Merge tag '0.10.0'

the tag was erroneously "orphaned". fixes #140. Conflicts: CHANGES.md
Ignore volume events if shairplay-sync is running
2024-01-31 20:35:28 +01:00 · 2023-11-18 15:41:30 +00:00 · 2023-10-24 19:17:58 +01:00 · 2023-09-12 21:04:02 +02:00 · 2023-09-04 13:47:20 +01:00 · 2023-09-04 13:45:57 +01:00
27 changed files with 337 additions and 1397 deletions
--- a/.github/workflows/docs.yml
+++ b/.github/workflows/docs.yml
@@ -1,40 +0,0 @@
-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
@@ -1,55 +0,0 @@
-# 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
@@ -8,8 +8,7 @@ doc/_build/
 gh-pages/
 build/
 dist/
-wheelhouse/
 .vscode/
 /__pycache__/
-*.egg-info/
+/pyalsaaudio.egg-info/
 *.raw
--- a/CHANGES.md
+++ b/CHANGES.md
@@ -1,17 +1,6 @@
-# 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 0.10.1
+- restore previous xrun behaviour, #131
+- type hints

 # Version 0.10.0
 - assorted improvements (#123 from @ossilator)
--- a/MANIFEST.in
+++ b/MANIFEST.in
@@ -1,2 +1,5 @@
-graft doc/
-graft examples/
+include *.py
+include CHANGES
+include TODO
+include LICENSE
+recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json  Makefile
--- a/16
+++ b/16
@@ -1,16 +0,0 @@
-.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
@@ -45,10 +45,15 @@ First, get the sources and change to the source directory:
  $ git clone https://github.com/larsimmisch/pyalsaaudio.git
  $ cd pyalsaaudio
 ```
-
-Then, build and install:
+  
+Then, build:
 ```
-  $ pip install .
+  $ python setup.py build
+```
+  
+And install:
+```
+  $ sudo python setup.py install
 ```

 # Using the API
--- a/src/alsaaudio.c
+++ b/src/alsaaudio.c
@@ -403,7 +403,7 @@ 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_format_physical_width(self->format)/8;
+	self->framesize = self->channels * snd_pcm_hw_params_get_sbits(hwparams)/8;

 	return res;
 }
@@ -598,12 +598,6 @@ 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);

@@ -784,16 +778,6 @@ 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);
@@ -1170,37 +1154,6 @@ 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)
 {
@@ -1243,6 +1196,37 @@ 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)
 {
@@ -1380,23 +1364,12 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
 	buffer = PyBytes_AS_STRING(buffer_obj);
 #endif

-	// 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_XRUN && state != SND_PCM_STATE_SETUP) ||
+		(res = snd_pcm_prepare(self->handle)) >= 0) {
 		Py_BEGIN_ALLOW_THREADS
 		res = snd_pcm_readi(self->handle, buffer, self->periodsize);
 		Py_END_ALLOW_THREADS
-
-		if (res == -EPIPE) {
-			// This means buffer overrun, which we need to report.
-			// However, we recover the stream, so the next PCM.read() will work
-			// again. If recovery fails (very unlikely), report that instead.
-			if (!(res = snd_pcm_prepare(self->handle)))
-				res = -EPIPE;
-		}
 	}

 	if (res != -EPIPE)
@@ -1412,118 +1385,10 @@ alsapcm_read(alsapcm_t *self, PyObject *args)
 			Py_DECREF(buffer_obj);
 			return NULL;
 		}
-		else
-		{
-			sizeout = res * self->framesize;
-		}
 	}

-	if (size != sizeout) {
-#if PY_MAJOR_VERSION < 3
-		/* If the following fails, it will free the object */
-		if (_PyString_Resize(&buffer_obj, sizeout))
-			return NULL;
-#else
-		/* If the following fails, it will free the object */
-		if (_PyBytes_Resize(&buffer_obj, sizeout))
-			return NULL;
-#endif
-	}
-
-	res_obj = PyLong_FromLong(res);
-	if (!res_obj) {
-		Py_DECREF(buffer_obj);
-		return NULL;
-	}
-	tuple_obj = PyTuple_New(2);
-	if (!tuple_obj) {
-		Py_DECREF(buffer_obj);
-		Py_DECREF(res_obj);
-		return NULL;
-	}
-	/* Steal reference counts */
-	PyTuple_SET_ITEM(tuple_obj, 0, res_obj);
-	PyTuple_SET_ITEM(tuple_obj, 1, buffer_obj);
-
-	return tuple_obj;
-}
-
-static PyObject *
-alsapcm_read_sw(alsapcm_t *self, PyObject *args)
-{
-	snd_pcm_state_t state;
-	int res;
-	int max_frames_to_read;
-	int size;
-	int sizeout = 0;
-	PyObject *buffer_obj, *tuple_obj, *res_obj;
-	char *buffer;
-
-	if (!PyArg_ParseTuple(args,"i:read_sw", &max_frames_to_read))
-		return NULL;
-
-	if (!self->handle) {
-		PyErr_SetString(ALSAAudioError, "PCM device is closed");
-		return NULL;
-  }
-
-	if (self->pcmtype != SND_PCM_STREAM_CAPTURE)
-	{
-		PyErr_Format(ALSAAudioError, "Cannot read from playback PCM [%s]",
-					 self->cardname);
-		return NULL;
-	}
-
-	size = self->framesize * max_frames_to_read;
-
-#if PY_MAJOR_VERSION < 3
-	buffer_obj = PyString_FromStringAndSize(NULL, size);
-	if (!buffer_obj)
-		return NULL;
-	buffer = PyString_AS_STRING(buffer_obj);
-#else
-	buffer_obj = PyBytes_FromStringAndSize(NULL, size);
-	if (!buffer_obj)
-		return NULL;
-	buffer = PyBytes_AS_STRING(buffer_obj);
-#endif
-
-	// 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))) {
-
-		Py_BEGIN_ALLOW_THREADS
-		res = snd_pcm_readi(self->handle, buffer, max_frames_to_read);
-		Py_END_ALLOW_THREADS
-
-		if (res == -EPIPE) {
-			// This means buffer overrun, which we need to report.
