Debugging underruns, needs cleanup

Small cleanup
Cleanup
2025-12-01 22:33:30 +00:00 · 2025-12-01 22:33:30 +00:00 · 2025-12-01 22:33:30 +00:00 · 2025-12-01 22:33:30 +00:00 · 2025-12-01 22:33:30 +00:00 · 2025-12-01 22:33:30 +00:00
27 changed files with 1420 additions and 373 deletions
--- a/.github/workflows/docs.yml
+++ b/.github/workflows/docs.yml
@@ -0,0 +1,40 @@
+on:
+  workflow_dispatch:
+  push:
+    branches:
+      - main
+
+concurrency:
+  group: "pages"
+  cancel-in-progress: false
+
+jobs:
+  build:
+    runs-on: ubuntu-latest
+    steps:
+      - uses: actions/checkout@v5
+      - name: Install uv
+        uses: astral-sh/setup-uv@v6
+      - name: Install ALSA headers
+        run: sudo apt-get install libasound2-dev
+      - name: Build HTML documentation
+        run: make -C doc html
+      - name: Upload artifacts
+        uses: actions/upload-pages-artifact@v4
+        with:
+          path: doc/html
+
+  deploy:
+    runs-on: ubuntu-latest
+    needs: build
+    permissions:
+      contents: read
+      pages: write
+      id-token: write
+    environment:
+      name: github-pages
+      url: ${{ steps.deployment.outputs.page_url }}
+    steps:
+      - name: Deploy to GitHub Pages
+        id: deployment
+        uses: actions/deploy-pages@v4
--- a/.github/workflows/release.yml
+++ b/.github/workflows/release.yml
@@ -0,0 +1,55 @@
+# Adapted from: https://github.com/pypa/cibuildwheel/blob/main/examples/github-deploy.yml
+
+on:
+  workflow_dispatch:
+  push:
+    tags: "*"
+
+jobs:
+  build_wheels:
+    name: Build wheels on ${{ matrix.os }}
+    runs-on: ${{ matrix.os }}
+    strategy:
+      matrix:
+        os:
+          - ubuntu-24.04
+          - ubuntu-24.04-arm
+    steps:
+      - uses: actions/checkout@v5
+      - name: Install uv
+        uses: astral-sh/setup-uv@v6
+      - name: Build wheels
+        run: make build_wheels
+      - uses: actions/upload-artifact@v4
+        with:
+          name: cibw-wheels-${{ matrix.os }}-${{ strategy.job-index }}
+          path: ./wheelhouse/*.whl
+  
+  build_sdist:
+    name: Build source distribution
+    runs-on: ubuntu-latest
+    steps:
+      - uses: actions/checkout@v5
+      - name: Install uv
+        uses: astral-sh/setup-uv@v6
+      - name: Build sdist
+        run: make build_sdist
+      - uses: actions/upload-artifact@v4
+        with:
+          name: cibw-sdist
+          path: dist/*.tar.gz
+
+  upload_pypi:
+    needs: [build_wheels, build_sdist]
+    runs-on: ubuntu-latest
+    environment: pypi
+    permissions:
+      id-token: write
+    steps:
+      - uses: actions/download-artifact@v5
+        with:
+          # unpacks all CIBW artifacts into dist/
+          pattern: cibw-*
+          path: dist
+          merge-multiple: true
+      - uses: pypa/gh-action-pypi-publish@release/v1
--- a/.gitignore
+++ b/.gitignore
@@ -8,7 +8,8 @@ doc/_build/
 gh-pages/
 build/
 dist/
+wheelhouse/
 .vscode/
 /__pycache__/
-/pyalsaaudio.egg-info/
+*.egg-info/
 *.raw
--- a/CHANGES.md
+++ b/CHANGES.md
@@ -1,7 +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 0.11.0
+- Fixed `Mixer.getvolume()` returning outdated value (#126)
+- Fixed PCM crashing with some sample formats due to buffer size
+  miscalculation
+- Fixed `PCM.read()` ignoring overruns (regression in 0.10.0)
+- Reverted to `PCM.write()` not throwing an exception on playback buffer
+  underrun; instead, return -EPIPE like `PCM.read()` does on overrun (#130)
+- Added `PCM.avail()` and `PCM.polldescriptors_revents()` functions
+- Added `nominal_bits` and `physical_bits` entries to `PCM.info()`'s
+  return value
+- Added Python type hint file, and adjusted documentation accordingly (#58)
+- Improvements to the examples, in particular isine.py (#42)
+
+Contributions by @ossilator and @viinikv.

 # Version 0.10.0
 - assorted improvements (#123 from @ossilator)
--- a/MANIFEST.in
+++ b/MANIFEST.in
@@ -1,5 +1,2 @@
-include *.py
-include CHANGES
-include TODO
-include LICENSE
-recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json  Makefile
+graft doc/
+graft examples/
--- a/16
+++ b/16
@@ -0,0 +1,16 @@
+.PHONY: lint test build_wheels build_sdist build
+
+lint:
+	uv run --group lint pyright; \
+	uv run --group lint stubtest alsaaudio
+
+test:
+	uv run tests/test.py
+
+build_wheels:
+	uvx cibuildwheel==3.1.4
+
+build_sdist:
+	uv build --sdist
+
+build: build_sdist build_wheels
--- a/README.md
+++ b/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
--- a/ci/setup_cibuildwheel.sh
+++ b/ci/setup_cibuildwheel.sh
@@ -0,0 +1,22 @@
+#!/bin/sh
+
+# NOTE: ensure to use the correct package manager depending on the manylinux
+# image used.
+
+if which yum; then
+	yum install -y alsa-lib-devel
+	exit
+fi
+
+if which apk; then
+	apk add alsa-lib-dev
+	exit
+fi
+
+if which apt-get; then
+	apt-get install -y libasound2-dev
+	exit
+fi
+
+echo "Cannot install ALSA headers! No package manager detected"
+exit 1
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -3,7 +3,7 @@

 # You can set these variables from the command line.
 SPHINXOPTS    =
-SPHINXBUILD   = sphinx-build
+SPHINXBUILD   = uv run --group doc sphinx-build
 PAPER         =

 # Internal variables.
@@ -70,9 +70,3 @@ linkcheck:
 	@echo
 	@echo "Link check complete; look for any errors in the above output " \
 	      "or in linkcheck/output.txt."
-
-gh-pages: html
-	cd gh-pages; git pull --rebase; cd ..; cp -r ./html/* gh-pages
-
-publish: gh-pages
-	cd gh-pages; git commit -a; git push; cd ..
--- a/doc/README.md
+++ b/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

--- a/doc/conf.py
+++ b/doc/conf.py
@@ -20,9 +20,8 @@
 # import sys
 # sys.path.insert(0, os.path.abspath('.'))

