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pyalsaaudio/doc/libalsaaudio.rst
Oswald Buddenhagen 196ca87a05 assorted improvements (#123) 
* fix draining/closing, take 2

commit 8abf06be introduced a pause() prior to draining, in an attempt
to work around clearly broken pulseaudio client behavior for capture
streams (drain() is supposed to imply a stop).

but as the workaround was also applied to playback streams, it would
cause nasty "clicks", as the stream would (obviously) stop before being
resumed for draining.

but draining is actually pointless for capture streams, as we're closing
right afterwards, so the samples are lost anyway.

what's more, destructors are not supposed to wait for anything, so
draining in alsapcm_dealloc() was wrong to start with. so we remove it.
note that this is a minor behavior change, which is reflected by the
adjustment of the playback test to have an explicit close() at the end.

finally, close() was also affected by the pulseaudio bug (which was not
addressed before), so there we make draining exclusive to playback
streams.

* fix memory leaks in *_polldescriptors()

the calloc'd pollfd arrays were not freed.

* fix memory handling in mixer access error paths

in case of error, alsamixer_new() would leak the object, while
alsamixer_list() might crash due to a null pointer.

as a drive-by, make alsamixer_gethandle() `static`.

* fix crashes when accessing already closed devices

PCM.htimestamp() gets the usual exception emission,
Mixer.close() gets a "double invocation" check like PCM.close() has.

* fix deprecation warning about PyEval_InitThreads()

PyEval_InitThreads is a no-op in since python 3.9.

* fix deprecation warning about PyUnicode_AsUnicode()

converting to ascii for the purpose of comparison is inefficient.

* remove redundant snd_pcm_hw_params_any() call

we just called it (and even error-checked it) a few lines above.

* add new high-speed samples rates

closes #89 (but alsa doesn't support 768khz yet).

* drop some pointless comments from the tex => sphinx conversion

amends 5c2a00655.

* remove bogus markup from the documentation

the poll objects are linked properly in a different way, and the
footnote appears outdated.

* unify line spacing in .rst files

one empty line, except for high-level sections, which get two.

while at it, trim whitespace on otherwise empty lines.

* formatting/language fixes in introduction document

* improve terminology document

mention xruns, and rework the definition of periods: concentrate on
relevant information, and remove the misinformation about period size
reduction being not that bad (pedantically, an application could run
somewhat asynchronously to the interrupts by using some timer, and
therefore actually save some of the overhead, but why would one use a
small period size in the first place then?).

also, language and formatting fixes.

* add missing and update incorrect/outdated documentation

for clarity, this includes docs which were previously omitted
(presumably) intentionally, but mark them as comments.

the getrec() and getmute() functions' docs are moved around, so they
appear in pairs with their set*() counterparts, like the *volume() ones
already did.

notably, this also fixes the docu of PCM_FORMAT_U8, which closes #104.

* add some best practices to the docu

addresses #110, among other things.

* purge pydoc from the source

it's been obsolete for a *long* time, and having it redundantly to the
rst sources is bad hygiene. it still contained some useful info, which
has been transplanted to the rst source in the previous commit.

* use data types closer to those of ALSA

this removes lots of casts around snd_pcm_hw_params_get_*() calls

we could go further with that to make the code clean if we enabled all
the warnings, but it doesn't seem worth the effort.

* reduce scope of GIL releases

it's pointless to enclose snd_pcm_close() and snd_pcm_pause(), as these
calls don't sleep.

* reshuffle XRUN recovery somewhat

perform it prior to invoking read()/write() if necessary, not right
after a failure event. this makes things more uniform and predictable.

we don't use snd_pcm_recover() any more, as we used it only for the
EPIPE case anyway, which boils down to snd_pcm_prepare() exactly.
handling ESTRPIPE as well might be desirable, but that's a separate
consideration.

* bump (minor) version

we're about to add new features.

* make period count configurable

the period count is just as important for playback latency as the period
size, so it makes no sense to have only one of them configurable.

as a drive-by, fix up the handling of periods in info() & dumpinfo().

* add PCM.drain()

for playback, this allows making sure that all written frames are
played, without using an external delay.

in principle, it's also usable for capture, but there isn't really a
practical reason to do so, as simply discarding excess captured frames
has no real cost.