-			// However, we recover the stream, so the next PCM.read() will work
-			// again. If recovery fails (very unlikely), report that instead.
-			if (!(res = snd_pcm_prepare(self->handle)))
-				res = -EPIPE;
-		}
-	}
-
-	if (res != -EPIPE)
-	{
-		if (res == -EAGAIN)
-		{
-			res = 0;
-		}
-		else if (res < 0) {
-			PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
-						 self->cardname);
-
-			Py_DECREF(buffer_obj);
-			return NULL;
-		}
-		else
-		{
-			sizeout = res * self->framesize;
-		}
+	if (res > 0 ) {
+		sizeout = res * self->framesize;
 	}

 	if (size != sizeout) {
@@ -1560,6 +1425,7 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
 {
 	int datalen;
 	char *data;
+	PyObject *rc = NULL;

 #if PY_MAJOR_VERSION < 3
 	if (!PyArg_ParseTuple(args,"s#:write", &data, &datalen))
@@ -1597,48 +1463,45 @@ static PyObject *alsapcm_write(alsapcm_t *self, PyObject *args)
 	}

 	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().
 	snd_pcm_state_t state = snd_pcm_state(self->handle);
-	if ((state != SND_PCM_STATE_SETUP) ||
-		!(res = snd_pcm_prepare(self->handle))) {
+
+
+	if ((state != SND_PCM_STATE_XRUN && state != SND_PCM_STATE_SETUP) ||
+		(res = snd_pcm_prepare(self->handle)) >= 0) {

 		Py_BEGIN_ALLOW_THREADS
 		res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
-		Py_END_ALLOW_THREADS
-
 		if (res == -EPIPE) {
-			// This means buffer underrun, which we need to report.
-			// However, we recover the stream, so the next PCM.write() will work
-			// again. If recovery fails (very unlikely), report that instead.
-			if (!(res = snd_pcm_prepare(self->handle)))
-				res = -EPIPE;
+			/* EPIPE means underrun */
+			res = snd_pcm_recover(self->handle, res, 1);
+			if (res >= 0) {
+				res = snd_pcm_writei(self->handle, data, datalen/self->framesize);
+			}
 		}
+		Py_END_ALLOW_THREADS
 	}

-	if (res != -EPIPE)
+	if (res == -EAGAIN) {
+		rc = PyLong_FromLong(0);
+	}
+	res = snd_pcm_prepare(self->handle);
+	if (res < 0)
 	{
-		if (res == -EAGAIN)
-		{
-			res = 0;
-		}
-		else if (res < 0) {
-			PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
-						 self->cardname);
+		PyErr_Format(ALSAAudioError, "%s [%s]", snd_strerror(res),
+					 self->cardname);

 #if PY_MAJOR_VERSION >= 3
-			PyBuffer_Release(&buf);
+		PyBuffer_Release(&buf);
 #endif

-			return NULL;
-		}
+		return NULL;
 	}

 #if PY_MAJOR_VERSION >= 3
 	PyBuffer_Release(&buf);
 #endif

-	return PyLong_FromLong(res);
+	return rc;
 }

 static PyObject *
@@ -1858,7 +1721,6 @@ 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},
@@ -1874,8 +1736,8 @@ 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},
-	{"read_sw", (PyCFunction)alsapcm_read_sw, METH_VARARGS},
 	{"write", (PyCFunction)alsapcm_write, METH_VARARGS},
 	{"avail", (PyCFunction)alsapcm_avail, METH_VARARGS},
 	{"pause", (PyCFunction)alsapcm_pause, METH_VARARGS},
--- a/ci/setup_cibuildwheel.sh
+++ b/ci/setup_cibuildwheel.sh
@@ -1,22 +0,0 @@
-#!/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   = uv run --group doc sphinx-build
+SPHINXBUILD   = sphinx-build
 PAPER         =