-import sys
-sys.path.insert(0, '..')
-from setup import pyalsa_version
+from importlib.metadata import version
+pyalsa_version = version("pyalsaaudio")


 # -- General configuration ------------------------------------------------
@@ -67,7 +66,7 @@ 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.
--- a/doc/libalsaaudio.rst
+++ b/doc/libalsaaudio.rst
@@ -10,14 +10,14 @@

 The :mod:`alsaaudio` module defines functions and classes for using ALSA.

-.. 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:**

@@ -34,7 +34,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.

   *New in 0.8*

-.. function:: cards()
+.. function:: cards() -> list[str]

   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`
@@ -46,7 +46,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
   .. function:: card_indexes()
   .. function:: card_name()

-.. function:: mixers(cardindex=-1, device='default')
+.. function:: mixers(cardindex: int = -1, device: str = 'default') -> list[str]

   List the available mixers. The arguments are:

@@ -82,7 +82,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
     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.

@@ -96,18 +96,20 @@ PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
 sound through speakers or a microphone. The PCM constructor takes the
 following arguments:

-.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1)
+.. 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`.

   =========================  ===============
@@ -157,7 +159,15 @@ following arguments:

     **Note:** This should not be used, as it bypasses most of ALSA's configuration.

-   This will construct a PCM object with the given settings.
+   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:*

@@ -179,63 +189,89 @@ following arguments:

 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.

@@ -267,37 +303,23 @@ PCM objects have the following methods:
   Returns a dictionary of supported format codes (integers) keyed by
   their standard ALSA names (strings).

-.. method:: PCM.setchannels(nchannels)
+.. 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.info()
-
-   Returns a dictionary with the PCM object's configured parameters.
-
-   Values are retrieved from the ALSA library if they are available;
-   otherwise they represent those stored by pyalsaaudio, and their keys
-   are prefixed with ' (call value) '.
-
-   *New in 0.9.1*
-
-.. method:: PCM.dumpinfo()
-
-   Dumps the PCM object's configured parameters to stdout.
-
-.. method:: PCM.state()
+.. 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),
@@ -312,12 +334,26 @@ PCM objects have the following methods:

   *New in 0.10*

-.. method:: PCM.read()
+.. 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
@@ -331,7 +367,7 @@ PCM objects have the following methods:
   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
@@ -360,18 +396,18 @@ PCM objects have the following methods:
   in the kernel, and playout will continue afterwards. Make sure that the
   stream is drained before discarding the PCM handle.

-.. method:: PCM.pause([enable=True])
+.. 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()
+.. method:: PCM.drop() -> int

   Stop the stream and drop residual buffered frames.

   *New in 0.9*

-.. method:: PCM.drain()
+.. method:: PCM.drain() -> int

   For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
   and then stop the stream. In :const:`PCM_NORMAL` mode,
@@ -381,44 +417,53 @@ PCM objects have the following methods:

   *New in 0.10*

-.. method:: PCM.close()
+.. 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()
+.. 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 
+   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)*.

@@ -447,11 +492,11 @@ PCM objects have the following methods:

 **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).

@@ -480,12 +525,12 @@ 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.
@@ -495,7 +540,7 @@ 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'``.
@@ -507,20 +552,20 @@ Mixer objects provides access to the ALSA mixer API.

 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:
@@ -532,7 +577,7 @@ 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
   ======================  ================
@@ -540,7 +585,7 @@ 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:
@@ -556,7 +601,7 @@ Mixer objects have the following methods:
   '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.
@@ -582,13 +627,13 @@ 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)
+.. 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=PCM_PLAYBACK, units=VOLUME_UNITS_RAW)
+.. 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
@@ -602,7 +647,7 @@ Mixer objects have the following methods:
   The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
   :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.

-.. method:: Mixer.getvolume(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)
+.. 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.
@@ -615,7 +660,7 @@ Mixer objects have the following methods:
   The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
   :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.

-.. method:: Mixer.setvolume(volume, channel=None, pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)
+.. 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.
@@ -632,14 +677,14 @@ Mixer objects have the following methods:
   The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
   :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.

-.. method:: Mixer.getmute()
+.. 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, [channel])
+.. 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.
@@ -649,14 +694,14 @@ Mixer objects have the following methods:

   This method will fail if the mixer has no playback mute capabilities

-.. method:: Mixer.getrec()
+.. 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, [channel])
+.. 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.
@@ -666,21 +711,21 @@ 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 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.

-.. method:: Mixer.handleevents()
+.. 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()
+.. method:: Mixer.close() -> None

   Closes the Mixer device.

@@ -719,7 +764,7 @@ 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::
@@ -734,12 +779,12 @@ 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::

@@ -787,7 +832,7 @@ 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
@@ -796,7 +841,7 @@ 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
--- a/doc/pyalsaaudio.rst
+++ b/doc/pyalsaaudio.rst
@@ -81,7 +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:  ---   ::
--- a/doc/terminology.rst
+++ b/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
@@ -22,8 +22,8 @@ Sample
   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
@@ -33,7 +33,7 @@ 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.

@@ -85,6 +85,21 @@ Period size
   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.

--- a/examples/isine.py
+++ b/examples/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)
+
--- a/examples/loopback.py
+++ b/examples/loopback.py
@@ -1,5 +1,5 @@
 #!/usr/bin/env python3
-# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4; python-indent: 4 -*-

 import sys
 import select
@@ -7,12 +7,21 @@ import logging
 import re
 import struct
 import subprocess
+import errno
+from enum import Enum, auto
 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

+# wake up at least after *idle_timer_period* seconds
+idle_timer_period = 0.25
+# grace period for opening the device in seconds
+open_grace_period = 0.5
+# close the device after *idle_close_timeout* seconds silence
+idle_close_timeout = 2.0
+
 poll_names = {
 	select.POLLIN: 'POLLIN',
 	select.POLLPRI: 'POLLPRI',
@@ -34,6 +43,10 @@ state_names = {
 	PCM_STATE_SUSPENDED: 'PCM_STATE_SUSPENDED'
 }

+type NamedProcess = tuple[str, subprocess.Popen]
+
+cmd_process : NamedProcess = None
+
 def poll_desc(mask):
 	return '|'.join([poll_names[bit] for bit, name in poll_names.items() if mask & bit])

@@ -57,28 +70,35 @@ class PollDescriptor(object):

 		return cls(name, fd, mask)

+class LoopbackState(Enum):
+	LISTENING = auto()
+	PLAYING = auto()
+	DEVICE_BUSY = auto()
+
 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 = run_after_stop.split(' ')
-		self.run_before_start = run_before_start.split(' ')
-		self.run_after_stop_did_run = False
+		self.run_after_stop = None
+		if run_after_stop:
+			self.run_after_stop = run_after_stop.split(' ')

