Experimenting with autoapi

getchannels/setchannels should be together IMO

CHANGES.md

@@ -1,3 +1,17 @@
+# Version 0.10.1
+- revert to not throwing an exception on playback buffer underrun;
+  instead, return -EPIPE like `PCM.read()` does on overrun; #131
+- type hints
+
+# Version 0.10.0
+- assorted improvements (#123 from @ossilator)
+  - support for `periods` in the `PCM` constructor.
+  - new functions `PCM.state()`, `PCM.drop()` and `PCM.drain()`
+  - improved underrun/overrun handling
+  - documentation improvements/consolidation (docstrings were removed in favour of online documentation)
+  - more sampling rates
+  - bug fixes
+
 # Version 0.9.2
 - Fix alsamixer_getvolume (#112 from @stephensp)
 
@@ -15,15 +29,15 @@
     supported formats - e.g. `{"U8": 1, "S16_LE": 2}`,
   - `getchannels()` returns a list of the supported channel numbers, e.g. `[1, 2]`,
   - `getrates()` returns supported sample rates for the device, e.g. `[48000]`,
-  - `getratebounds()` returns the device's official minimum and maximum supported 
+  - `getratebounds()` returns the device's official minimum and maximum supported
     sample rates as a tuple, e.g. `(4000, 48000)`.
-  
+
   (#82 contributed by @jdstmporter)
 
 - Prevent hang on close after capturing audio (#80 contributed by @daym)
 
 # Version 0.8.5:
-- Return an empty string/bytestring when `read()` detects an 
+- Return an empty string/bytestring when `read()` detects an
   overrun. Previously the returned data was undefined (contributed by @jcea)
 
 - Unlimited setperiod buffer size when reading frames (contributed by @jcea)

MANIFEST.in

@@ -1,5 +1,6 @@
 include *.py
+include *.pyi
 include CHANGES
 include TODO
 include LICENSE
-recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json  Makefile
+recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json  Makefile

README.md

@@ -2,7 +2,7 @@
 
 For documentation, see http://larsimmisch.github.io/pyalsaaudio/
 
-> Author: Casper Wilstrup (cwi@aves.dk)  
+> Author: Casper Wilstrup (cwi@aves.dk)
 > Maintainer: Lars Immisch (lars@ibp.de)
 
 This package contains wrappers for accessing the
@@ -45,12 +45,12 @@ First, get the sources and change to the source directory:
   $ git clone https://github.com/larsimmisch/pyalsaaudio.git
   $ cd pyalsaaudio
 ```
-  
+
 Then, build:
 ```
   $ python setup.py build
 ```
-  
+
 And install:
 ```
   $ sudo python setup.py install

alsaaudio.pyi

@@ -0,0 +1,135 @@
+from typing import list
+
+PCM_PLAYBACK: int
+PCM_CAPTURE: int
+
+PCM_NORMAL: int
+PCM_NONBLOCK: int
+PCM_ASYNC: int
+
+PCM_FORMAT_S8: int
+PCM_FORMAT_U8: int
+PCM_FORMAT_S16_LE: int
+PCM_FORMAT_S16_BE: int
+PCM_FORMAT_U16_LE: int
+PCM_FORMAT_U16_BE: int
+PCM_FORMAT_S24_LE: int
+PCM_FORMAT_S24_BE: int
+PCM_FORMAT_U24_LE: int
+PCM_FORMAT_U24_BE: int
+PCM_FORMAT_S32_LE: int
+PCM_FORMAT_S32_BE: int
+PCM_FORMAT_U32_LE: int
+PCM_FORMAT_U32_BE: int
+PCM_FORMAT_FLOAT_LE: int
+PCM_FORMAT_FLOAT_BE: int
+PCM_FORMAT_FLOAT64_LE: int
+PCM_FORMAT_FLOAT64_BE: int
+PCM_FORMAT_MU_LAW: int
+PCM_FORMAT_A_LAW: int
+PCM_FORMAT_IMA_ADPCM: int
+PCM_FORMAT_MPEG: int
+PCM_FORMAT_GSM: int
+PCM_FORMAT_S24_3LE: int
+PCM_FORMAT_S24_3BE: int
+PCM_FORMAT_U24_3LE: int
+PCM_FORMAT_U24_3BE: int
+
+PCM_TSTAMP_NONE: int
+PCM_TSTAMP_ENABLE: int
+
+PCM_TSTAMP_TYPE_GETTIMEOFDAY: int
+PCM_TSTAMP_TYPE_MONOTONIC: int
+PCM_TSTAMP_TYPE_MONOTONIC_RAW: int
+
+PCM_FORMAT_DSD_U8: int
+PCM_FORMAT_DSD_U16_LE: int
+PCM_FORMAT_DSD_U32_LE: int
+PCM_FORMAT_DSD_U32_BE: int
+
+PCM_STATE_OPEN: int
+PCM_STATE_SETUP: int
+PCM_STATE_PREPARED: int
+PCM_STATE_RUNNING: int
+PCM_STATE_XRUN: int
+PCM_STATE_DRAINING: int
+PCM_STATE_PAUSED: int
+PCM_STATE_SUSPENDED: int
+PCM_STATE_DISCONNECTED: int
+
+MIXER_CHANNEL_ALL: int
+
+MIXER_SCHN_UNKNOWN: int
+MIXER_SCHN_FRONT_LEFT: int
+MIXER_SCHN_FRONT_RIGHT: int
+MIXER_SCHN_REAR_LEFT: int
+MIXER_SCHN_REAR_RIGHT: int
+MIXER_SCHN_FRONT_CENTER: int
+MIXER_SCHN_WOOFER: int
+MIXER_SCHN_SIDE_LEFT: int
+MIXER_SCHN_SIDE_RIGHT: int
+MIXER_SCHN_REAR_CENTER: int
+MIXER_SCHN_MONO: int
+
+VOLUME_UNITS_PERCENTAGE: int
+VOLUME_UNITS_RAW: int
+VOLUME_UNITS_DB: int
+
+def pcms(pcmtype: int) -> list[str]: ...
+def cards() -> list[str]: ...
+def mixers(cardindex: int = -1, device: str = 'default') -> list[str]: ...
+def asoundlib_version() -> str: ...
+
+class PCM:
+	def __init__(type: int = PCM_PLAYBACK, mode: int = PCM_NORMAL, rate: int = 44100, channels: int = 2,
+		format: int = PCM_FORMAT_S16_LE, periodsize: int = 32, periods: int = 4,
+		device: str = 'default', cardindex: int = -1) -> PCM: ...
+	def close() -> None: ...
+	def dumpinfo() -> None: ...
+	def info() -> dict: ...
+	def state() -> int: ...
+	def htimestamp() -> tuple[int, int, int]: ...
+	def set_tstamp_mode(mode: int = PCM_TSTAMP_ENABLE) -> None: ...
+	def get_tstamp_mode() -> int: ...
+	def set_tstamp_type(type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY) -> None: ...
+	def get_tstamp_type() -> int: ...
+	def getformats() -> dict: ...
+	def getratebounds() -> tuple[int, int]: ...
+	def getrates() -> int | tuple[int, int] | list[int]: ...
+	def getchannels() -> list[int]: ...
+	def setchannels(nchannels: int) -> None: ...
+	def pcmtype() -> int: ...
+	def pcmmode() -> int: ...
+	def cardname() -> str: ...
+	def setrate(rate: int) -> None: ...
+	def setformat(format: int) -> int: ...
+	def setperiodsize(period: int) -> int: ...
+	def read() -> tuple[int, bytes]: ...
+	def write(data: bytes) -> int: ...
+	def avail() -> int: ...
+	def pause(enable: bool = True) -> int: ...
+	def drop() -> int: ...
+	def drain() -> int: ...
+	def polldescriptors() -> list[tuple[int, int]]: ...
+	def polldescriptors_revents(descriptors: list[tuple[int, int]]) -> int: ...
+
+class Mixer:
+	def __init__(control: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> Mixer: ...
+	def cardname() -> str: ...
+	def close() -> None: ...
+	def mixer() -> str: ...
+	def mixerid() -> int: ...
+	def switchcap() -> int: ...
+	def volumecap() -> int: ...
+	def getvolume(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int: ...
+	def getrange(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]: ...
+	def getenum() -> tuple[str, list[str]]: ...
+	def getmute() -> list[int]: ...
+	def getrec() -> list[int]: ...
+	def setvolume(volume: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None: ...
+	def setenum(index: int) -> None: ...
+	def setmute(mute: bool, channel: (int | None) = None) -> None: ...
+	def setrec(capture: int, channel: (int | None) = None) -> None: ...
+	def polldescriptors() -> list[tuple[int, int]]: ...
+	def handleevents() -> int: ...
+