* add PCM.state() and associated enum values

in principle, the state is already available from info(), but that's a
rather heavy function for something one might want to query often.

a practical use case might be checking whether a playback stream is done
draining, for example.
2023-04-15 21:45:32 +02:00

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****************
:mod:`alsaaudio`
****************
.. module:: alsaaudio
:platform: Linux
.. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
.. moduleauthor:: Lars Immisch <lars@ibp.de>
The :mod:`alsaaudio` module defines functions and classes for using ALSA.
.. function:: pcms(pcmtype=PCM_PLAYBACK)
List available PCM devices by name.
Arguments are:
* *pcmtype* - 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 ALSA cards by name. This function is only moderately
useful. If you want to see a list of available PCM devices, use :func:`pcms`
instead.
..
Omitted by intention due to being superseded by cards():
.. function:: card_indexes()
.. function:: card_name()
.. function:: mixers(cardindex=-1, device='default')
List the available mixers. The arguments are:
* *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.
**Note:** This should not be used, as it bypasses most of ALSA's configuration.
* *device* - the name of the device on which the mixer resides. The default
is ``'default'``.
**Note:** For a list of available controls, you can also use ``amixer`` on
the commandline::
$ amixer
To elaborate the example, calling :func:`mixers` with the argument
``cardindex=0`` should give the same list of Mixer controls as::
$ amixer -c 0
And calling :func:`mixers` with the argument ``device='foo'`` should give
the same list of Mixer controls as::
$ 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.
.. function:: asoundlib_version()
Return a Python string containing the ALSA version found.
.. _pcm-objects:
PCM Objects
-----------
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_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1)
This class is used to represent a PCM device (either for playback and
recording). The arguments are:
* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default).
* *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
(default).
* *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``.
* *channels* - the number of channels. The default value is 2 (stereo).
* *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures.
The default value is :const:`PCM_FORMAT_S16_LE`.
========================= ===============
Format Description
========================= ===============
``PCM_FORMAT_S8`` Signed 8 bit samples for each channel
``PCM_FORMAT_U8`` Unsigned 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 in 4 bytes)
``PCM_FORMAT_S24_BE`` Signed 24 bit samples for each channel (Big Endian byte order in 4 bytes)
``PCM_FORMAT_U24_LE`` Unsigned 24 bit samples for each channel (Little Endian byte order in 4 bytes)
``PCM_FORMAT_U24_BE`` Unsigned 24 bit samples for each channel (Big Endian byte order in 4 bytes)
``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
``PCM_FORMAT_S24_3LE`` Signed 24 bit samples for each channel (Little Endian byte order in 3 bytes)
``PCM_FORMAT_S24_3BE`` Signed 24 bit samples for each channel (Big Endian byte order in 3 bytes)
``PCM_FORMAT_U24_3LE`` Unsigned 24 bit samples for each channel (Little Endian byte order in 3 bytes)
``PCM_FORMAT_U24_3BE`` Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
========================= ===============
* *periodsize* - the period size in frames.
Make sure you understand :ref:`the meaning of periods <term-period>`.
The default value is 32, which is below the actual minimum of most devices,
and will therefore likely be larger in practice.
* *periods* - the number of periods in the buffer. The default value is 4.
* *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.
**Note:** This should not be used, as it bypasses most of ALSA's configuration.
This will construct a PCM object with the given settings.
*Changed in 0.10:*
- Added the optional named parameter `periods`.
*Changed in 0.9:*
- Added the optional named parameters `rate`, `channels`, `format` and `periodsize`.
*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.info()
The info function returns a dictionary containing the configuration of a PCM device. As ALSA takes into account limitations of the hardware and software devices the configuration achieved might not correspond to the values used during creation. There is therefore a need to check the realised configuration before processing the sound coming from the device or before sending sound to a device. A small subset of parameters can be set, but cannot be queried. These parameters are stored by alsaaudio and returned as they were given by the user, to distinguish them from parameters retrieved from ALSA these parameters have a name prefixed with **" (call value) "**. Yet another set of properties derives directly from the hardware and can be obtained through ALSA.
=========================== ============================= ==================================================================
Key Description (Reference) Type
=========================== ============================= ==================================================================
name PCM():device string
card_no *index of card* integer (negative indicates device not associable with a card)
device_no *index of PCM device* integer
subdevice_no *index of PCM subdevice* integer
state *name of PCM state* string
access_type *name of PCM access type* string