-		self.waitBeforeOpen = False
-		self.queue = []
+		self.run_before_start = None
+		if run_before_start:
+			self.run_before_start = run_before_start.split(' ')

-		self.period_size = 0
-		self.silent_periods = 0
+		self.state = None
+		self.last_state_change = None
+		self.silence_start = None

 	@staticmethod
 	def compute_energy(data):
@@ -92,43 +112,84 @@ class Loopback(object):
 	@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}')
+			global cmd_process
+			cmd_process = (cmd, subprocess.Popen(cmd))
+
+	@staticmethod
+	def check_command_idle_handler():
+		# an idle handler to watch the process created above
+		if cmd_process:
+			rc = cmd_process[1].poll()
+			if rc:
+				if rc.returncode:
+					logging.warning(f'run {cmd_process[0]}, return code {rc.returncode}')
+				else:
+					logging.info(f'run {cmd_process[0]}, return code {rc.returncode}')
+				cmd_process = None

 	def register(self, reactor):
-		reactor.register_timeout_handler(self.timeout_handler)
+		reactor.register_idle_handler(self.check_command_idle_handler)
+		reactor.register_idle_handler(self.idle_handler)
 		reactor.register(self.capture_pd, self)

 	def start(self):
+		assert self.state == None, "start must only be called once"
 		# start reading data
-		size, data = self.capture.read()
+		size, _ = self.capture.read()
 		if size:
-			self.queue.append(data)
+			logging.warning(f'initial data discarded ({size} bytes)')

-	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
+		self.state = LoopbackState.LISTENING
+
+	def set_state(self, new_state: LoopbackState) -> LoopbackState:
+		'''Implement the Loopback state as a state machine'''
+
+		if self.state == new_state:
+			return self.state
+
+		logging.info(f'{self.state} -> {new_state}')
+		self.last_state_change = datetime.now()
+
+		if new_state == LoopbackState.LISTENING:
+			if self.state == LoopbackState.PLAYING:
+				self.playback.close()
+				self.playback = None
+				self.last_capture_event = None
+				if self.volume_handler:
+					self.volume_handler.stop()
+				self.run_command(self.run_after_stop)
+			elif self.state == LoopbackState.DEVICE_BUSY:
+				pass
+		elif new_state == LoopbackState.PLAYING:
+			if self.state == LoopbackState.LISTENING:
+				try:
+					self.run_command(self.run_before_start)
+					self.playback = PCM(**self.playback_args)
+					period_size = self.playback.info()['period_size']
+					logging.info(f'opened playback device with period_size {period_size}')
 					if self.volume_handler:
-						self.volume_handler.stop()
-					self.run_command(self.run_after_stop)
+						self.volume_handler.start()
+				except ALSAAudioError as e:
+					logging.warning('opening PCM playback device failed: %s', e)
+					self.state = LoopbackState.DEVICE_BUSY
+					return self.state
+			elif self.state == LoopbackState.DEVICE_BUSY:
+				# Only try to reopen the device after the grace period
+				if datetime.now() - self.last_state_change > timedelta(seconds=open_grace_period):
+					return self.set_state(LoopbackState.PLAYING)
+		elif new_state == LoopbackState.DEVICE_BUSY:
+			logging.error(f'{new_state} is internal and cannot be set directly')
+
+		self.state = new_state
+		return self.state
+
+	def idle_handler(self):
+		if self.state == LoopbackState.PLAYING:
+			if datetime.now() - self.last_capture_event > timedelta(seconds=idle_close_timeout):
+				logging.info('timeout - closing playback device')
+				self.set_state(LoopbackState.LISTENING)
 			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()
@@ -136,6 +197,15 @@ class Loopback(object):
 			return None

 	def handle_capture_event(self, eventmask, name):
+
+		if eventmask & select.POLLERR == select.POLLERR:
+			# This is typically an underrun caused by the external command being run synchronously
+   			# (on the same thread)
+			state = self.capture.state()
+			if state == PCM_STATE_XRUN:
+				self.capture.drop()
+				logging.warning(f'POLLERR for capture device: {state_names[state]}')
+
 		'''called when data is available for reading'''
 		self.last_capture_event = datetime.now()
 		size, data = self.capture.read()
@@ -143,65 +213,37 @@ class Loopback(object):
 			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
+		# compute the energy and go back to LISTENING if nothing is captured
+		energy = self.compute_energy(data)
 		if energy == 0:
-			self.silent_periods = self.silent_periods + 1
+			if self.silence_start is None:
+				self.silence_start = datetime.now()

-			# 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
+			# turn off playback after idle_close_timeout when there was only silence
+			if datetime.now() - self.silence_start > timedelta(seconds=idle_close_timeout):
+				logging.debug('silence')
+				self.set_state(LoopbackState.LISTENING)
+				return False
 		else:
-			self.silent_periods = 0
+			self.silence_start = None

-		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)}')
+		loop_state = self.set_state(LoopbackState.PLAYING)
+		if loop_state != LoopbackState.PLAYING:
+			logging.warning(f'setting state PLAYING failed: {str(loop_state)}')
 			return False

-		try:
-			data = self.pop()
-			if data:
-				space = self.playback.avail()
+		if data:
+			space = self.playback.avail()
+			if space > 0:
 				written = self.playback.write(data)
-				logging.debug(f'wrote {written} bytes while space was {space}')
-		except ALSAAudioError:
-			logging.error('underrun', exc_info=1)
+				if written == -errno.EPIPE:
+					logging.warning('playback underrun')
+					self.playback.write(data)
+				silence = ''
+				if energy == 0:
+					silence = '(silence)'
+				logging.debug(f'wrote {written} bytes while space was {space} {silence}')

 		return True

@@ -229,15 +271,18 @@ class VolumeForwarder(object):
 		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:
+			logging.info(f'start volume is {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]
+		logging.info(f'stop volume is {self.volume}')

 	def __call__(self, fd, eventmask, name):
 		if not self.active:
@@ -257,7 +302,7 @@ class Reactor(object):
 	def __init__(self):
 		self.poll = select.poll()
 		self.descriptors = {}
-		self.timeout_handlers = set()
+		self.idle_handlers = set()

 	def register(self, polldescriptor, callable):
 		logging.debug(f'registered {polldescriptor.name}: {poll_desc(polldescriptor.mask)}')
@@ -268,17 +313,17 @@ class Reactor(object):
 		self.poll.unregister(polldescriptor.fd)
 		del self.descriptors[polldescriptor.fd]

-	def register_timeout_handler(self, callable):
-		self.timeout_handlers.add(callable)
+	def register_idle_handler(self, callable):
+		self.idle_handlers.add(callable)

