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865 lines
34 KiB
ReStructuredText
865 lines
34 KiB
ReStructuredText
****************
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:mod:`alsaaudio`
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****************
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.. module:: alsaaudio
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:platform: Linux
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.. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
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.. moduleauthor:: Lars Immisch <lars@ibp.de>
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The :mod:`alsaaudio` module defines functions and classes for using ALSA.
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.. function:: pcms(pcmtype: int = PCM_PLAYBACK) ->list[str]
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List available PCM devices by name.
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Arguments are:
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* *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
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(default).
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**Note:**
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For :const:`PCM_PLAYBACK`, the list of device names should be equivalent
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to the list of device names that ``aplay -L`` displays on the commandline::
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$ aplay -L
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For :const:`PCM_CAPTURE`, the list of device names should be equivalent
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to the list of device names that ``arecord -L`` displays on the
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commandline::
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$ arecord -L
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*New in 0.8*
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.. function:: cards() -> list[str]
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List the available ALSA cards by name. This function is only moderately
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useful. If you want to see a list of available PCM devices, use :func:`pcms`
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instead.
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..
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Omitted by intention due to being superseded by cards():
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.. function:: card_indexes()
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.. function:: card_name()
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.. function:: mixers(cardindex: int = -1, device: str = 'default') -> list[str]
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List the available mixers. The arguments are:
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* *cardindex* - the card index. If this argument is given, the device name
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is constructed as: 'hw:*cardindex*' and
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the `device` keyword argument is ignored. ``0`` is the first hardware sound
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card.
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**Note:** This should not be used, as it bypasses most of ALSA's configuration.
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* *device* - the name of the device on which the mixer resides. The default
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is ``'default'``.
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**Note:** For a list of available controls, you can also use ``amixer`` on
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the commandline::
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$ amixer
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To elaborate the example, calling :func:`mixers` with the argument
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``cardindex=0`` should give the same list of Mixer controls as::
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$ amixer -c 0
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And calling :func:`mixers` with the argument ``device='foo'`` should give
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the same list of Mixer controls as::
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$ amixer -D foo
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*Changed in 0.8*:
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- The keyword argument `device` is new and can be used to
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select virtual devices. As a result, the default behaviour has subtly
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changed. Since 0.8, this functions returns the mixers for the default
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device, not the mixers for the first card.
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.. function:: asoundlib_version() -> str
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Return a Python string containing the ALSA version found.
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.. _pcm-objects:
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PCM Objects
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-----------
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PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
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sound through speakers or a microphone. The PCM constructor takes the
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following arguments:
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.. class:: PCM(type: int = PCM_PLAYBACK, mode: int = PCM_NORMAL, rate: int = 44100, channels: int = 2,
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format: int = PCM_FORMAT_S16_LE, periodsize: int = 32, periods: int = 4,
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device: str = 'default', cardindex: int = -1) -> PCM
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This class is used to represent a PCM device (either for playback or
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recording). The constructor's arguments are:
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* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
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(default).
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* *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
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(default).
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* *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``.
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* *channels* - the number of channels. The default value is 2 (stereo).
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* *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures.
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The default value is :const:`PCM_FORMAT_S16_LE`.
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========================= ===============
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Format Description
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========================= ===============
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``PCM_FORMAT_S8`` Signed 8 bit samples for each channel
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``PCM_FORMAT_U8`` Unsigned 8 bit samples for each channel
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``PCM_FORMAT_S16_LE`` Signed 16 bit samples for each channel Little Endian byte order)
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``PCM_FORMAT_S16_BE`` Signed 16 bit samples for each channel (Big Endian byte order)
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``PCM_FORMAT_U16_LE`` Unsigned 16 bit samples for each channel (Little Endian byte order)
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``PCM_FORMAT_U16_BE`` Unsigned 16 bit samples for each channel (Big Endian byte order)
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``PCM_FORMAT_S24_LE`` Signed 24 bit samples for each channel (Little Endian byte order in 4 bytes)
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``PCM_FORMAT_S24_BE`` Signed 24 bit samples for each channel (Big Endian byte order in 4 bytes)
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``PCM_FORMAT_U24_LE`` Unsigned 24 bit samples for each channel (Little Endian byte order in 4 bytes)
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``PCM_FORMAT_U24_BE`` Unsigned 24 bit samples for each channel (Big Endian byte order in 4 bytes)
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``PCM_FORMAT_S32_LE`` Signed 32 bit samples for each channel (Little Endian byte order)
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``PCM_FORMAT_S32_BE`` Signed 32 bit samples for each channel (Big Endian byte order)
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``PCM_FORMAT_U32_LE`` Unsigned 32 bit samples for each channel (Little Endian byte order)
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``PCM_FORMAT_U32_BE`` Unsigned 32 bit samples for each channel (Big Endian byte order)
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``PCM_FORMAT_FLOAT_LE`` 32 bit samples encoded as float (Little Endian byte order)
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``PCM_FORMAT_FLOAT_BE`` 32 bit samples encoded as float (Big Endian byte order)
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``PCM_FORMAT_FLOAT64_LE`` 64 bit samples encoded as float (Little Endian byte order)
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``PCM_FORMAT_FLOAT64_BE`` 64 bit samples encoded as float (Big Endian byte order)
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``PCM_FORMAT_MU_LAW`` A logarithmic encoding (used by Sun .au files and telephony)
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``PCM_FORMAT_A_LAW`` Another logarithmic encoding
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``PCM_FORMAT_IMA_ADPCM`` A 4:1 compressed format defined by the Interactive Multimedia Association.