doc/README.md

@@ -26,7 +26,7 @@ Don't forget to update the documentation.
 The documentation is published through the `gh-pages` branch.
 
 To publish the documentation, you need to clone the `gh-pages` branch of this repository into
-`doc/gh-pages`. In `doc`, do: 
+`doc/gh-pages`. In `doc`, do:
 
     git clone -b gh-pages git@github.com:larsimmisch/pyalsaaudio.git gh-pages
 

doc/conf.py

@@ -34,7 +34,9 @@ from setup import pyalsa_version
 # Add any Sphinx extension module names here, as strings. They can be
 # extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
 # ones.
-extensions = []
+extensions = ['autoapi.extension']
+
+autoapi_dirs = ['..']
 
 # Add any paths that contain templates here, relative to this directory.
 templates_path = ['_templates']
@@ -67,7 +69,7 @@ release = version
 #
 # This is also used if you do content translation via gettext catalogs.
 # Usually you set "language" from the command line for these cases.
-language = None
+language = 'en'
 
 # List of patterns, relative to source directory, that match files and
 # directories to ignore when looking for source files.

doc/index.rst

@@ -1,8 +1,3 @@
-.. alsaaudio documentation documentation master file, created by
-   sphinx-quickstart on Thu Mar 30 23:52:21 2017.
-   You can adapt this file completely to your liking, but it should at least
-   contain the root `toctree` directive.
-
 alsaaudio documentation
 ===================================================
 
@@ -18,15 +13,13 @@ Download
 ========
 
 * `Download from pypi <https://pypi.python.org/pypi/pyalsaaudio>`_
-  
-      
+
 Github
 ======
 
 * `Repository <https://github.com/larsimmisch/pyalsaaudio/>`_
 * `Bug tracker <https://github.com/larsimmisch/pyalsaaudio/issues>`_
 
- 
 Indices and tables
 ==================
 
@@ -34,5 +27,3 @@ Indices and tables
 * :ref:`modindex`
 * :ref:`search`
 
-    
-  

doc/libalsaaudio.rst

@@ -5,42 +5,19 @@
 .. module:: alsaaudio
    :platform: Linux
 
-
-.. % \declaremodule{builtin}{alsaaudio}	% standard library, in C
-.. % not standard, in C
-
 .. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
 .. moduleauthor:: Lars Immisch <lars@ibp.de>
-                  
-.. % Author of the module code;
-
-
 
 The :mod:`alsaaudio` module defines functions and classes for using ALSA.
 
-.. % ---- 3.1. ----
-.. % For each function, use a ``funcdesc'' block.  This has exactly two
-.. % parameters (each parameters is contained in a set of curly braces):
-.. % the first parameter is the function name (this automatically
-.. % generates an index entry); the second parameter is the function's
-.. % argument list.  If there are no arguments, use an empty pair of
-.. % curly braces.  If there is more than one argument, separate the
-.. % arguments with backslash-comma.  Optional parts of the parameter
-.. % list are contained in \optional{...} (this generates a set of square
-.. % brackets around its parameter).  Arguments are automatically set in
-.. % italics in the parameter list.  Each argument should be mentioned at
-.. % least once in the description; each usage (even inside \code{...})
-.. % should be enclosed in \var{...}.
-
-
-.. function:: pcms(pcmtype=PCM_PLAYBACK)
+.. function:: pcms(pcmtype: int = PCM_PLAYBACK) ->list[str]
 
    List available PCM devices by name.
-   
+
    Arguments are:
 
    * *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).  
+     (default).
 
    **Note:**
 
@@ -57,13 +34,13 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
 
    *New in 0.8*
 
-.. function:: cards()
+..
+   Omitted by intention due to being superseded by 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.
-   
-.. function:: mixers(cardindex=-1, device='default')
+   .. function:: card_indexes()
+   .. function:: card_name()
+
+.. function:: mixers(cardindex: int = -1, device: str = 'default') -> list[str]
 
    List the available mixers. The arguments are:
 
@@ -72,12 +49,14 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
      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
@@ -91,16 +70,17 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
       $ 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()
+.. function:: asoundlib_version() -> str
 
    Return a Python string containing the ALSA version found.
 
+
 .. _pcm-objects:
 
 PCM Objects
@@ -110,20 +90,22 @@ 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, device='default', cardindex=-1)
+.. class:: PCM(type: int = PCM_PLAYBACK, mode: int = PCM_NORMAL, rate: int = 44100, channels: int = 2,
+               format: int = PCM_FORMAT_S16_LE, periodsize: int = 32, periods: int = 4,
+               device: str = 'default', cardindex: int = -1) -> PCM
 
-   This class is used to represent a PCM device (either for playback and
-   recording). The arguments are:
+   This class is used to represent a PCM device (either for playback or
+   recording). The constructor's arguments are:
 
    * *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).  
+     (default).
    * *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
-     (default). 
+     (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. 
+   * *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
    =========================  ===============
@@ -156,7 +138,11 @@ following arguments:
    ``PCM_FORMAT_U24_3BE``     Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
    =========================  ===============
 
-   * *periodsize* - the period size in frames. Each write should consist of *periodsize* frames. The default value is 32.
+   * *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'``.
@@ -165,14 +151,28 @@ following arguments:
      the `device` keyword argument is ignored.
      ``0`` is the first hardware sound card.
 