-	def unregister_timeout_handler(self, callable):
-		self.timeout_handlers.remove(callable)
+	def unregister_idle_handler(self, callable):
+		self.idle_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)
+			events = self.poll.poll(idle_timer_period)
 			for fd, ev in events:
 				polldescriptor, handler = self.descriptors[fd]

@@ -287,8 +332,8 @@ class Reactor(object):

 				handler(fd, ev, polldescriptor.name)

-			if datetime.now() - last_timeout_ev > timedelta(seconds=0.25):
-				for t in self.timeout_handlers:
+			if datetime.now() - last_timeout_ev > timedelta(seconds=idle_timer_period):
+				for t in self.idle_handlers:
 					t()
 				last_timeout_ev = datetime.now()

@@ -338,6 +383,16 @@ if __name__ == '__main__':
 		'periods': args.periods
 	}

+	capture_args = {
+		'type': PCM_CAPTURE,
+		'mode': PCM_NONBLOCK,
+		'device': args.input,
+		'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.
@@ -372,8 +427,7 @@ if __name__ == '__main__':
 			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)
+			capture = PCM(**capture_args)
 			input_mixer_card = capture.info()['card_no']

 		if output_mixer_card is None:
--- a/examples/mixertest.py
+++ b/examples/mixertest.py
--- a/examples/play_rusage.py
+++ b/examples/play_rusage.py
--- a/examples/playbacktest.py
+++ b/examples/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()
-        
+
--- a/examples/playwav.py
+++ b/examples/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)