(call value) type PCM():type integer
(call value) type_name PCM():type string
(call value) mode PCM():mode integer
(call value) mode_name PCM():mode string
format PCM():format integer
format_name PCM():format string
format_description PCM():format string
subformat_name *name of PCM subformat* string
subformat_description *description of subformat* string
channels PCM():channels integer
rate PCM():rate integer (Hz)
period_time *period duration* integer (:math:`\mu s`)
period_size PCM():period_size integer (frames)
buffer_time *buffer time* integer (:math:`\mu s`) (negative indicates error)
buffer_size *buffer size* integer (frames) (negative indicates error)
get_periods *approx. periods in buffer* integer (negative indicates error)
rate_numden *numerator, denominator* tuple (integer (Hz), integer (Hz))
significant_bits *significant bits in sample* integer (negative indicates error)
is_batch *hw: double buffering* boolean (True: hardware supported)
is_block_transfer *hw: block transfer* boolean (True: hardware supported)
is_double *hw: double buffering* boolean (True: hardware supported)
is_half_duplex *hw: half-duplex* boolean (True: hardware supported)
is_joint_duplex *hw: joint-duplex* boolean (True: hardware supported)
can_overrange *hw: overrange detection* boolean (True: hardware supported)
can_mmap_sample_resolution *hw: sample-resol. mmap* boolean (True: hardware supported)
can_pause *hw: pause* boolean (True: hardware supported)
can_resume *hw: resume* boolean (True: hardware supported)
can_sync_start *hw: synchronized start* boolean (True: hardware supported)
=========================== ============================= ==================================================================
The italicized descriptions give a summary of the "full" description as it can be found in the `ALSA documentation <https://www.alsa-project.org/alsa-doc>`_. "hw:": indicates that the property indicated relates to the hardware. Parameters passed to the PCM object during instantation are prefixed with "PCM():", they are described there for the keyword argument indicated after "PCM():".
.. method:: PCM.pcmtype()
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 :const:`PCM_NONBLOCK`,
:const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
.. method:: PCM.cardname()
Return the name of the sound card used by this PCM object.
..
Omitted by intention due to not really fitting the c'tor-based setup concept:
.. method:: PCM.getchannels()
Returns list of the device's supported channel counts.
.. method:: PCM.getratebounds()
Returns the card's minimum and maximum supported sample rates as
a tuple of integers.
.. method:: PCM.getrates()
Returns the sample rates supported by the device.
The returned value can be of one of the following, depending on
the card's properties:
* Card supports only a single rate: returns the rate
* Card supports a continuous range of rates: returns a tuple of
the range's lower and upper bounds (inclusive)
* Card supports a collection of well-known rates: returns a list of
the supported rates
.. method:: PCM.getformats()
Returns a dictionary of supported format codes (integers) keyed by
their standard ALSA names (strings).
.. method:: PCM.setchannels(nchannels)
.. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
.. method:: PCM.setrate(rate)
.. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
.. method:: PCM.setformat(format)
.. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
.. method:: PCM.setperiodsize(period)
.. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
.. method:: PCM.info()
Returns a dictionary with the PCM object's configured parameters.
Values are retrieved from the ALSA library if they are available;
otherwise they represent those stored by pyalsaaudio, and their keys
are prefixed with ' (call value) '.
*New in 0.9.1*
.. method:: PCM.dumpinfo()
Dumps the PCM object's configured parameters to stdout.
.. method:: PCM.state()
Returs the current state of the stream, which can be one of
:const:`PCM_STATE_OPEN` (this should not actually happen),
:const:`PCM_STATE_SETUP` (after :func:`drop` or :func:`drain`),
:const:`PCM_STATE_PREPARED` (after construction),
:const:`PCM_STATE_RUNNING`,
:const:`PCM_STATE_XRUN`,
:const:`PCM_STATE_DRAINING`,
:const:`PCM_STATE_PAUSED`,
:const:`PCM_STATE_SUSPENDED`, and
:const:`PCM_STATE_DISCONNECTED`.
*New in 0.10*
.. method:: PCM.read()
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 :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.
In case of an overrun, this function will return a negative size: :const:`-EPIPE`.
This indicates that data was lost, even if the operation itself succeeded.
Try using a larger periodsize.
.. method:: PCM.write(data)
Writes (plays) the sound in data. The length of data *must* be a
multiple of the frame size, and *should* be exactly the size of a
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 :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 :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.
Note that this call completing means only that the samples were buffered
in the kernel, and playout will continue afterwards. Make sure that the
stream is drained before discarding the PCM handle.
.. method:: PCM.pause([enable=True])
If *enable* is :const:`True`, playback or capture is paused.
Otherwise, playback/capture is resumed.
.. method:: PCM.drop()
Stop the stream and drop residual buffered frames.
*New in 0.9*
.. method:: PCM.drain()
For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
and then stop the stream. In :const:`PCM_NORMAL` mode,