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``PCM_FORMAT_MPEG`` MPEG encoded audio?
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``PCM_FORMAT_GSM`` 9600 bits/s constant rate encoding for speech
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``PCM_FORMAT_S24_3LE`` Signed 24 bit samples for each channel (Little Endian byte order in 3 bytes)
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``PCM_FORMAT_S24_3BE`` Signed 24 bit samples for each channel (Big Endian byte order in 3 bytes)
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``PCM_FORMAT_U24_3LE`` Unsigned 24 bit samples for each channel (Little Endian byte order in 3 bytes)
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``PCM_FORMAT_U24_3BE`` Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
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========================= ===============
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* *periodsize* - the period size in frames.
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Make sure you understand :ref:`the meaning of periods <term-period>`.
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The default value is 32, which is below the actual minimum of most devices,
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and will therefore likely be larger in practice.
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* *periods* - the number of periods in the buffer. The default value is 4.
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* *device* - the name of the PCM device that should be used (for example
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a value from the output of :func:`pcms`). The default value is
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``'default'``.
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* *cardindex* - the card index. If this argument is given, the device name
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is constructed as 'hw:*cardindex*' and
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the `device` keyword argument is ignored.
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``0`` is the first hardware sound card.
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**Note:** This should not be used, as it bypasses most of ALSA's configuration.
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The defaults mentioned above are values passed by :mod:alsaaudio
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to ALSA, not anything internal to ALSA.
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**Note:** For default and non-default values alike, there is no
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guarantee that a PCM device supports the requested configuration,
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and ALSA may pick realizable values which it believes to be closest
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to the request. Therefore, after creating a PCM object, it is
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necessary to verify whether its realized configuration is acceptable.
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The :func:info method can be used to query it.
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*Changed in 0.10:*
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- Added the optional named parameter `periods`.
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*Changed in 0.9:*
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- Added the optional named parameters `rate`, `channels`, `format` and `periodsize`.
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*Changed in 0.8:*
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- The `card` keyword argument is still supported,
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but deprecated. Please use `device` instead.
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- The keyword argument `cardindex` was added.
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The `card` keyword is deprecated because it guesses the real ALSA
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name of the card. This was always fragile and broke some legitimate usecases.
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PCM objects have the following methods:
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.. method:: PCM.info() -> dict
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Returns a dictionary containing the configuration of a PCM device.
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A small subset of properties reflects fixed parameters given by the
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user, stored within alsaaudio. To distinguish them from properties
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retrieved from ALSA when the call is made, they have their name
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prefixed with **" (call value) "**.
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Descriptions of properties which can be directly set during PCM object
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instantiation carry the prefix "PCM():", followed by the respective
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constructor parameter. Note that due to device limitations, the values
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may deviate from those originally requested.
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Yet another set of properties cannot be set, and derives directly from
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the hardware, possibly depending on other properties. Those properties'
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descriptions are prefixed with "hw:" below.