--- a/examples/recordtest.py
+++ b/examples/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)
--- a/pyproject.toml
+++ b/pyproject.toml
@@ -0,0 +1,70 @@
+[project]
+name = "pyalsaaudio"
+version = "0.11.0"
+description = "ALSA bindings"
+readme.text = """
+This package contains wrappers for accessing the ALSA API from Python.
+It is fairly complete for PCM devices and Mixer access.
+"""
+readme.content-type = "text/plain"
+
+license.file = "LICENSE"
+
+# NOTE: license as a table has been deprecated by PEP-639, however support in
+# setuptools has been introduced in v77.0 which is not installable on Python
+# 3.9. Since this project doesn't constraint the minimum python version, we have
+# to stick to the legacy `license` definition for now.
+
+# license="PSF-2.0"
+# license-files=["LICENSE"]
+
+classifiers = [
+	"Development Status :: 4 - Beta",
+	"Intended Audience :: Developers",
+	"Operating System :: POSIX :: Linux",
+	"Programming Language :: Python :: 3",
+	"Topic :: Multimedia :: Sound/Audio",
+	"Topic :: Multimedia :: Sound/Audio :: Mixers",
+	"Topic :: Multimedia :: Sound/Audio :: Players",
+	"Topic :: Multimedia :: Sound/Audio :: Capture/Recording",
+]
+
+[[project.authors]]
+name = "Casper Wilstrup"
+email = "cwi@aves.dk"
+
+[[project.maintainers]]
+name = "Lars Immisch"
+email = "lars@ibp.de"
+
+[project.urls]
+homepage = "http://larsimmisch.github.io/pyalsaaudio/"
+source = "https://github.com/larsimmisch/pyalsaaudio"
+documentation = "https://larsimmisch.github.io/pyalsaaudio/"
+issues = "https://github.com/larsimmisch/pyalsaaudio/issues"
+
+
+[build-system]
+requires = ["setuptools"]
+build-backend = "setuptools.build_meta"
+
+
+[tool.setuptools]
+ext-modules = [
+	{ name = "alsaaudio", sources = ["src/alsaaudio.c"], libraries = ["asound"] }
+]
+
+
+[tool.cibuildwheel]
+build-frontend = "build[uv]"
+before-all = "ci/setup_cibuildwheel.sh"
+
+
+[dependency-groups]
+lint = [
+	"mypy>=1.17.1",
+	"pyright>=1.1.403",
+]
+doc = [
+	"sphinx>=8.1.3",
+]
--- a/setup.py
+++ b/setup.py
@@ -1,40 +0,0 @@
-#!/usr/bin/env python
-
-'''This package contains wrappers for accessing the ALSA API from Python.
-It is fairly complete for PCM devices and Mixer access.
-'''
-
-from setuptools import setup
-from setuptools.extension import Extension
-from sys import version
-
-pyalsa_version = '0.10.1'
-
-if __name__ == '__main__':
-    setup(
-        name = 'pyalsaaudio',
-        version = pyalsa_version,
-        description = 'ALSA bindings',
-        long_description = __doc__,
-        author = 'Casper Wilstrup',
-        author_email='cwi@aves.dk',
-        maintainer = 'Lars Immisch',
-        maintainer_email = 'lars@ibp.de',
-        license='PSF',
-        platforms=['posix'],
-        url='http://larsimmisch.github.io/pyalsaaudio/',
-        classifiers = [
-            'Development Status :: 4 - Beta',
-            'Intended Audience :: Developers',
-            'License :: OSI Approved :: Python Software Foundation License',
-            'Operating System :: POSIX :: Linux',
-            'Programming Language :: Python :: 2',
-            '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'],
-                               libraries=['asound'])]
-    )
--- a/src/alsaaudio-stubs/init.pyi
+++ b/src/alsaaudio-stubs/init.pyi
@@ -0,0 +1,154 @@
+from typing import Final, final
+
+PCM_PLAYBACK: Final[int]
+PCM_CAPTURE: Final[int]
+
+PCM_NORMAL: Final[int]
+PCM_NONBLOCK: Final[int]
+PCM_ASYNC: Final[int]
+
+PCM_FORMAT_S8: Final[int]
+PCM_FORMAT_U8: Final[int]
+PCM_FORMAT_S16_LE: Final[int]
+PCM_FORMAT_S16_BE: Final[int]
+PCM_FORMAT_U16_LE: Final[int]
+PCM_FORMAT_U16_BE: Final[int]
+PCM_FORMAT_S24_LE: Final[int]
+PCM_FORMAT_S24_BE: Final[int]
+PCM_FORMAT_U24_LE: Final[int]
+PCM_FORMAT_U24_BE: Final[int]
+PCM_FORMAT_S32_LE: Final[int]
+PCM_FORMAT_S32_BE: Final[int]
+PCM_FORMAT_U32_LE: Final[int]
+PCM_FORMAT_U32_BE: Final[int]
+PCM_FORMAT_FLOAT_LE: Final[int]
+PCM_FORMAT_FLOAT_BE: Final[int]
+PCM_FORMAT_FLOAT64_LE: Final[int]
+PCM_FORMAT_FLOAT64_BE: Final[int]
+PCM_FORMAT_MU_LAW: Final[int]
+PCM_FORMAT_A_LAW: Final[int]
+PCM_FORMAT_IMA_ADPCM: Final[int]
+PCM_FORMAT_MPEG: Final[int]
+PCM_FORMAT_GSM: Final[int]
+PCM_FORMAT_S24_3LE: Final[int]
+PCM_FORMAT_S24_3BE: Final[int]
+PCM_FORMAT_U24_3LE: Final[int]
+PCM_FORMAT_U24_3BE: Final[int]
+
+PCM_TSTAMP_NONE: Final[int]
+PCM_TSTAMP_ENABLE: Final[int]
+
+PCM_TSTAMP_TYPE_GETTIMEOFDAY: Final[int]
+PCM_TSTAMP_TYPE_MONOTONIC: Final[int]
+PCM_TSTAMP_TYPE_MONOTONIC_RAW: Final[int]
+
+PCM_FORMAT_DSD_U8: Final[int]
+PCM_FORMAT_DSD_U16_LE: Final[int]
+PCM_FORMAT_DSD_U32_LE: Final[int]
+PCM_FORMAT_DSD_U32_BE: Final[int]
+
+PCM_STATE_OPEN: Final[int]
+PCM_STATE_SETUP: Final[int]
+PCM_STATE_PREPARED: Final[int]
+PCM_STATE_RUNNING: Final[int]
+PCM_STATE_XRUN: Final[int]
+PCM_STATE_DRAINING: Final[int]
+PCM_STATE_PAUSED: Final[int]
+PCM_STATE_SUSPENDED: Final[int]
+PCM_STATE_DISCONNECTED: Final[int]
+
+MIXER_CHANNEL_ALL: Final[int]
+
+# NOTE: Omit for now - use case unknown (see alsaaudio.c)
+#MIXER_SCHN_UNKNOWN: Final[int]
+#MIXER_SCHN_FRONT_LEFT: Final[int]
+#MIXER_SCHN_FRONT_RIGHT: Final[int]
+#MIXER_SCHN_REAR_LEFT: Final[int]
+#MIXER_SCHN_REAR_RIGHT: Final[int]
+#MIXER_SCHN_FRONT_CENTER: Final[int]
+#MIXER_SCHN_WOOFER: Final[int]
+#MIXER_SCHN_SIDE_LEFT: Final[int]
+#MIXER_SCHN_SIDE_RIGHT: Final[int]
+#MIXER_SCHN_REAR_CENTER: Final[int]
+#MIXER_SCHN_MONO: Final[int]
+
+VOLUME_UNITS_PERCENTAGE: Final[int]
+VOLUME_UNITS_RAW: Final[int]
+VOLUME_UNITS_DB: Final[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: ...
+
+def card_indexes() -> list[int]: ...
+def card_name(index: int): ...
+
+_DEPRECATED = ...
+
+@final
+class PCM:
+	def __init__(
+		self,
+		type: int = PCM_PLAYBACK,
+		mode: int = PCM_NORMAL,
+		device: str = "default",
+		cardindex: int = -1,
+		card: str = _DEPRECATED,
+		rate: int = 44100,
+		channels: int = 2,
+		format: int = PCM_FORMAT_S16_LE,
+		periodsize: int = 32,
+		periods: int = 4,
+	) -> None: ...
+	def close(self) -> None: ...
+	def dumpinfo(self) -> None: ...
+	def info(self) -> dict: ...
+	def state(self) -> int: ...
+	def htimestamp(self) -> tuple[int, int, int]: ...
+	def set_tstamp_mode(self, mode: int = PCM_TSTAMP_ENABLE) -> None: ...
+	def get_tstamp_mode(self) -> int: ...
+	def set_tstamp_type(self, type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY) -> None: ...
+	def get_tstamp_type(self) -> int: ...
+	def getformats(self) -> dict: ...
+	def getratebounds(self) -> tuple[int, int]: ...
+	def getrates(self) -> int | tuple[int, int] | list[int]: ...
+	def getchannels(self) -> list[int]: ...
+	def setchannels(self, nchannels: int) -> None: ...
+	def pcmtype(self) -> int: ...
+	def pcmmode(self) -> int: ...
+	def cardname(self) -> str: ...
+	def setrate(self, rate: int) -> None: ...
+	def setformat(self, format: int) -> int: ...
+	def setperiodsize(self, period: int) -> int: ...
+	def read(self) -> tuple[int, bytes]: ...
+	def write(self, data: bytes) -> int: ...
+	def avail(self) -> int: ...
+	def pause(self, enable: bool = True) -> int: ...
+	def drop(self) -> int: ...
+	def drain(self) -> int: ...
+	def polldescriptors(self) -> list[tuple[int, int]]: ...
+	def polldescriptors_revents(self, descriptors: list[tuple[int, int]]) -> int: ...
+
+@final
+class Mixer:
+	def __init__(self, control: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> None: ...
+	def cardname(self) -> str: ...
+	def close(self) -> None: ...
+	def mixer(self) -> str: ...
+	def mixerid(self) -> int: ...
+	def switchcap(self) -> int: ...
+	def volumecap(self) -> int: ...
+	def getvolume(self, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int: ...
+	def getrange(self, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]: ...
+	def getenum(self) -> tuple[str, list[str]]: ...
+	def getmute(self) -> list[int]: ...
+	def getrec(self) -> list[int]: ...
+	def setvolume(self, volume: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None: ...
+	def setenum(self, index: int) -> None: ...
+	def setmute(self, mute: bool, channel: (int | None) = None) -> None: ...
+	def setrec(self, capture: int, channel: (int | None) = None) -> None: ...
+	def polldescriptors(self) -> list[tuple[int, int]]: ...
+	def handleevents(self) -> int: ...
+
+class ALSAAudioError(Exception): ...
--- a/src/alsaaudio.c
+++ b/src/alsaaudio.c
@@ -106,6 +106,9 @@ typedef struct {

 	snd_pcm_t *handle;

+	// Debug logging
+	int state;
+
 	// Configurable parameters
 	unsigned int channels;
 	unsigned int rate;
@@ -360,6 +363,40 @@ alsapcm_list(PyObject *self, PyObject *args, PyObject *kwds)
 	return result;
 }