-   This will construct a PCM object with the given settings.
+     **Note:** This should not be used, as it bypasses most of ALSA's configuration.
+
+   The defaults mentioned above are values passed by :mod:alsaaudio
+   to ALSA, not anything internal to ALSA.
+
+   **Note:** For default and non-default values alike, there is no
+   guarantee that a PCM device supports the requested configuration,
+   and ALSA may pick realizable values which it believes to be closest
+   to the request. Therefore, after creating a PCM object, it is
+   necessary to verify whether its realized configuration is acceptable.
+   The :func:info method can be used to query it.
+
+   *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.
 
@@ -180,158 +180,284 @@ following arguments:
 
    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()
+.. method:: PCM.info() -> dict
 
-   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) 
-   ===========================  =============================  ==================================================================
+   Returns a dictionary containing the configuration of a PCM device.
 
-   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():".
+   A small subset of properties reflects fixed parameters given by the
+   user, stored within alsaaudio. To distinguish them from properties
+   retrieved from ALSA when the call is made, they have their name
+   prefixed with **" (call value) "**.
 
+   Descriptions of properties which can be directly set during PCM object
+   instantiation carry the prefix "PCM():", followed by the respective
+   constructor parameter. Note that due to device limitations, the values
+   may deviate from those originally requested.
 
-.. method:: PCM.pcmtype()
+   Yet another set of properties cannot be set, and derives directly from
+   the hardware, possibly depending on other properties. Those properties'
+   descriptions are prefixed with "hw:" below.
+
+   ===========================  ====================================  ==================================================================
+        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* [#tss]_  integer (negative indicates error)
+   nominal_bits                 *nominal bits in sample* [#tss]_      integer (negative indicates error)
+   physical_bits                *sample width in bits* [#tss]_        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)
+   ===========================  ====================================  ==================================================================
+.. [#tss] More information in the :ref:`terminology section for sample size <term-sample-size>`
+
+..
+
+   The italicized descriptions give a summary of the "full" description
+   as can be found in the
+   `ALSA documentation <https://www.alsa-project.org/alsa-doc>`_.
+
+   *New in 0.9.1*
+
+.. method:: PCM.dumpinfo()
+
+   Dumps the PCM object's configured parameters to stdout.
+
+.. method:: PCM.pcmtype() -> int
 
    Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
    :const:`PCM_PLAYBACK`.
 
-
-.. method:: PCM.pcmmode()
+.. method:: PCM.pcmmode() -> int
 
    Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
    :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
 
-
-.. method:: PCM.cardname()
+.. method:: PCM.cardname() -> string
 
    Return the name of the sound card used by this PCM object.
 
-.. method:: PCM.setchannels(nchannels)
+..
+   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: int) -> int
 
    .. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
 
-.. method:: PCM.setrate(rate)
+.. method:: PCM.setrate(rate: int) -> int
 
    .. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
 
-.. method:: PCM.setformat(format)
-   
+.. method:: PCM.setformat(format: int) -> int
+
    .. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
 
-.. method:: PCM.setperiodsize(period)
+.. method:: PCM.setperiodsize(period: int) -> int
 
    .. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
 
-.. method:: PCM.read()
+.. method:: PCM.state() -> int
+
+   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.avail() -> int
+
+   For :const:`PCM_PLAYBACK` PCM objects, returns the number of writable
+   (that is, free) frames in the buffer.
+
+   For :const:`PCM_CAPTURE` PCM objects, returns the number of readable
+   (that is, filled) frames in the buffer.
+
+   An attempt to read/write more frames than indicated will block (in
+   :const:`PCM_NORMAL` mode) or fail and return zero (in
+   :const:`PCM_NONBLOCK` mode).
+
+   *New in 0.11*
+
+.. method:: PCM.read() -> tuple[int, bytes]
 
    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 
+   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.
+   In case of a buffer overrun, this function will return the negative
+   size :const:`-EPIPE`, and no data is read.
+   This indicates that data was lost. To resume capturing, just call read
+   again, but note that the stream was already corrupted.
+   To avoid the problem in the future, try using a larger period size
+   and/or more periods, at the cost of higher latency.
 
-.. method:: PCM.write(data)
+.. method:: PCM.write(data: bytes) -> int
 
    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.
+   If the data was successfully written, the call returns the size of the
+   data *in frames*.
+
+   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.
 
    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.
+   should be written again at a later time.
 
+   In case of a buffer underrun, this function will return the negative
+   size :const:`-EPIPE`, and no data is written.
+   At this point, the playback was already corrupted. If you want to play
+   the data nonetheless, call write again with the same data.
+   To avoid the problem in the future, try using a larger period size
+   and/or more periods, at the cost of higher latency.
 
-.. method:: PCM.pause([enable=True])
+   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: int = True]) -> int
 
    If *enable* is :const:`True`, playback or capture is paused.
    Otherwise, playback/capture is resumed.
 
+.. method:: PCM.drop() -> int
 
-.. method:: PCM.polldescriptors()
+   Stop the stream and drop residual buffered frames.
 
-   Returns a tuple of *(file descriptor, eventmask)* that can be used to
-   wait for changes on the PCM with *select.poll*.
+   *New in 0.9*
 
-   The *eventmask* value is compatible with `poll.register`__ in the Python 
+.. method:: PCM.drain() -> int
+
+   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() -> None
+
+   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() -> list[tuple[int, int]]
+
+   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])
+.. method:: PCM.polldescriptors_revents(descriptors: list[tuple[int, int]]) -> int
+
+   Processes the descriptor list returned by :func:`polldescriptors` after
+   using it with *select.poll*, and returns a single *eventmask* value that
+   is meaningful for deciding whether :func:`read` or :func:`write` should
+   be called.
+
+   *New in 0.11*
+
+.. method:: PCM.set_tstamp_mode([mode: int = 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()
+.. method:: PCM.get_tstamp_mode() -> int
 
    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])
+.. method:: PCM.set_tstamp_type([type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY]) -> None
 
    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()
+.. method:: PCM.get_tstamp_type() -> int
 
    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()
+.. method:: PCM.htimestamp() -> tuple[int, int, int]
 
    Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*.
 
@@ -347,7 +473,7 @@ PCM objects have the following methods:
    ``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.
 
    =================================  ===========================================
@@ -358,16 +484,13 @@ PCM objects have the following methods:
                                       update.
    =================================  ===========================================
 
-
-__ poll_objects_
-
 **A few hints on using PCM devices for playback**
 
-The most common reason for problems with playback of PCM audio is that writes 
+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
+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).
 
@@ -396,13 +519,12 @@ Mixer Objects
 
 Mixer objects provides access to the ALSA mixer API.
 
-
-.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default')
+.. 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 
+     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.
@@ -412,35 +534,32 @@ Mixer objects provides access to the ALSA mixer API.
    * *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. 
+     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()
+.. method:: Mixer.cardname() -> str
 
    Return the name of the sound card used by this Mixer object
 
-
-.. method:: Mixer.mixer()
+.. method:: Mixer.mixer() -> str
 
    Return the name of the specific mixer controlled by this object, For example
    ``'Master'`` or ``'PCM'``
 
-
-.. method:: Mixer.mixerid()
+.. method:: Mixer.mixerid() -> int
 
    Return the ID of the ALSA mixer controlled by this object.
 
