for each channel (Little Endian byte order)
``PCM_FORMAT_U16_BE`` Unsigned 16 bit samples for each channel (Big Endian byte order)
``PCM_FORMAT_S24_LE`` Signed 24 bit samples for each channel (Little Endian byte order)
``PCM_FORMAT_S24_BE`` Signed 24 bit samples for each channel (Big Endian byte order)}
``PCM_FORMAT_U24_LE`` Unsigned 24 bit samples for each channel (Little Endian byte order)
``PCM_FORMAT_U24_BE`` Unsigned 24 bit samples for each channel (Big Endian byte order)
``PCM_FORMAT_S32_LE`` Signed 32 bit samples for each channel (Little Endian byte order)
``PCM_FORMAT_S32_BE`` Signed 32 bit samples for each channel (Big Endian byte order)
``PCM_FORMAT_U32_LE`` Unsigned 32 bit samples for each channel (Little Endian byte order)
``PCM_FORMAT_U32_BE`` Unsigned 32 bit samples for each channel (Big Endian byte order)
``PCM_FORMAT_FLOAT_LE`` 32 bit samples encoded as float (Little Endian byte order)
``PCM_FORMAT_FLOAT_BE`` 32 bit samples encoded as float (Big Endian byte order)
``PCM_FORMAT_FLOAT64_LE`` 64 bit samples encoded as float (Little Endian byte order)
``PCM_FORMAT_FLOAT64_BE`` 64 bit samples encoded as float (Big Endian byte order)
``PCM_FORMAT_MU_LAW`` A logarithmic encoding (used by Sun .au files and telephony)
``PCM_FORMAT_A_LAW`` Another logarithmic encoding
``PCM_FORMAT_IMA_ADPCM`` A 4:1 compressed format defined by the Interactive Multimedia Association.
``PCM_FORMAT_MPEG`` MPEG encoded audio?
``PCM_FORMAT_GSM`` 9600 bits/s constant rate encoding for speech
========================= ===============
.. method:: PCM.setperiodsize(period)
Sets the actual period size in frames. Each write should consist of
exactly this number of frames, and each read will return this
number of frames (unless the device is in ``PCM_NONBLOCK`` mode, in
number of frames (unless the device is in :const:`PCM_NONBLOCK` mode, in
which case it may return nothing at all)
.. method:: PCM.read()
In ``PCM_NORMAL`` mode, this function blocks until a full period is
In :const:`PCM_NORMAL` mode, this function blocks until a full period is
available, and then returns a tuple (length,data) where *length* is
the number of frames of captured data, and *data* is the captured
sound frames as a string. The length of the returned data will be
periodsize\*framesize bytes.
In ``PCM_NONBLOCK`` mode, the call will not block, but will return
In :const:`PCM_NONBLOCK` mode, the call will not block, but will return
``(0,'')`` if no new period has become available since the last
call to read.
@@ -192,20 +241,20 @@ PCM objects have the following methods:
period. If less than 'period size' frames are provided, the actual
playout will not happen until more data is written.
If the device is not in ``PCM_NONBLOCK`` mode, this call will block if
If the device is not in :const:`PCM_NONBLOCK` mode, this call will block if
the kernel buffer is full, and until enough sound has been played
to allow the sound data to be buffered. The call always returns the
size of the data provided.
In ``PCM_NONBLOCK`` mode, the call will return immediately, with a
In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a
return value of zero, if the buffer is full. In this case, the data
should be written at a later time.
.. method:: PCM.pause([enable=1])
.. method:: PCM.pause([enable=True])
If *enable* is 1, playback or capture is paused. If *enable* is 0,
playback/capture is resumed.
If *enable* is :const:`True`, playback or capture is paused.
Otherwise, playback/capture is resumed.
**A few hints on using PCM devices for playback**
@@ -215,10 +264,10 @@ to PCM devices must *exactly* match the data rate of the device.