+const char* state_name(int state) {
+	switch (state) {
+		case SND_PCM_STATE_OPEN:
+			return "SND_PCM_STATE_OPEN";
+		case SND_PCM_STATE_SETUP:
+			return "SND_PCM_STATE_SETUP";
+		case SND_PCM_STATE_PREPARED:
+			return "SND_PCM_STATE_PREPARED";
+		case SND_PCM_STATE_RUNNING:
+			return "SND_PCM_STATE_RUNNING";
+		case SND_PCM_STATE_XRUN:
+			return "SND_PCM_STATE_XRUN";
+		case SND_PCM_STATE_DISCONNECTED:
+			return "SND_PCM_STATE_DISCONNECTED";
+		case SND_PCM_STATE_DRAINING:
+			return "SND_PCM_STATE_DRAINING";
+		case SND_PCM_STATE_PAUSED:
+			return "SND_PCM_STATE_PAUSED";
+		case SND_PCM_STATE_SUSPENDED:
+			return "SND_PCM_STATE_SUSPENDED";
+		default:
+			return "invalid PCM state";
+	}
+}
+
+void print_state(alsapcm_t *self)
+{
+	int state = snd_pcm_state(self->handle);
+	if (state != self->state) {
+		printf("[%s %s] %s->%s\n", self->cardname, self->pcmtype == SND_PCM_STREAM_CAPTURE ? "capture" : "playback", state_name(self->state), state_name(state));
+		self->state = state;
+	}
+}
+
 static int alsapcm_setup(alsapcm_t *self)
 {
 	int res,dir;
@@ -403,7 +440,9 @@ static int alsapcm_setup(alsapcm_t *self)
 	snd_pcm_hw_params_get_period_size(hwparams, &self->periodsize, &dir);
 	snd_pcm_hw_params_get_periods(hwparams, &self->periods, &dir);

-	self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8;
+	self->framesize = self->channels * snd_pcm_format_physical_width(self->format)/8;
+
+	self->state = snd_pcm_state(self->handle);

 	return res;
 }
@@ -598,6 +637,12 @@ alsapcm_dumpinfo(alsapcm_t *self, PyObject *args)
 	val = snd_pcm_hw_params_get_sbits(hwparams);
 	printf("significant bits = %d\n", val);

+	val = snd_pcm_format_width(self->format);
+	printf("nominal bits = %d\n", val);
+
+	val = snd_pcm_format_physical_width(self->format);
+	printf("physical bits = %d\n", val);
+
 	val = snd_pcm_hw_params_is_batch(hwparams);
 	printf("is batch = %d\n", val);

@@ -778,6 +823,16 @@ alsapcm_info(alsapcm_t *self, PyObject *args)
 	PyDict_SetItemString(info,"significant_bits", value);
 	Py_DECREF(value);

+	val = snd_pcm_format_width(self->format);
+	value=PyLong_FromUnsignedLong((unsigned long) val);
+	PyDict_SetItemString(info,"nominal_bits", value);
+	Py_DECREF(value);
+
+	val = snd_pcm_format_physical_width(self->format);
+	value=PyLong_FromUnsignedLong((unsigned long) val);
+	PyDict_SetItemString(info,"physical_bits", value);
+	Py_DECREF(value);
+
 	val = snd_pcm_hw_params_is_batch(hwparams);
 	value=PyBool_FromLong((unsigned long) val);
 	PyDict_SetItemString(info,"is_batch", value);
@@ -1154,6 +1209,37 @@ alsapcm_getchannels(alsapcm_t *self,PyObject *args)
 	return out;
 }

+static PyObject *
+alsapcm_setchannels(alsapcm_t *self, PyObject *args)
+{
+	int channels, saved;
+	int res;
+
+	if (!PyArg_ParseTuple(args,"i:setchannels", &channels))
+		return NULL;
+
+	if (!self->handle) {
+		PyErr_SetString(ALSAAudioError, "PCM device is closed");
+		return NULL;
+	}
+
+	PyErr_WarnEx(PyExc_DeprecationWarning,
+				 "This function is deprecated. "
+				 "Please use the named parameter `channels` to `PCM()` instead", 1);
+
+	saved = self->channels;
+	self->channels = channels;
+	res = alsapcm_setup(self);
+	if (res < 0)
+	{
+		self->channels = saved;
+		PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
+					 self->cardname);
+		return NULL;
+	}
+	return PyLong_FromLong(self->channels);
+}
+
 static PyObject *
 alsapcm_pcmtype(alsapcm_t *self, PyObject *args)
 {
@@ -1196,37 +1282,6 @@ alsapcm_cardname(alsapcm_t *self, PyObject *args)
 	return PyUnicode_FromString(self->cardname);
 }

-static PyObject *
-alsapcm_setchannels(alsapcm_t *self, PyObject *args)
-{
-	int channels, saved;
-	int res;
-
-	if (!PyArg_ParseTuple(args,"i:setchannels", &channels))
-		return NULL;
-
-	if (!self->handle) {
-		PyErr_SetString(ALSAAudioError, "PCM device is closed");
-		return NULL;
-	}
-
-	PyErr_WarnEx(PyExc_DeprecationWarning,
-				 "This function is deprecated. "
-				 "Please use the named parameter `channels` to `PCM()` instead", 1);
-
-	saved = self->channels;
-	self->channels = channels;
-	res = alsapcm_setup(self);
-	if (res < 0)
-	{
-		self->channels = saved;
-		PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
-					 self->cardname);
-		return NULL;
-	}
-	return PyLong_FromLong(self->channels);
-}
-
 static PyObject *
 alsapcm_setrate(alsapcm_t *self, PyObject *args)
 {
@@ -1331,7 +1386,6 @@ static PyObject *
 alsapcm_read(alsapcm_t *self, PyObject *args)
 {
 	snd_pcm_state_t state;
-	int res;
 	int size = self->framesize * self->periodsize;
 	int sizeout = 0;
 	PyObject *buffer_obj, *tuple_obj, *res_obj;
@@ -1364,11 +1418,19 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
 	buffer = PyBytes_AS_STRING(buffer_obj);
 #endif

+	int res = 0;
+
+	print_state(self);
+
 	// After drop() and drain(), we need to prepare the stream again.
 	// Note that fresh streams are already prepared by snd_pcm_hw_params().
 	state = snd_pcm_state(self->handle);
-	if ((state != SND_PCM_STATE_SETUP) ||
-		!(res = snd_pcm_prepare(self->handle))) {
+	if (state == SND_PCM_STATE_SETUP) {
+		res = snd_pcm_prepare(self->handle);
+		printf("[%s] called snd_pcm_prepare: %d\n", self->cardname, res);
+	}
+
+	if (res == 0) {

 		Py_BEGIN_ALLOW_THREADS
 		res = snd_pcm_readi(self->handle, buffer, self->periodsize);
@@ -1472,6 +1534,8 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
 		return NULL;
 	}

+	print_state(self);
+
 	int res;
 	// After drop() and drain(), we need to prepare the stream again.
 	// Note that fresh streams are already prepared by snd_pcm_hw_params().
@@ -1561,6 +1625,9 @@ static PyObject *alsapcm_pause(alsapcm_t *self, PyObject *args)

 		return NULL;
 	}
+
+	print_state(self);
+
 	return PyLong_FromLong(res);
 }

@@ -1583,6 +1650,8 @@ static PyObject *alsapcm_drop(alsapcm_t *self)
 		return NULL;
 	}