-
-.. method:: Mixer.switchcap()
+.. method:: Mixer.switchcap() -> int
 
    Returns a list of the switches which are defined by this specific mixer.
    Possible values in this list are:
@@ -452,7 +571,7 @@ Mixer objects have the following methods:
    '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 
+   '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
    ======================  ================
@@ -460,8 +579,7 @@ Mixer objects have the following methods:
    To manipulate these switches use the :meth:`setrec` or
    :meth:`setmute` methods
 
-
-.. method:: Mixer.volumecap()
+.. method:: Mixer.volumecap() -> int
 
    Returns a list of the volume control capabilities of this
    mixer. Possible values in the list are:
@@ -476,8 +594,8 @@ Mixer objects have the following methods:
    'Capture Volume'          Manipulate sound capture volume
    'Joined Capture Volume'   Manipulate sound capture volume for all channels at a time
    ========================  ================
-   
-.. method:: Mixer.getenum()
+
+.. method:: Mixer.getenum() -> tuple[str, list[str]]
 
    For enumerated controls, return the currently selected item and  the list of
    items available.
@@ -503,48 +621,43 @@ Mixer objects have the following methods:
    This method will return an empty tuple if the mixer is not an  enumerated
    control.
 
+.. method:: Mixer.setenum(index: int) -> None
 
-.. method:: Mixer.getmute()
+   For enumerated controls, sets the currently selected item.
+   *index* is an index into the list of available enumerated items returned
+   by :func:`getenum`.
 
-   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.getrange(pcmtype=PCM_PLAYBACK)
+.. 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.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.getvolume(pcmtype=PCM_PLAYBACK)
+.. 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 integer percentages.
+   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)
+.. 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
-   controls the new volume setting as an integer percentage.
+   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
@@ -555,7 +668,17 @@ Mixer objects have the following methods:
    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])
+   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.
@@ -565,8 +688,14 @@ Mixer objects have the following methods:
 
    This method will fail if the mixer has no playback mute capabilities
 
+.. method:: Mixer.getrec() -> list[int]
 
-.. method:: Mixer.setrec(capture, [channel])
+   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.
@@ -576,22 +705,24 @@ Mixer objects have the following methods:
 
    This method will fail if the mixer has no capture switch capabilities.
 
-.. method:: Mixer.polldescriptors()
+.. method:: Mixer.polldescriptors() -> list[tuple[int, int]]
 
-   Returns a tuple of *(file descriptor, eventmask)* that can be used to
-   wait for changes on the mixer with *select.poll*.
+   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 
+   The *eventmask* value is compatible with `poll.register`__ in the Python
    :const:`select` module.
 
-__ poll_objects_
+.. method:: Mixer.handleevents() -> int
 
-.. method:: Mixer.handleevents()
-
-   Acknowledge events on the *polldescriptors* file descriptors
+   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.
@@ -614,8 +745,6 @@ 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.
 
-.. % ==== 4. ====
-
 
 .. _pcm-example:
 
@@ -629,7 +758,7 @@ The following example are provided:
 * `playbacktest.py`
 * `mixertest.py`
 
-All examples (except `mixertest.py`) accept the commandline option 
+All examples (except `mixertest.py`) accept the commandline option
 *-c <cardname>*.
 
 To determine a valid card name, use the commandline ALSA player::
@@ -644,12 +773,12 @@ or::
    >>> alsaaudio.pcms()
 
 mixertest.py accepts the commandline options *-d <device>* and
-*-c <cardindex>*. 
+*-c <cardindex>*.
 