this function blocks until all pending playback is drained.
For :const:`PCM_CAPTURE` PCM objects, this function is not very useful.
*New in 0.10*
.. method:: PCM.close()
Closes the PCM device.
For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
this function blocks until all pending playback is drained.
.. method:: PCM.polldescriptors()
Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the PCM with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module.
.. method:: PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE])
Set the ALSA timestamp mode on the device. The mode argument can be set to
either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
.. method:: PCM.get_tstamp_mode()
Return the integer value corresponding to the ALSA timestamp mode. The
return value can be either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
.. method:: PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY])
Set the ALSA timestamp mode on the device. The type argument
can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
.. method:: PCM.get_tstamp_type()
Return the integer value corresponding to the ALSA timestamp type. The
return value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
.. method:: PCM.htimestamp()
Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*.
The type of output is controlled by the tstamp_type, as described in the table below.
================================= ===========================================
Timestamp Type Description
================================= ===========================================
``PCM_TSTAMP_TYPE_GETTIMEOFDAY`` System-wide realtime clock with seconds
since epoch.
``PCM_TSTAMP_TYPE_MONOTONIC`` Monotonic time from an unspecified starting
time. Progress is NTP synchronized.
``PCM_TSTAMP_TYPE_MONOTONIC_RAW`` Monotonic time from an unspecified starting
time using only the system clock.
================================= ===========================================
The timestamp mode is controlled by the tstamp_mode, as described in the table below.
================================= ===========================================
Timestamp Mode Description
================================= ===========================================
``PCM_TSTAMP_NONE`` No timestamp.
``PCM_TSTAMP_ENABLE`` Update timestamp at every hardware position
update.
================================= ===========================================
**A few hints on using PCM devices for playback**
The most common reason for problems with playback of PCM audio is that writes
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
(: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 :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.
In GUI programs, however, it may be a better strategy to setup the device,
preload the buffer with a few periods by calling write a couple of times, and
then use some timer method to write one period size of data to the device every
period. The purpose of the preloading is to avoid underrun clicks if the used
timer doesn't expire exactly on time.
Also note, that most timer APIs that you can find for Python will
accummulate time delays: If you set the timer to expire after 1/10'th
of a second, the actual timeout will happen slightly later, which will
accumulate to quite a lot after a few seconds. Hint: use time.time()
to check how much time has really passed, and add extra writes as nessecary.
.. _mixer-objects:
Mixer Objects
-------------
Mixer objects provides access to the ALSA mixer API.
.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default')
Arguments are:
* *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()
Return the name of the sound card used by this Mixer object
.. method:: Mixer.mixer()
Return the name of the specific mixer controlled by this object, For example
``'Master'`` or ``'PCM'``
.. method:: Mixer.mixerid()
Return the ID of the ALSA mixer controlled by this object.
.. method:: Mixer.switchcap()
Returns a list of the switches which are defined by this specific mixer.
Possible values in this list are:
====================== ================
Switch Description
====================== ================
'Mute' This mixer can mute
'Joined Mute' This mixer can mute all channels at the same time
'Playback Mute' This mixer can mute the playback output
'Joined Playback Mute' Mute playback for all channels at the same time}
'Capture Mute' Mute sound capture
'Joined Capture Mute' Mute sound capture for all channels at a time}
'Capture Exclusive' Not quite sure what this is
====================== ================
To manipulate these switches use the :meth:`setrec` or
:meth:`setmute` methods
.. method:: Mixer.volumecap()
Returns a list of the volume control capabilities of this
mixer. Possible values in the list are:
======================== ================
Capability Description
======================== ================
'Volume' This mixer can control volume
'Joined Volume' This mixer can control volume for all channels at the same time
'Playback Volume' This mixer can manipulate the playback output
'Joined Playback Volume' Manipulate playback volumne for all channels at the same time
'Capture Volume' Manipulate sound capture volume
'Joined Capture Volume' Manipulate sound capture volume for all channels at a time
======================== ================
.. method:: Mixer.getenum()
For enumerated controls, return the currently selected item and the list of
items available.
Returns a tuple *(string, list of strings)*.
For example, my soundcard has a Mixer called *Mono Output Select*. Using
*amixer*, I get::
$ amixer get "Mono Output Select"