+	print_state(self);
+
 	return PyLong_FromLong(res);
 }

@@ -1607,6 +1676,8 @@ static PyObject *alsapcm_drain(alsapcm_t *self)
 		return NULL;
 	}

+	print_state(self);
+
 	return PyLong_FromLong(res);
 }

@@ -1734,6 +1805,7 @@ static PyMethodDef alsapcm_methods[] = {
 	{"pcmtype", (PyCFunction)alsapcm_pcmtype, METH_VARARGS},
 	{"pcmmode", (PyCFunction)alsapcm_pcmmode, METH_VARARGS},
 	{"cardname", (PyCFunction)alsapcm_cardname, METH_VARARGS},
+	{"getchannels", (PyCFunction)alsapcm_getchannels, METH_VARARGS},
 	{"setchannels", (PyCFunction)alsapcm_setchannels, METH_VARARGS},
 	{"setrate", (PyCFunction)alsapcm_setrate, METH_VARARGS},
 	{"setformat", (PyCFunction)alsapcm_setformat, METH_VARARGS},
@@ -1749,7 +1821,6 @@ static PyMethodDef alsapcm_methods[] = {
 	{"getformats", (PyCFunction)alsapcm_getformats, METH_VARARGS},
 	{"getratebounds", (PyCFunction)alsapcm_getratemaxmin, METH_VARARGS},
 	{"getrates", (PyCFunction)alsapcm_getrates, METH_VARARGS},
-	{"getchannels", (PyCFunction)alsapcm_getchannels, METH_VARARGS},
 	{"read", (PyCFunction)alsapcm_read, METH_VARARGS},
 	{"write", (PyCFunction)alsapcm_write, METH_VARARGS},
 	{"avail", (PyCFunction)alsapcm_avail, METH_VARARGS},
--- a/tests/test.py
+++ b/tests/test.py
@@ -13,14 +13,16 @@ import alsaaudio
 import warnings
 from contextlib import closing

+MethodsType = list[tuple[str, tuple[int, ...] | None]]
+
 # we can't test read and write well - these are tested otherwise
-PCMMethods = [
+PCMMethods: MethodsType = [
 	('pcmtype', None),
 	('pcmmode', None),
 	('cardname', None)
 ]

-PCMDeprecatedMethods = [
+PCMDeprecatedMethods: MethodsType = [
 	('setchannels', (2,)),
 	('setrate', (44100,)),
 	('setformat', (alsaaudio.PCM_FORMAT_S8,)),
@@ -30,20 +32,21 @@ PCMDeprecatedMethods = [
 # A clever test would look at the Mixer capabilities and selectively run the
 # omitted tests, but I am too tired for that.

-MixerMethods = [('cardname', None),
-				('mixer', None),
-				('mixerid', None),
-				('switchcap', None),
-				('volumecap', None),
-				('getvolume', None),
-				('getrange', None),
-				('getenum', None),
-#				('getmute', None),
-#				('getrec', None),
-#				('setvolume', (60,)),
-#				('setmute', (0,))
-#				('setrec', (0')),
-				]
+MixerMethods: MethodsType = [
+	('cardname', None),
+	('mixer', None),
+	('mixerid', None),
+	('switchcap', None),
+	('volumecap', None),
+	('getvolume', None),
+	('getrange', None),
+	('getenum', None),
+#	('getmute', None),
+#	('getrec', None),
+#	('setvolume', (60,)),
+#	('setmute', (0,))
+#	('setrec', (0')),
+]

 class MixerTest(unittest.TestCase):
 	"""Test Mixer objects"""
@@ -162,25 +165,25 @@ class PollDescriptorArgsTest(unittest.TestCase):
 	def testArgsNoList(self):
 		with closing(alsaaudio.PCM()) as pcm:
 			with self.assertRaises(TypeError):
-				pcm.polldescriptors_revents('foo')
+				pcm.polldescriptors_revents('foo') # pyright: ignore[reportArgumentType]

 	def testArgsListButNoTuples(self):
 		with closing(alsaaudio.PCM()) as pcm:
 			with self.assertRaises(TypeError):
-				pcm.polldescriptors_revents(['foo', 1])
+				pcm.polldescriptors_revents(['foo', 1])  # pyright: ignore[reportArgumentType]

 	def testArgsListButInvalidTuples(self):
 		with closing(alsaaudio.PCM()) as pcm:
 			with self.assertRaises(TypeError):
-				pcm.polldescriptors_revents([('foo', 'bar')])
+				pcm.polldescriptors_revents([('foo', 'bar')]) # pyright: ignore[reportArgumentType]

 	def testArgsListTupleWrongLength(self):
 		with closing(alsaaudio.PCM()) as pcm:
 			with self.assertRaises(TypeError):
-				pcm.polldescriptors_revents([(1, )])
+				pcm.polldescriptors_revents([(1, )]) # pyright: ignore[reportArgumentType]

 			with self.assertRaises(TypeError):
-				pcm.polldescriptors_revents([(1, 2, 3)])
+				pcm.polldescriptors_revents([(1, 2, 3)]) # pyright: ignore[reportArgumentType]