 playwav.py
 ~~~~~~~~~~
 
-**playwav.py** plays a wav file. 
+**playwav.py** plays a wav file.
 
 To test PCM playback (on your default soundcard), run::
 
@@ -657,6 +786,7 @@ To test PCM playback (on your default soundcard), run::
 
 recordtest.py and playbacktest.py
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
 **recordtest.py** and **playbacktest.py** will record and play a raw
 sound file in CD quality.
 
@@ -678,7 +808,7 @@ 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'
@@ -696,7 +826,7 @@ The output might look like this::
      'Mix'
      'Mix Mono'
 
-With a single argument - the *control*, it will display the settings of 
+With a single argument - the *control*, it will display the settings of
 that control; for example::
 
   $ ./mixertest.py Master
@@ -705,7 +835,7 @@ that control; for example::
   Channel 0 volume: 61%
   Channel 1 volume: 61%
 
-With two arguments, the *control* and a *parameter*, it will set the 
+With two arguments, the *control* and a *parameter*, it will set the
 parameter on the mixer::
 
   $ ./mixertest.py Master mute
@@ -726,9 +856,3 @@ argument::
   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.
-
-.. _poll_objects: http://docs.python.org/library/select.html#poll-objects

doc/pyalsaaudio.rst

@@ -7,33 +7,19 @@ Introduction
 
 .. |release| replace:: version
 
-.. % At minimum, give your name and an email address.  You can include a
-.. % snail-mail address if you like.
-
-.. % This makes the Abstract go on a separate page in the HTML version;
-.. % if a copyright notice is used, it should go immediately after this.
-.. % 
-
-
 .. _front:
 
 This software is licensed under the PSF license - the same one used by the
 majority of the python distribution. Basically you can use it for anything you
 wish (even commercial purposes). There is no warranty whatsoever.
 
-.. % Copyright statement should go here, if needed.
-
-.. % The abstract should be a paragraph or two long, and describe the
-.. % scope of the document.
-
-
 .. topic:: Abstract
 
    This package contains wrappers for accessing the ALSA API from Python. It is
    currently fairly complete for PCM devices and Mixer access. MIDI sequencer
    support is low on our priority list, but volunteers are welcome.
 
-   If you find bugs in the wrappers please use thegithub issue tracker. 
+   If you find bugs in the wrappers please use the github issue tracker.
    Please don't send bug reports regarding ALSA specifically. There are several
    bugs in this API, and those should be reported to the ALSA team - not me.
 
@@ -64,8 +50,8 @@ More information about ALSA may be found on the project homepage
 ALSA and Python
 ===============
 
-The older Linux sound API (OSS) which is now deprecated is well supported from
-the standard Python library, through the ossaudiodev module. No native ALSA
+The older Linux sound API (OSS) -- which is now deprecated -- is well supported
+by the standard Python library, through the ossaudiodev module. No native ALSA
 support exists in the standard library.
 
 There are a few other "ALSA for Python" projects available, including at least
@@ -95,7 +81,7 @@ and need the ALSA headers for compilation.  Verify that you have
 
 Naturally you also need to use a kernel with proper ALSA support. This is the
 default in Linux kernel 2.6 and later. If you are using kernel version 2.4 you
-may need to install the ALSA patches yourself - although most distributions 
+may need to install the ALSA patches yourself - although most distributions
 ship with ALSA kernels.
 
 To install, execute the following:  ---   ::
@@ -106,6 +92,7 @@ And then as root:  ---   ::
 
    # python setup.py install
 
+
 *******
 Testing
 *******
@@ -130,7 +117,7 @@ with ``Ctl-C``.
 
 Play back the recording with::
 
-   $ python playbacktest.py-d <device>  <filename>
+   $ python playbacktest.py -d <device> <filename>
 
 There is a minimal test suite in :code:`test.py`, but it is
 a bit dependent on the ALSA configuration and may fail without indicating

doc/terminology.rst

@@ -6,7 +6,7 @@ In order to use PCM devices it is useful to be familiar with some concepts and
 terminology.
 
 Sample
-   PCM audio, whether it is input or output, consists of *samples*. 
+   PCM audio, whether it is input or output, consists of *samples*.
    A single sample represents the amplitude of one channel of sound
    at a certain point in time. A lot of individual samples are
    necessary to represent actual sound; for CD audio, 44100 samples
@@ -19,26 +19,26 @@ Sample
 
    Musically, the sample size determines the dynamic range. The
    dynamic range is the difference between the quietest and the
-   loudest signal that can be resproduced.
+   loudest signal that can be reproduced.
 
 Frame
-   A frame consists of exactly one sample per channel. If there is only one 
-   channel (Mono sound) a frame is simply a single sample. If the sound is 
+   A frame consists of exactly one sample per channel. If there is only one
+   channel (Mono sound) a frame is simply a single sample. If the sound is
    stereo, each frame consists of two samples, etc.
 
 Frame size
    This is the size in bytes of each frame. This can vary a lot: if each sample
-    is 8 bits, and we're handling mono sound, the frame size is one byte. 
-	Similarly in 6 channel audio with 64 bit floating point samples, the frame 
-	size is 48 bytes
+   is 8 bits, and we're handling mono sound, the frame size is one byte.
+   For six channel audio with 64 bit floating point samples, the frame size
+   is 48 bytes.
 
 Rate
-   PCM sound consists of a flow of sound frames. The sound rate controls how 
+   PCM sound consists of a flow of sound frames. The sound rate controls how
    often the current frame is replaced. For example, a rate of 8000 Hz
    means that a new frame is played or captured 8000 times per second.
 
 Data rate
-   This is the number of bytes, which must be recorded or provided per
+   This is the number of bytes which must be consumed or provided per
    second at a certain frame size and rate.
 
    8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
@@ -46,29 +46,60 @@ Data rate
 
    At the other end of the scale, 96000 Hz, 6 channel sound with 64
    bit (8 bytes) samples has a data rate of 96000 \* 6 \* 8 = 4608
-   kb/s (almost 5 MB sound data per second)
+   kb/s (almost 5 MB sound data per second).
+
+   If the data rate requirement is not met, an overrun (on capture) or
+   underrun (on playback) occurs; the term "xrun" is used to refer to
+   either event.
+
+.. _term-period:
 
 Period
-   When the hardware processes data this is done in chunks of frames. The time
-   interval between each processing (A/D or D/A conversion) is known
-   as the period.
-   The size of the period has direct implication on the latency of the
-   sound input or output. For low-latency the period size should be
-   very small, while low CPU resource usage would usually demand
-   larger period sizes. With ALSA, the CPU utilization is not impacted
-   much by the period size, since the kernel layer buffers multiple
-   periods internally, so each period generates an interrupt and a
-   memory copy, but userspace can be slower and read or write multiple
-   periods at the same time.
+   The CPU processes sample data in chunks of frames, so-called periods
+   (also called fragments by some systems). The operating system kernel's
+   sample buffer must hold at least two periods (at any given time, one
+   is processed by the sound hardware, and one by the CPU).
+
+   The completion of a *period* triggers a CPU interrupt, which causes
+   processing and context switching overhead. Therefore, a smaller period
+   size causes higher CPU resource usage at a given data rate.
+
+   A bigger size of the *buffer* improves the system's resilience to xruns.