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=========================== ==================================== ==================================================================
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Key Description (Reference) Type
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=========================== ==================================== ==================================================================
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name PCM():device string
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card_no *index of card* integer (negative indicates device not associable with a card)
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device_no *index of PCM device* integer
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subdevice_no *index of PCM subdevice* integer
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state *name of PCM state* string
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access_type *name of PCM access type* string
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(call value) type PCM():type integer
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(call value) type_name PCM():type string
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(call value) mode PCM():mode integer
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(call value) mode_name PCM():mode string
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format PCM():format integer
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format_name PCM():format string
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format_description PCM():format string
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subformat_name *name of PCM subformat* string
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subformat_description *description of subformat* string
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channels PCM():channels integer
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rate PCM():rate integer (Hz)
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period_time *period duration* integer (:math:`\mu s`)
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period_size PCM():period_size integer (frames)
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buffer_time *buffer time* integer (:math:`\mu s`) (negative indicates error)
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buffer_size *buffer size* integer (frames) (negative indicates error)
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get_periods *approx. periods in buffer* integer (negative indicates error)
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rate_numden *numerator, denominator* tuple (integer (Hz), integer (Hz))
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significant_bits *significant bits in sample* [#tss]_ integer (negative indicates error)
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nominal_bits *nominal bits in sample* [#tss]_ integer (negative indicates error)
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physical_bits *sample width in bits* [#tss]_ integer (negative indicates error)
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is_batch *hw: double buffering* boolean (True: hardware supported)
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is_block_transfer *hw: block transfer* boolean (True: hardware supported)
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is_double *hw: double buffering* boolean (True: hardware supported)
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is_half_duplex *hw: half-duplex* boolean (True: hardware supported)
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is_joint_duplex *hw: joint-duplex* boolean (True: hardware supported)
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can_overrange *hw: overrange detection* boolean (True: hardware supported)
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can_mmap_sample_resolution *hw: sample-resol. mmap* boolean (True: hardware supported)
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can_pause *hw: pause* boolean (True: hardware supported)
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can_resume *hw: resume* boolean (True: hardware supported)
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can_sync_start *hw: synchronized start* boolean (True: hardware supported)
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=========================== ==================================== ==================================================================
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.. [#tss] More information in the :ref:`terminology section for sample size <term-sample-size>`
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..
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The italicized descriptions give a summary of the "full" description
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as can be found in the
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`ALSA documentation <https://www.alsa-project.org/alsa-doc>`_.
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*New in 0.9.1*
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.. method:: PCM.dumpinfo()
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Dumps the PCM object's configured parameters to stdout.
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.. method:: PCM.pcmtype() -> int
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Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
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:const:`PCM_PLAYBACK`.
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.. method:: PCM.pcmmode() -> int
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Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
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:const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
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.. method:: PCM.cardname() -> string
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Return the name of the sound card used by this PCM object.
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..
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Omitted by intention due to not really fitting the c'tor-based setup concept:
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.. method:: PCM.getchannels()
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Returns list of the device's supported channel counts.
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.. method:: PCM.getratebounds()
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Returns the card's minimum and maximum supported sample rates as
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a tuple of integers.
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.. method:: PCM.getrates()
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Returns the sample rates supported by the device.
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The returned value can be of one of the following, depending on
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the card's properties:
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* Card supports only a single rate: returns the rate
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* Card supports a continuous range of rates: returns a tuple of
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the range's lower and upper bounds (inclusive)
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* Card supports a collection of well-known rates: returns a list of
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the supported rates
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.. method:: PCM.getformats()
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Returns a dictionary of supported format codes (integers) keyed by
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their standard ALSA names (strings).
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.. method:: PCM.setchannels(nchannels: int) -> int
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.. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
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.. method:: PCM.setrate(rate: int) -> int
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.. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
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.. method:: PCM.setformat(format: int) -> int
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.. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
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.. method:: PCM.setperiodsize(period: int) -> int
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.. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
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.. method:: PCM.state() -> int
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Returs the current state of the stream, which can be one of
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:const:`PCM_STATE_OPEN` (this should not actually happen),
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:const:`PCM_STATE_SETUP` (after :func:`drop` or :func:`drain`),
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:const:`PCM_STATE_PREPARED` (after construction),
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:const:`PCM_STATE_RUNNING`,
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:const:`PCM_STATE_XRUN`,
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:const:`PCM_STATE_DRAINING`,
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:const:`PCM_STATE_PAUSED`,
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:const:`PCM_STATE_SUSPENDED`, and
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:const:`PCM_STATE_DISCONNECTED`.
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*New in 0.10*
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.. method:: PCM.avail() -> int
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For :const:`PCM_PLAYBACK` PCM objects, returns the number of writable
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(that is, free) frames in the buffer.
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For :const:`PCM_CAPTURE` PCM objects, returns the number of readable
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(that is, filled) frames in the buffer.
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An attempt to read/write more frames than indicated will block (in
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:const:`PCM_NORMAL` mode) or fail and return zero (in
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:const:`PCM_NONBLOCK` mode).
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*New in 0.11*
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.. method:: PCM.read() -> tuple[int, bytes]
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In :const:`PCM_NORMAL` mode, this function blocks until a full period is
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available, and then returns a tuple (length,data) where *length* is
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the number of frames of captured data, and *data* is the captured
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sound frames as a string. The length of the returned data will be
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periodsize\*framesize bytes.