If too little data is written to the device, it will underrun, and
ugly clicking sounds will occur. Conversely, of too much data is
written to the device, the write function will either block
(``PCM_NORMAL`` mode) or return zero (``PCM_NONBLOCK`` mode).
(:const:`PCM_NORMAL` mode) or return zero (:const:`PCM_NONBLOCK` mode).
If your program does nothing but play sound, the best strategy is to put the
device in ``PCM_NORMAL`` mode, and just write as much data to the device as
device in :const:`PCM_NORMAL` mode, and just write as much data to the device as
possible. This strategy can also be achieved by using a separate
thread with the sole task of playing out sound.
@@ -243,23 +292,31 @@ Mixer Objects
Mixer objects provides access to the ALSA mixer API.
.. class:: Mixer(control='Master', id=0, cardindex=0)
*control* - specifies which control to manipulate using this mixer
object. The list of available controls can be found with the
:mod:`alsaaudio`.\ :func:`mixers` function. The default value is
'Master' - other common controls include 'Master Mono', 'PCM', 'Line', etc.
*id* - the id of the mixer control. Default is 0
*cardindex* - specifies which card should be used [#f3]_. 0 is the
first sound card.
.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default')
Arguments are:
**Note:** For a list of available controls, you can also use **amixer**::
amixer
* *control* - specifies which control to manipulate using this mixer
object. The list of available controls can be found with the
:mod:`alsaaudio`.\ :func:`mixers` function. The default value is
``'Master'`` - other common controls may be ``'Master Mono'``, ``'PCM'``,
``'Line'``, etc.
* *id* - the id of the mixer control. Default is ``0``.
* *cardindex* - specifies which card should be used. If this argument
is given, the device name is constructed like this: 'hw:*cardindex*' and
the `device` keyword argument is ignored. ``0`` is the
first sound card.
* *device* - the name of the device on which the mixer resides. The default
value is ``'default'``.
*Changed in 0.8*:
- The keyword argument `device` is new and can be used to select virtual
devices.
Mixer objects have the following methods:
.. method:: Mixer.cardname()
@@ -270,7 +327,7 @@ Mixer objects have the following methods:
.. method:: Mixer.mixer()
Return the name of the specific mixer controlled by this object, For example
'Master' or 'PCM'
``'Master'`` or ``'PCM'``
.. method:: Mixer.mixerid()
@@ -354,10 +411,10 @@ Mixer objects have the following methods:
Return the volume range of the ALSA mixer controlled by this object.
The optional *direction* argument can be either 'playback' or
'capture', which is relevant if the mixer can control both playback
and capture volume. The default value is 'playback' if the mixer
has this capability, otherwise 'capture'
The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
:const:`PCM_CAPTURE`, which is relevant if the mixer can control both
playback and capture volume. The default value is :const:`PCM_PLAYBACK`
if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
.. method:: Mixer.getrec()
@@ -373,13 +430,13 @@ Mixer objects have the following methods:
Returns a list with the current volume settings for each channel. The list
elements are integer percentages.
The optional *direction* argument can be either 'playback' or
'capture', which is relevant if the mixer can control both playback
and capture volume. The default value is 'playback' if the mixer
has this capability, otherwise 'capture'
The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
:const:`PCM_CAPTURE`, which is relevant if the mixer can control both
playback and capture volume. The default value is :const:`PCM_PLAYBACK`
if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
.. method:: Mixer.setvolume(volume,[channel], [direction])
.. method:: Mixer.setvolume(volume, [channel], [direction])
Change the current volume settings for this mixer. The *volume* argument
controls the new volume setting as an integer percentage.
@@ -388,11 +445,10 @@ Mixer objects have the following methods:
only for this channel. This assumes that the mixer can control the
volume for the channels independently.
The optional *direction* argument can be either 'playback' or 'capture' is
relevant if the mixer has independent playback and capture volume
capabilities, and controls which of the volumes if changed. The
default is 'playback' if the mixer has this capability, otherwise 'capture'.