+   The buffer being split into a bigger number of smaller periods also does
+   that, as it allows it to be drained / topped up sooner.
+
+   On the other hand, a bigger size of the *buffer* also increases the
+   playback latency, that is, the time it takes for a frame from being
+   sent out by the application to being actually audible.
+
+   Similarly, a bigger *period* size increases the capture latency.
+
+   The trade-off between latency, xrun resilience, and resource usage
+   must be made depending on the application.
 
 Period size
-   This is the size of each period in Hz. *Not bytes, but Hz!.* In 
-   :mod:`alsaaudio` the period size is set directly, and it is
+   This is the size of each period in frames. *Not bytes, but frames!*
+   In :mod:`alsaaudio` the period size is set directly, and it is
    therefore important to understand the significance of this
    number. If the period size is configured to for example 32,
    each write should contain exactly 32 frames of sound data, and each
    read will return either 32 frames of data or nothing at all.
 
+.. _term-sample-size:
+
+Sample size
+   Each sample takes *physical_bits* of space. *nominal_bits* tells
+   how many least significant bits are used. This is the bit depth
+   in the format name and sometimes called just *sample bits* or
+   *format width*. *significant_bits* tells how many most significant
+   bits of the *nominal_bits* are used by the sample. This can be thought
+   of as the *sample resolution*. This is visualized as follows::
+
+    MSB |00000000 XXXXXXXX XXXXXXXX 00000000| LSB
+                  |--significant--|
+                  |---------nominal---------|
+        |-------------physical--------------|
+
 Once you understand these concepts, you will be ready to use the PCM API. Read
 on.
 

isine.py

@@ -7,6 +7,7 @@
 from __future__ import print_function
 
 import sys
+import time
 from threading import Thread
 from multiprocessing import Queue
 
@@ -15,14 +16,15 @@ if sys.version_info[0] < 3:
 else:
     from queue import Empty
 
-from math import pi, sin
+from math import pi, sin, ceil
 import struct
+import itertools
 import alsaaudio
 
 sampling_rate = 48000
 
 format = alsaaudio.PCM_FORMAT_S16_LE
-framesize = 2 # bytes per frame for the values above
+pack_format = 'h'  # short int, matching S16
 channels = 2
 
 def nearest_frequency(frequency):
@@ -34,28 +36,28 @@ def generate(frequency, duration = 0.125):
     # generate a buffer with a sine wave of `frequency`
     # that is approximately `duration` seconds long
 
-    # the buffersize we approximately want
-    target_size = int(sampling_rate * channels * duration)
-
     # the length of a full sine wave at the frequency
     cycle_size = int(sampling_rate / frequency)
 
-    # number of full cycles we can fit into target_size
-    factor = int(target_size / cycle_size)
+    # number of full cycles we can fit into the duration
+    factor = int(ceil(duration * frequency))
+
+    # total number of frames
+    size = cycle_size * factor
 
-    size = max(int(cycle_size * factor), 1)
-    
     sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \
              for i in range(size)]
-             
-    return struct.pack('%dh' % size, *sine)
-                                                                                  
+
+    if channels > 1:
+        sine = list(itertools.chain.from_iterable(itertools.repeat(x, channels) for x in sine))
+
+    return struct.pack(str(size * channels) + pack_format, *sine)
+
 
 class SinePlayer(Thread):
-    
+
     def __init__(self, frequency = 440.0):
-        Thread.__init__(self)
-        self.setDaemon(True)
+        Thread.__init__(self, daemon=True)
         self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
         self.queue = Queue()
         self.change(frequency)
@@ -73,22 +75,23 @@ class SinePlayer(Thread):
 
         buf = generate(f)
         self.queue.put(buf)
-                        
+
     def run(self):
         buffer = None
         while True:
             try:
                 buffer = self.queue.get(False)
-                self.device.setperiodsize(int(len(buffer) / framesize))
-                self.device.write(buffer)
             except Empty:
-                if buffer:
-                    self.device.write(buffer)
-                
+                pass
+            if buffer:
+                if self.device.write(buffer) < 0:
+                    print("Playback buffer underrun! Continuing nonetheless ...")
+
 
 isine = SinePlayer()
 isine.start()
 
-def change(f):
-    isine.change(f)
-    
+time.sleep(1)
+isine.change(1000)
+time.sleep(1)
+

loopback.py

@@ -0,0 +1,403 @@
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
+
+import sys
+import select
+import logging
+import re
+import struct
+import subprocess
+from datetime import datetime, timedelta
+from alsaaudio import (PCM, pcms, PCM_PLAYBACK, PCM_CAPTURE, PCM_NONBLOCK, Mixer,
+	PCM_STATE_OPEN, PCM_STATE_SETUP, PCM_STATE_PREPARED, PCM_STATE_RUNNING, PCM_STATE_XRUN, PCM_STATE_DRAINING,
+	PCM_STATE_PAUSED, PCM_STATE_SUSPENDED, ALSAAudioError)
+from argparse import ArgumentParser
+
+poll_names = {
+	select.POLLIN: 'POLLIN',
+	select.POLLPRI: 'POLLPRI',
+	select.POLLOUT: 'POLLOUT',
+	select.POLLERR: 'POLLERR',
+	select.POLLHUP: 'POLLHUP',
+	select.POLLRDHUP: 'POLLRDHUP',
+	select.POLLNVAL: 'POLLNVAL'
+}
+
+state_names = {
+	PCM_STATE_OPEN: 'PCM_STATE_OPEN',
+	PCM_STATE_SETUP: 'PCM_STATE_SETUP',
+	PCM_STATE_PREPARED: 'PCM_STATE_PREPARED',
+	PCM_STATE_RUNNING: 'PCM_STATE_RUNNING',
+	PCM_STATE_XRUN: 'PCM_STATE_XRUN',
+	PCM_STATE_DRAINING: 'PCM_STATE_DRAINING',
+	PCM_STATE_PAUSED: 'PCM_STATE_PAUSED',
+	PCM_STATE_SUSPENDED: 'PCM_STATE_SUSPENDED'
+}
+
+def poll_desc(mask):
+	return '|'.join([poll_names[bit] for bit, name in poll_names.items() if mask & bit])
+
+class PollDescriptor(object):
+	'''File Descriptor, event mask and a name for logging'''
+	def __init__(self, name, fd, mask):
+		self.name = name
+		self.fd = fd
+		self.mask = mask
+
+	def as_tuple(self):
+		return (self.fd, self.mask)
+
+	@classmethod
+	def from_alsa_object(cls, name, alsaobject, mask=None):
+		# TODO maybe refactor: we ignore objects that have more then one polldescriptor
+		fd, alsamask = alsaobject.polldescriptors()[0]
+
+		if mask is None:
+			mask = alsamask
+
+		return cls(name, fd, mask)
+
+class Loopback(object):
+	'''Loopback state and event handling'''
+
+	def __init__(self, capture, playback_args, volume_handler, run_after_stop=None, run_before_start=None):
+		self.playback_args = playback_args
+		self.playback = None
+		self.volume_handler = volume_handler
+		self.capture_started = None
+		self.last_capture_event = None
+
+		self.capture = capture
+		self.capture_pd = PollDescriptor.from_alsa_object('capture', capture)
+
+		self.run_after_stop = None
+		if run_after_stop:
+			self.run_after_stop = run_after_stop.split(' ')
+
+		self.run_before_start = None
+		if run_before_start:
+			self.run_before_start = run_before_start.split(' ')
+		self.run_after_stop_did_run = False
+
+		self.waitBeforeOpen = False
+		self.queue = []
+
+		self.period_size = 0
+		self.silent_periods = 0
+
+	@staticmethod
+	def compute_energy(data):
+		values = struct.unpack(f'{len(data)//2}h', data)
+		e = 0
+		for v in values:
+			e = e + v * v
+
+		return e
+
+	@staticmethod
+	def run_command(cmd):
+		if cmd:
+			rc = subprocess.run(cmd)
+			if rc.returncode:
+				logging.warning(f'run {cmd}, return code {rc.returncode}')
+			else:
+				logging.info(f'run {cmd}, return code {rc.returncode}')
+
+	def register(self, reactor):
+		reactor.register_timeout_handler(self.timeout_handler)
+		reactor.register(self.capture_pd, self)
+
+	def start(self):
+		# start reading data
+		size, data = self.capture.read()