 if __name__ == '__main__':
--- a/uv.lock
+++ b/uv.lock
@@ -0,0 +1,534 @@
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Author	SHA1	Message	Date
Lars Immisch	a34bdd7a02	Debugging underruns, needs cleanup	2025-12-01 22:33:30 +00:00
Lars Immisch	5a4111dd8c	Small cleanup	2025-12-01 22:33:30 +00:00
Lars Immisch	2a8f2a6699	Cleanup	2025-12-01 22:33:30 +00:00
Lars Immisch	a4168d7525	The capture device does not need to be reopened; better logging/comments	2025-12-01 22:33:30 +00:00
Lars Immisch	ac97e641bc	Restarting the capture device looks ok	2025-12-01 22:33:30 +00:00
Lars Immisch	caafa8ae21	Make commands optional	2025-12-01 22:33:30 +00:00
Lars Immisch	2fae2c0ed7	Experment with state machine	2025-12-01 22:33:29 +00:00
Lars Immisch	acc6f152a3	Adressing TODO in release.yml Initial test with test.pypi.org was successful	2025-12-01 23:26:37 +01:00
Matteo Bernardini	0d6e0e7bb4	chore: use makefile for reference on building sdist and wheels	2025-12-01 23:26:37 +01:00
Matteo Bernardini	4f3c299e53	fix: drop py2 trove classifier	2025-12-01 23:26:37 +01:00
Matteo Bernardini	e78c145d9e	ci: deploy on tags only	2025-12-01 23:26:37 +01:00
Matteo Bernardini	6acae81487	ci: fix permissions	2025-12-01 23:26:37 +01:00
Matteo Bernardini	5eed8adfe5	ci: install ALSA headers	2025-12-01 23:26:37 +01:00
Matteo Bernardini	e3e51cc34d	ci: install `uv`	2025-12-01 23:26:37 +01:00
Matteo Bernardini	f6f53ce92a	ci: automatically build docs	2025-12-01 23:26:37 +01:00
Matteo Bernardini	38d0882379	chore: split dependency groups	2025-12-01 23:26:37 +01:00
Matteo Bernardini	6822cffcc8	ci: remove unsupported setting	2025-12-01 23:26:37 +01:00
Matteo Bernardini	8c1e370e04	ci: auto-detect package manager to use	2025-12-01 23:26:37 +01:00
Matteo Bernardini	6d17bc92b7	ci: use legacy `license` definition to allow building on py3.9	2025-12-01 23:26:37 +01:00
Matteo Bernardini	3e41a9eb97	ci: enforce `setuptools` version	2025-12-01 23:26:37 +01:00
Matteo Bernardini	df8886dde1	ci: ensure alsa headers are installed	2025-12-01 23:26:37 +01:00
Matteo Bernardini	aac2843f73	chore: stub workflow for automatic docs deployment	2025-12-01 23:26:37 +01:00
Matteo Bernardini	7987986ee4	chore: top-level makefile for common dev tasks	2025-12-01 23:26:37 +01:00
Matteo Bernardini	ae93ddc48e	ci: workflow to automate releases on tags	2025-12-01 23:26:37 +01:00
Matteo Bernardini	a48389b750	chore: make `pyright` happy on tests	2025-12-01 23:26:37 +01:00
Matteo Bernardini	0b8db5f46d	fix: `PCM` constructor stubs not coherent with runtime	2025-12-01 23:26:37 +01:00
Matteo Bernardini	6b2c4c22db	fix(docs): add `sphinx` dep, fix version retrieval	2025-12-01 23:26:37 +01:00
Matteo Bernardini	a7896ee069	fix: stub consistency to runtime	2025-12-01 23:26:37 +01:00
Matteo Bernardini	ed83b3e29b	chore(deps): add `mypy` (for `stubtest` utility)	2025-12-01 23:26:37 +01:00
Matteo Bernardini	e96f550862	fix: use `Final` for enum constants	2025-12-01 23:26:37 +01:00
Matteo Bernardini	40f4647d8c	fix: missing `self` in type stubs	2025-12-01 23:26:37 +01:00
Matteo Bernardini	3a8c53851d	chore(deps): add `pyright` for type-checking	2025-12-01 23:26:37 +01:00
Matteo Bernardini	f6770e3a42	fix: add `ALSAAudioError` to type stubs	2025-12-01 23:26:37 +01:00
Matteo Bernardini	0ac28e9eea	fix: include doc/ and examples/ in sdist	2025-12-01 23:26:37 +01:00
Matteo Bernardini	fdc5f3782e	chore: modernise packaging. - use pyproject.toml to define project metadata and setuptools-specific configuration, removing the need for a setup.py file - move sources in conventional src/ directory - rework stubs so that they're visible downstream. PEP-561 doesn't support top-level .pyi files, so I made a dummy package instead `alsaaudio-stubs`	2025-12-01 23:26:37 +01:00
Matteo Bernardini	6efef83429	chore: move unit tests to conventional directory	2025-12-01 23:26:37 +01:00
Matteo Bernardini	df5c2f4685	chore: move examples to dedicated directory	2025-12-01 23:26:37 +01:00
Lars Immisch	0b3f1f41c7	Prepare 0.11.0 Bump version and update changelog.	2024-05-30 23:15:27 +02:00
Lars Immisch	3e360b1bb7	Update MANIFEST.in for type hints	2024-05-14 20:56:51 +01:00
Lars Immisch	44ccbf839d	Fix sphinx warning	2024-05-07 19:07:47 +02:00
Lars Immisch	2c2e43d3d1	Add type hints & docs	2024-05-07 19:07:47 +02:00
Lars Immisch	a142b70033	Reorder for consistency between alsapcm_methods and code getchannels/setchannels should be together IMO	2024-05-07 19:07:47 +02:00
Lars Immisch	26ba938e04	Make commands optional	2024-04-19 18:32:38 +01:00
Lars Immisch	f5e9d52c74	Add missing attribute	2024-04-19 13:10:39 +01:00
Ville Viinikka	436c31f9fd	Add nominal_bits and physical_bits info Adds the information discussed in pull request #144	2024-03-13 10:59:11 +01:00
Ville Viinikka	eda913b203	Use correct sample bit width snd_pcm_hw_params_get_sbits gives the number of significant bits, not the actual number of bits stored. Change to snd_pcm_format_physical_width. This fixes a bug where, for example on my hardware: format = 'S32_LE' significant bits = 24 physical bits = 32 the program will segfault because the allocated buffer is too small.	2024-02-20 18:31:00 +01:00
Lars Immisch	0aba948277	Whitespace cleanup. I ended up using Visual Studio Code and did a global regex replace ` +\n` -> `\n` (StackOverflow)	2024-02-20 18:08:09 +01:00
Oswald Buddenhagen	9b7b767594	fix docu typo	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	db87f2ced5	document new avail() and polldescriptors_revents() PCM functions amends `43a94b3` and `5221311`.	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	f179db2d9b	de-duplicate PCM.info() documentation ... and move the dumpinfo() docu. amends `4e098da` - clearly, i'm blind.	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	420b538321	improve documentation of PCM c'tor and info() method reformulate and redistribute the information, somewhat inspired by text provided by Ronald van Elburg in response to issue #110.	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	ae5c4aad9b	add xrun handling to the examples it's very primitive, but it shows adequately what can happen and what to do about it minimally (that is, complain and move on).	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	d23b26b2e5	isine example: fix stereo handling (#42 ) while it's usually not actually necessary to generate a stereo signal (alsa's default plughw device will happily duplicate it for us), we still do it for demo purposes, just because. a more realistic demo would actually use numpy, as that's what the library will most likely be used with, but anyway.	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	1d63226e56	isine example: simplify calculations in generate() i found them a tad hard to follow ...	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	664f81a777	isine example: simplify thread run loop avoid code duplication.	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	eb51d11619	isine example: actually play some tones the thread in the background actually needs time to do something sensible. this is most easily achieved by simply sleeping in the foreground thread. i addition to the 440 Hz tone, also play 1 kHz, to demonstrate how the change() function is used.	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	2f74e8e8a4	isine example: fix use of deprecated Thread.setDaemon()	2024-02-06 00:39:15 +01:00
Oswald Buddenhagen	6f52de9da0	isine example: remove questionable setting of period size there is no need to be pedantic about the period size, especially with a blocking device. what's more, attempting to set it on an already playing device would error out, and it would be rather counter- productive to temporarily stop it.	2024-02-06 00:39:15 +01:00