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In :const:`PCM_NONBLOCK` mode, the call will not block, but will return
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``(0,'')`` if no new period has become available since the last
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call to read.
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In case of a buffer overrun, this function will return the negative
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size :const:`-EPIPE`, and no data is read.
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This indicates that data was lost. To resume capturing, just call read
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again, but note that the stream was already corrupted.
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To avoid the problem in the future, try using a larger period size
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and/or more periods, at the cost of higher latency.
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.. method:: PCM.write(data: bytes) -> int
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Writes (plays) the sound in data. The length of data *must* be a
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multiple of the frame size, and *should* be exactly the size of a
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period. If less than 'period size' frames are provided, the actual
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playout will not happen until more data is written.
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If the data was successfully written, the call returns the size of the
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data *in frames*.
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If the device is not in :const:`PCM_NONBLOCK` mode, this call will block
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if the kernel buffer is full, and until enough sound has been played
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to allow the sound data to be buffered.
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In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a
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return value of zero, if the buffer is full. In this case, the data
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should be written again at a later time.
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In case of a buffer underrun, this function will return the negative
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size :const:`-EPIPE`, and no data is written.
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At this point, the playback was already corrupted. If you want to play
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the data nonetheless, call write again with the same data.
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To avoid the problem in the future, try using a larger period size
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and/or more periods, at the cost of higher latency.
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Note that this call completing means only that the samples were buffered
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in the kernel, and playout will continue afterwards. Make sure that the
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stream is drained before discarding the PCM handle.
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.. method:: PCM.pause([enable: int = True]) -> int
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If *enable* is :const:`True`, playback or capture is paused.
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Otherwise, playback/capture is resumed.
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.. method:: PCM.drop() -> int
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Stop the stream and drop residual buffered frames.
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*New in 0.9*
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.. method:: PCM.drain() -> int
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For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
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and then stop the stream. In :const:`PCM_NORMAL` mode,
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this function blocks until all pending playback is drained.
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For :const:`PCM_CAPTURE` PCM objects, this function is not very useful.
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*New in 0.10*
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.. method:: PCM.close() -> None
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Closes the PCM device.
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For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
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this function blocks until all pending playback is drained.
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.. method:: PCM.polldescriptors() -> list[tuple[int, int]]
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Returns a list of tuples of *(file descriptor, eventmask)* that can be
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used to wait for changes on the PCM with *select.poll*.
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The *eventmask* value is compatible with `poll.register`__ in the Python
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:const:`select` module.
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.. method:: PCM.polldescriptors_revents(descriptors: list[tuple[int, int]]) -> int
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Processes the descriptor list returned by :func:`polldescriptors` after
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using it with *select.poll*, and returns a single *eventmask* value that
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is meaningful for deciding whether :func:`read` or :func:`write` should
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be called.
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*New in 0.11*
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.. method:: PCM.set_tstamp_mode([mode: int = PCM_TSTAMP_ENABLE])
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Set the ALSA timestamp mode on the device. The mode argument can be set to
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either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
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.. method:: PCM.get_tstamp_mode() -> int
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Return the integer value corresponding to the ALSA timestamp mode. The
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return value can be either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
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.. method:: PCM.set_tstamp_type([type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY]) -> None
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Set the ALSA timestamp mode on the device. The type argument
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can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
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:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
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.. method:: PCM.get_tstamp_type() -> int
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Return the integer value corresponding to the ALSA timestamp type. The
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return value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
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:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
|
|
|
|
.. method:: PCM.htimestamp() -> tuple[int, int, int]
|
|
|
|
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, if 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: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> Mixer
|
|
|
|
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() -> str
|
|
|
|
Return the name of the sound card used by this Mixer object
|
|
|
|
.. method:: Mixer.mixer() -> str
|
|
|
|
Return the name of the specific mixer controlled by this object, For example
|
|
``'Master'`` or ``'PCM'``
|
|
|
|
.. method:: Mixer.mixerid() -> int
|
|
|
|
Return the ID of the ALSA mixer controlled by this object.
|
|
|
|
.. method:: Mixer.switchcap() -> int
|
|
|
|
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() -> int
|
|
|
|
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() -> tuple[str, list[str]]
|
|
|
|
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: int) -> None
|
|
|
|
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: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]
|
|
|
|
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: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int
|
|
|
|
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: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None
|
|
|
|
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() -> list[int]
|
|
|
|
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: bool, channel: (int | None) = None) -> None
|
|
|
|
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() -> list[int]
|
|
|
|
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: int, channel: (int | None) = None) -> None
|
|
|
|
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() -> list[tuple[int, int]]
|
|
|
|
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() -> int
|
|
|
|
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() -> None
|
|
|
|
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%
|