+		if size:
+			self.queue.append(data)
+
+	def timeout_handler(self):
+		if self.playback and self.capture_started:
+			if self.last_capture_event:
+				if datetime.now() - self.last_capture_event > timedelta(seconds=2):
+					logging.info('timeout - closing playback device')
+					self.playback.close()
+					self.playback = None
+					self.capture_started = None
+					if self.volume_handler:
+						self.volume_handler.stop()
+					self.run_command(self.run_after_stop)
+			return
+
+		self.waitBeforeOpen = False
+
+		if not self.run_after_stop_did_run and not self.playback:
+			if self.volume_handler:
+				self.volume_handler.stop()
+			self.run_command(self.run_after_stop)
+			self.run_after_stop_did_run = True
+
+	def pop(self):
+		if len(self.queue):
+			return self.queue.pop()
+		else:
+			return None
+
+	def handle_capture_event(self, eventmask, name):
+		'''called when data is available for reading'''
+		self.last_capture_event = datetime.now()
+		size, data = self.capture.read()
+		if not size:
+			logging.warning(f'capture event but no data')
+			return False
+
+		energy = self.compute_energy(data)
+		logging.debug(f'energy: {energy}')
+
+		# the usecase is a USB capture device where we get perfect silence when it's idle
+		if energy == 0:
+			self.silent_periods = self.silent_periods + 1
+
+			# turn off playback after two seconds of silence
+			# 2 channels * 2 seconds * 2 bytes per sample
+			fps = self.playback_args['rate']  * 8 // (self.playback_args['periodsize'] * self.playback_args['periods'])
+
+			logging.debug(f'{self.silent_periods} of {fps} silent periods: {self.playback}')
+
+			if self.silent_periods > fps and self.playback:
+				logging.info(f'closing playback due to silence')
+				self.playback.close()
+				self.playback = None
+				if self.volume_handler:
+					self.volume_handler.stop()
+				self.run_command(self.run_after_stop)
+				self.run_after_stop_did_run = True
+
+			if not self.playback:
+				return
+		else:
+			self.silent_periods = 0
+
+		if not self.playback:
+			if self.waitBeforeOpen:
+				return False
+			try:
+				if self.volume_handler:
+					self.volume_handler.start()
+				self.run_command(self.run_before_start)
+				self.playback = PCM(**self.playback_args)
+				self.period_size = self.playback.info()['period_size']
+				logging.info(f'opened playback device with period_size {self.period_size}')
+			except ALSAAudioError as e:
+				logging.info('opening PCM playback device failed: %s', e)
+				self.waitBeforeOpen = True
+				return False
+
+			self.capture_started = datetime.now()
+			logging.info(f'{self.playback} capture started: {self.capture_started}')
+
+		self.queue.append(data)
+
+		if len(self.queue) <= 2:
+			logging.info(f'buffering: {len(self.queue)}')
+			return False
+
+		try:
+			data = self.pop()
+			if data:
+				space = self.playback.avail()
+				written = self.playback.write(data)
+				logging.debug(f'wrote {written} bytes while space was {space}')
+		except ALSAAudioError:
+			logging.error('underrun', exc_info=1)
+
+		return True
+
+	def __call__(self, fd, eventmask, name):
+
+		if fd == self.capture_pd.fd:
+			real_mask = self.capture.polldescriptors_revents([self.capture_pd.as_tuple()])
+			if real_mask:
+				return self.handle_capture_event(real_mask, name)
+			else:
+				logging.debug('null capture event')
+				return False
+		else:
+			real_mask = self.playback.polldescriptors_revents([self.playback_pd.as_tuple()])
+			if real_mask:
+				return self.handle_playback_event(real_mask, name)
+			else:
+				logging.debug('null playback event')
+				return False
+
+class VolumeForwarder(object):
+	'''Volume control event handling'''
+
+	def __init__(self, capture_control, playback_control):
+		self.playback_control = playback_control
+		self.capture_control = capture_control
+		self.active = True
+		self.volume = None
+
+	def start(self):
+		self.active = True
+		if self.volume:
+			self.volume = playback_control.setvolume(self.volume)
+
+	def stop(self):
+		self.active = False
+		self.volume = self.playback_control.getvolume(pcmtype=PCM_CAPTURE)[0]
+
+	def __call__(self, fd, eventmask, name):
+		if not self.active:
+			return
+
+		volume = self.capture_control.getvolume(pcmtype=PCM_CAPTURE)
+		# indicate that we've handled the event
+		self.capture_control.handleevents()
+		logging.info(f'{name} adjusting volume to {volume}')
+		if volume:
+			self.playback_control.setvolume(volume[0])
+
+
+class Reactor(object):
+	'''A wrapper around select.poll'''
+
+	def __init__(self):
+		self.poll = select.poll()
+		self.descriptors = {}
+		self.timeout_handlers = set()
+
+	def register(self, polldescriptor, callable):
+		logging.debug(f'registered {polldescriptor.name}: {poll_desc(polldescriptor.mask)}')
+		self.descriptors[polldescriptor.fd] = (polldescriptor, callable)
+		self.poll.register(polldescriptor.fd, polldescriptor.mask)
+
+	def unregister(self, polldescriptor):
+		self.poll.unregister(polldescriptor.fd)
+		del self.descriptors[polldescriptor.fd]
+
+	def register_timeout_handler(self, callable):
+		self.timeout_handlers.add(callable)
+
+	def unregister_timeout_handler(self, callable):
+		self.timeout_handlers.remove(callable)
+
+	def run(self):
+		last_timeout_ev = datetime.now()
+		while True:
+			# poll for a bit, then send a timeout to registered handlers
+			events = self.poll.poll(0.25)
+			for fd, ev in events:
+				polldescriptor, handler = self.descriptors[fd]
+
+				# very chatty - log all events
+				# logging.debug(f'{polldescriptor.name}: {poll_desc(ev)} ({ev})')
+
+				handler(fd, ev, polldescriptor.name)
+
+			if datetime.now() - last_timeout_ev > timedelta(seconds=0.25):
+				for t in self.timeout_handlers:
+					t()
+				last_timeout_ev = datetime.now()
+
+
+if __name__ == '__main__':
+
+	logging.basicConfig(format='%(asctime)s %(levelname)s %(message)s', level=logging.INFO)
+
+	parser = ArgumentParser(description='ALSA loopback (with volume forwarding)')
+
+	playback_pcms = pcms(pcmtype=PCM_PLAYBACK)
+	capture_pcms = pcms(pcmtype=PCM_CAPTURE)
+
+	if not playback_pcms:
+		logging.error('no playback PCM found')
+		sys.exit(2)
+
+	if not capture_pcms:
+		logging.error('no capture PCM found')
+		sys.exit(2)
+
+	parser.add_argument('-d', '--debug', action='store_true')
+	parser.add_argument('-i', '--input', default=capture_pcms[0])
+	parser.add_argument('-o', '--output', default=playback_pcms[0])
+	parser.add_argument('-r', '--rate', type=int, default=44100)
+	parser.add_argument('-c', '--channels', type=int, default=2)
+	parser.add_argument('-p', '--periodsize', type=int, default=444) # must be divisible by 6 for 44k1
+	parser.add_argument('-P', '--periods', type=int, default=2)
+	parser.add_argument('-I', '--input-mixer', help='Control of the input mixer, can contain the card index, e.g. Digital:2')
+	parser.add_argument('-O', '--output-mixer', help='Control of the output mixer, can contain the card index, e.g. PCM:1')
+	parser.add_argument('-A', '--run-after-stop', help='command to run when the capture device is idle/silent')
+	parser.add_argument('-B', '--run-before-start', help='command to run when the capture device becomes active')
+	parser.add_argument('-V', '--volume', help='Initial volume (default is leave unchanged)')
+
+	args = parser.parse_args()
+
+	if args.debug:
+		logging.getLogger().setLevel(logging.DEBUG)
+
+	playback_args = {
+		'type': PCM_PLAYBACK,
+		'mode': PCM_NONBLOCK,
+		'device': args.output,
+		'rate': args.rate,
+		'channels': args.channels,
+		'periodsize': args.periodsize,
+		'periods': args.periods
+	}
+
+	reactor = Reactor()
+
+	# If args.input_mixer and args.output_mixer are set, forward the capture volume to the playback volume.
+	# The usecase is a capture device that is implemented using g_audio, i.e. the Linux USB gadget driver.