Simple mixer control 'Mono Output Select',0
Capabilities: enum
Items: 'Mix' 'Mic'
Item0: 'Mix'
Using :mod:`alsaaudio`, one could do::
>>> import alsaaudio
>>> m = alsaaudio.Mixer('Mono Output Select')
>>> m.getenum()
('Mix', ['Mix', 'Mic'])
This method will return an empty tuple if the mixer is not an enumerated
control.
.. method:: Mixer.setenum(index)
For enumerated controls, sets the currently selected item.
*index* is an index into the list of available enumerated items returned
by :func:`getenum`.
.. method:: Mixer.getrange(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_RAW)
Return the volume range of the ALSA mixer controlled by this object.
The value is a tuple of integers whose meaning is determined by the
*units* argument.
The optional *pcmtype* 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`.
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.getvolume(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)
Returns a list with the current volume settings for each channel. The list
elements are integers whose meaning is determined by the *units* argument.
The optional *pcmtype* 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`.
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.setvolume(volume, channel=None, pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)
Change the current volume settings for this mixer. The *volume* argument
is an integer whose meaning is determined by the *units* argument.
If the optional argument *channel* is present, the volume is set
only for this channel. This assumes that the mixer can control the
volume for the channels independently.
The optional *pcmtype* 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`.
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.getmute()
Return a list indicating the current mute setting for each channel.
0 means not muted, 1 means muted.
This method will fail if the mixer has no playback switch capabilities.
.. method:: Mixer.setmute(mute, [channel])
Sets the mute flag to a new value. The *mute* argument is either 0 for not
muted, or 1 for muted.
The optional *channel* argument controls which channel is
muted. The default is to set the mute flag for all channels.
This method will fail if the mixer has no playback mute capabilities
.. method:: Mixer.getrec()
Return a list indicating the current record mute setting for each channel.
0 means not recording, 1 means recording.
This method will fail if the mixer has no capture switch capabilities.
.. 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.
The optional *channel* argument controls which channel is
changed. The default is to set the capture flag for all channels.
This method will fail if the mixer has no capture switch capabilities.
.. method:: Mixer.polldescriptors()
Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the mixer with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module.
.. method:: Mixer.handleevents()
Acknowledge events on the :func:`polldescriptors` file descriptors
to prevent subsequent polls from returning the same events again.
Returns the number of events that were acknowledged.
.. method:: Mixer.close()
Closes the Mixer device.
**A rant on the ALSA Mixer API**
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
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.
If my design choises prevents you from doing something that the underlying API
would have allowed, please let me know, so I can incorporate these needs into
future versions.
If the current state of affairs annoys you, the best you can do is to write a
HOWTO on the API and make this available on the net. Until somebody does this,
the availability of ALSA mixer capable devices will stay quite limited.
Unfortunately, I'm not able to create such a HOWTO myself, since I only
understand half of the API, and that which I do understand has come from a
painful trial and error process.
.. _pcm-example:
Examples
--------
The following example are provided:
* `playwav.py`
* `recordtest.py`
* `playbacktest.py`
* `mixertest.py`
All examples (except `mixertest.py`) accept the commandline option
*-c <cardname>*.
To determine a valid card name, use the commandline ALSA player::
$ aplay -L
or::
$ python
>>> import alsaaudio
>>> alsaaudio.pcms()
mixertest.py accepts the commandline options *-d <device>* and
*-c <cardindex>*.
playwav.py
~~~~~~~~~~
**playwav.py** plays a wav file.
To test PCM playback (on your default soundcard), run::
$ python playwav.py <wav file>
recordtest.py and playbacktest.py
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
**recordtest.py** and **playbacktest.py** will record and play a raw
sound file in CD quality.
To test PCM recordings (on your default soundcard), run::
$ python recordtest.py <filename>
Speak into the microphone, and interrupt the recording at any time
with ``Ctl-C``.
Play back the recording with::
$ python playbacktest.py <filename>
mixertest.py
~~~~~~~~~~~~
Without arguments, **mixertest.py** will list all available *controls* on the
default soundcard.
The output might look like this::
$ ./mixertest.py
Available mixer controls:
'Master'
'Master Mono'
'Headphone'
'PCM'
'Line'
'Line In->Rear Out'
'CD'
'Mic'
'PC Speaker'
'Aux'
'Mono Output Select'
'Capture'
'Mix'
'Mix Mono'
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%
With two arguments, the *control* and a *parameter*, it will set the
parameter on the mixer::
$ ./mixertest.py Master mute
This will mute the Master mixer.
Or::
$ ./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%
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