The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
:const:`PCM_CAPTURE`, which is relevant if the mixer can control both
playback and capture volume. The default value is :const:`PCM_PLAYBACK`
if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
.. method:: Mixer.setmute(mute, [channel])
@@ -405,7 +461,7 @@ Mixer objects have the following methods:
This method will fail if the mixer has no playback mute capabilities
.. method:: Mixer.setrec(capture,[channel])
.. method:: Mixer.setrec(capture, [channel])
Sets the capture mute flag to a new value. The *capture* argument
is either 0 for no capture, or 1 for capture.
@@ -425,7 +481,7 @@ Mixer objects have the following methods:
The ALSA mixer API is extremely complicated - and hardly documented at all.
:mod:`alsaaudio` implements a much simplified way to access this API. In
designing the API I've had to make some choices which may limit what can and
cannot be controlled through the API. However, If I had chosen to implement the
cannot be controlled through the API. However, if I had chosen to implement the
full API, I would have reexposed the horrible complexity/documentation ratio of
the underlying API. At least the :mod:`alsaaudio` API is easy to
understand and use.
@@ -469,10 +525,10 @@ or::
$ python
>>> import alsaaudio
>>> alsaaudio.cards()
>>> alsaaudio.pcms()
mixertest.py accepts the commandline option *-c <cardindex>*. Card
indices start at 0.
mixertest.py accepts the commandline options *-d <device>* and
*-c <cardindex>*.
playwav.py
~~~~~~~~~~
@@ -502,11 +558,13 @@ Play back the recording with::
mixertest.py
~~~~~~~~~~~~
Without arguments, **mixertest.py** will list all available *controls*.
Without arguments, **mixertest.py** will list all available *controls* on the
default soundcard.
The output might look like this::
$ ./mixertest.py
Available mixer controls:
$ ./mixertest.py
Available mixer controls:
'Master'
'Master Mono'
'Headphone'
@@ -525,27 +583,34 @@ The output might look like this::
With a single argument - the *control*, it will display the settings of
that control; for example::
$ ./mixertest.py Master
Mixer name: 'Master'
Capabilities: Playback Volume Playback Mute
Channel 0 volume: 61%
Channel 1 volume: 61%
$ ./mixertest.py Master
Mixer name: 'Master'
Capabilities: Playback Volume Playback Mute
Channel 0 volume: 61%
Channel 1 volume: 61%
With two arguments, the *control* and a *parameter*, it will set the
parameter on the mixer::
$ ./mixertest.py Master mute
$ ./mixertest.py Master mute
This will mute the Master mixer.
Or::
$ ./mixertest.py Master 40
$ ./mixertest.py Master 40
This sets the volume to 40% on all channels.
To select a different soundcard, use either the *device* or *cardindex*
argument::
$ ./mixertest.py -c 0 Master
Mixer name: 'Master'
Capabilities: Playback Volume Playback Mute
Channel 0 volume: 61%
Channel 1 volume: 61%
.. rubric:: Footnotes
.. [#f1] ALSA also allows ``PCM_ASYNC``, but this is not supported yet.
.. [#f2] :mod:`alsaaudio` will leave any name alone that has a ':' (colon) in it.
.. [#f3] This is inconsistent with the PCM objects, which use names, but it is consistent with aplay and amixer.
_sources/pyalsaaudio.txt
+4 -4
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@@ -2,10 +2,10 @@
Introduction
************
:Author: Casper Wilstrup
:Author: Lars Immisch
:Author: Casper Wilstrup <cwi@aves.dk>
:Author: Lars Immisch <lars@ibp.de>
.. |release| replace:: 0.4
.. |release| replace:: version
.. % At minimum, give your name and an email address. You can include a
.. % snail-mail address if you like.
@@ -116,7 +116,7 @@ First of all, run::
This is a small test suite that mostly performs consistency tests. If
it fails, please file a `bug report
<http://sourceforge.net/tracker/?group_id=120651>`_.
<https://github.com/larsimmisch/pyalsaaudio/issues>`_.
To test PCM recordings (on your default soundcard), verify your
microphone works, then do::