+	# When a USB device (eg. an iPad) is connected to this machine, its volume events will go to the volume control
+	# of the output device
+
+	capture = None
+	playback = None
+	volume_handler = None
+	if args.input_mixer and args.output_mixer:
+		re_mixer = re.compile(r'([a-zA-Z0-9]+):?([0-9+])?')
+
+		input_mixer_card = None
+		m = re_mixer.match(args.input_mixer)
+		if m:
+			input_mixer = m.group(1)
+			if m.group(2):
+				input_mixer_card = int(m.group(2))
+		else:
+			parser.print_usage()
+			sys.exit(1)
+
+		output_mixer_card = None
+		m = re_mixer.match(args.output_mixer)
+		if m:
+			output_mixer = m.group(1)
+			if m.group(2):
+				output_mixer_card = int(m.group(2))
+		else:
+			parser.print_usage()
+			sys.exit(1)
+
+		if input_mixer_card is None:
+			capture = PCM(type=PCM_CAPTURE, mode=PCM_NONBLOCK, device=args.input, rate=args.rate,
+				channels=args.channels, periodsize=args.periodsize, periods=args.periods)
+			input_mixer_card = capture.info()['card_no']
+
+		if output_mixer_card is None:
+			playback = PCM(**playback_args)
+			output_mixer_card = playback.info()['card_no']
+			playback.close()
+
+		playback_control = Mixer(control=output_mixer, cardindex=int(output_mixer_card))
+		capture_control = Mixer(control=input_mixer, cardindex=int(input_mixer_card))
+
+		volume_handler = VolumeForwarder(capture_control, playback_control)
+		reactor.register(PollDescriptor.from_alsa_object('capture_control', capture_control, select.POLLIN), volume_handler)
+
+	if args.volume and playback_control:
+		playback_control.setvolume(int(args.volume))
+
+	loopback = Loopback(capture, playback_args, volume_handler, args.run_after_stop, args.run_before_start)
+	loopback.register(reactor)
+	loopback.start()
+
+	reactor.run()

mixertest.py

@@ -72,8 +72,13 @@ def show_mixer(name, kwargs):
     volumes = mixer.getvolume()
     volumes_dB = mixer.getvolume(units=alsaaudio.VOLUME_UNITS_DB)
     for i in range(len(volumes)):
-        print("Channel %i volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
-        
+        print("Channel %i playback volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
+
+    volumes = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE)
+    volumes_dB = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE, units=alsaaudio.VOLUME_UNITS_DB)
+    for i in range(len(volumes)):
+        print("Channel %i capture volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
+
     try:
         mutes = mixer.getmute()
         for i in range(len(mutes)):
@@ -113,7 +118,7 @@ def set_mixer(name, args, kwargs):
             mixer.setmute(1, channel)
         else:
             mixer.setmute(0, channel)
-        
+
     elif args in ['rec','unrec']:
         # Enable/disable recording
         if args == 'rec':

playbacktest.py

@@ -47,7 +47,8 @@ if __name__ == '__main__':
     # Read data from stdin
     data = f.read(320)
     while data:
-        out.write(data)
+        if out.write(data) < 0:
+            print("Playback buffer underrun! Continuing nonetheless ...")
         data = f.read(320)
-            
-        
+    out.close()
+

playwav.py

@@ -39,7 +39,8 @@ def play(device, f):
 	data = f.readframes(periodsize)
 	while data:
 		# Read data from stdin
-		device.write(data)
+		if device.write(data) < 0:
+			print("Playback buffer underrun! Continuing nonetheless ...")
 		data = f.readframes(periodsize)
 
 

recordtest.py

@@ -40,7 +40,7 @@ if __name__ == '__main__':
 
 	f = open(args[0], 'wb')
 
-	# Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz 
+	# Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz
 	# and 16 bit little endian samples
 	# The period size controls the internal number of frames per period.
 	# The significance of this parameter is documented in the ALSA api.
@@ -49,8 +49,8 @@ if __name__ == '__main__':
 	# This means that the reads below will return either 320 bytes of data
 	# or 0 bytes of data. The latter is possible because we are in nonblocking
 	# mode.
-	inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, 
-		channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, 
+	inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK,
+		channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE,
 		periodsize=160, device=device)
 
 	loops = 1000000
@@ -59,6 +59,8 @@ if __name__ == '__main__':
 		# Read data from device
 		l, data = inp.read()
 
-		if l:
+		if l < 0:
+			print("Capture buffer overrun! Continuing nonetheless ...")
+		elif l:
 			f.write(data)
 			time.sleep(.001)

setup.py

@@ -8,7 +8,7 @@ from setuptools import setup
 from setuptools.extension import Extension
 from sys import version
 
-pyalsa_version = '0.9.2'
+pyalsa_version = '0.10.1'
 
 if __name__ == '__main__':
     setup(
@@ -29,12 +29,12 @@ if __name__ == '__main__':
             'License :: OSI Approved :: Python Software Foundation License',
             'Operating System :: POSIX :: Linux',
             'Programming Language :: Python :: 2',
-            'Programming Language :: Python :: 3',    
+            'Programming Language :: Python :: 3',
             'Topic :: Multimedia :: Sound/Audio',
             'Topic :: Multimedia :: Sound/Audio :: Mixers',
             'Topic :: Multimedia :: Sound/Audio :: Players',
             'Topic :: Multimedia :: Sound/Audio :: Capture/Recording',
             ],
-        ext_modules=[Extension('alsaaudio',['alsaaudio.c'], 
+        ext_modules=[Extension('alsaaudio',['alsaaudio.c'],
                                libraries=['asound'])]
     )

test.py

@@ -11,6 +11,7 @@
 import unittest
 import alsaaudio
 import warnings
+from contextlib import closing
 
 # we can't test read and write well - these are tested otherwise
 PCMMethods = [
@@ -20,7 +21,7 @@ PCMMethods = [
 ]
 
 PCMDeprecatedMethods = [
-	('setchannels', (2,)), 
+	('setchannels', (2,)),
 	('setrate', (44100,)),
 	('setformat', (alsaaudio.PCM_FORMAT_S8,)),
 	('setperiodsize', (320,))
@@ -49,10 +50,10 @@ class MixerTest(unittest.TestCase):
 
 	def testMixer(self):
 		"""Open the default Mixers and the Mixers on every card"""
-		
+
 		for c in alsaaudio.card_indexes():
 			mixers = alsaaudio.mixers(cardindex=c)
-			
+
 			for m in mixers:
 				mixer = alsaaudio.Mixer(m, cardindex=c)
 				mixer.close()
@@ -73,7 +74,7 @@ class MixerTest(unittest.TestCase):
 		mixer.close()
 
 	def testMixerClose(self):
-		"""Run common Mixer methods on a closed object and verify it raises an 
+		"""Run common Mixer methods on a closed object and verify it raises an
 		error"""
 
 		mixers = alsaaudio.mixers()
@@ -133,7 +134,7 @@ class PCMTest(unittest.TestCase):
 				pcm = alsaaudio.PCM(card='default')
 			except alsaaudio.ALSAAudioError:
 				pass
-				
+
 			# Verify we got a DepreciationWarning
 			self.assertEqual(len(w), 1, "PCM(card='default') expected a warning" )
 			self.assertTrue(issubclass(w[-1].category, DeprecationWarning), "PCM(card='default') expected a DeprecationWarning")
@@ -156,5 +157,31 @@ class PCMTest(unittest.TestCase):
 				self.assertEqual(len(w), 1, method + " expected a warning")
 				self.assertTrue(issubclass(w[-1].category, DeprecationWarning), method + " expected a DeprecationWarning")
 
+class PollDescriptorArgsTest(unittest.TestCase):
+	'''Test invalid args for polldescriptors_revents (takes a list of tuples of 2 integers)'''
+	def testArgsNoList(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents('foo')
+
+	def testArgsListButNoTuples(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents(['foo', 1])
+
+	def testArgsListButInvalidTuples(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents([('foo', 'bar')])
+
+	def testArgsListTupleWrongLength(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents([(1, )])
+
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents([(1, 2, 3)])
+
+
 if __name__ == '__main__':
 	unittest.main()