 # Internal variables.
@@ -70,3 +70,9 @@ 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
@@ -1,22 +1,32 @@
 # Make a new release

-Create and push a tag naming the version (i.e. 0.11.1):
+Update the version in setup.py

-    git tag 0.11.1
-    git push origin 0.11.1
+    pyalsa_version = '0.9.0'

-This should trigger a build via a github actions and publish pre-built binaries to pypi.org 
+Commit and push the update.
+
+Create and push a tag naming the version (i.e. 0.9.0):
+
+    git tag 0.9.0
+    git push origin 0.9.0
+
+Create the package:
+
+    python3 setup.py sdist
+
+Upload the package
+
+    twine upload dist/*
+
+Don't forget to update the documentation.

 # Publish the documentation

-All commits to main should trigger a rebuild of the documentation.
-
-## Historical background
-
 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
@@ -20,8 +20,9 @@
 # import sys
 # sys.path.insert(0, os.path.abspath('.'))

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


 # -- General configuration ------------------------------------------------
@@ -66,7 +67,7 @@ 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 = 'en'
+language = None

 # 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: int = PCM_PLAYBACK) ->list[str]
+.. function:: pcms(pcmtype=PCM_PLAYBACK)

   List available PCM devices by name.

   Arguments are:

   * *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).
+     (default).  

   **Note:**

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

   *New in 0.8*

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

   List the available ALSA cards by name. This function is only moderately
   useful. If you want to see a list of available PCM devices, use :func:`pcms`
@@ -46,7 +46,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
   .. function:: card_indexes()
   .. function:: card_name()

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

   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() -> str
+.. function:: asoundlib_version()

   Return a Python string containing the ALSA version found.

@@ -96,20 +96,18 @@ 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: 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
+.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, periods=4, device='default', cardindex=-1)

-   This class is used to represent a PCM device (either for playback or
-   recording). The constructor's arguments are:
+   This class is used to represent a PCM device (either for playback and
+   recording). The arguments are:

   * *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).
+     (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`.

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

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

   *Changed in 0.10:*

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

 PCM objects have the following methods:

-.. method:: PCM.info() -> dict
+.. method:: PCM.info()

-   Returns a dictionary containing the configuration of a PCM device.
+   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) 
+   ===========================  =============================  ==================================================================

-   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) "**.
+   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():".

-   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.

-   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
+.. method:: PCM.pcmtype()

   Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
   :const:`PCM_PLAYBACK`.

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

   Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
   :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`

-.. method:: PCM.cardname() -> string
+.. method:: PCM.cardname()

   Return the name of the sound card used by this PCM object.

@@ -303,23 +267,37 @@ 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: int) -> int
+.. method:: PCM.setchannels(nchannels)

   .. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.

-.. method:: PCM.setrate(rate: int) -> int
+.. method:: PCM.setrate(rate)

   .. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.

-.. method:: PCM.setformat(format: int) -> int
+.. method:: PCM.setformat(format)

   .. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.

-.. method:: PCM.setperiodsize(period: int) -> int
+.. method:: PCM.setperiodsize(period)

   .. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.

-.. method:: PCM.state() -> int
+.. method:: PCM.info()
+
+   Returns a dictionary with the PCM object's configured parameters.
+
+   Values are retrieved from the ALSA library if they are available;
+   otherwise they represent those stored by pyalsaaudio, and their keys
+   are prefixed with ' (call value) '.
+
+   *New in 0.9.1*
+
+.. method:: PCM.dumpinfo()
+
+   Dumps the PCM object's configured parameters to stdout.
+
+.. method:: PCM.state()

   Returs the current state of the stream, which can be one of
   :const:`PCM_STATE_OPEN` (this should not actually happen),
@@ -334,80 +312,54 @@ PCM objects have the following methods:

   *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]
+.. method:: PCM.read()

   In :const:`PCM_NORMAL` mode, this function blocks until a full period is
   available, and then returns a tuple (length,data) where *length* is
   the number of frames of captured data, and *data* is the captured
-   sound frames as a string. The length of the returned data will be
+   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 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.
+   In case of an overrun, this function will return a negative size: :const:`-EPIPE`.
+   This indicates that data was lost, even if the operation itself succeeded.
+   Try using a larger periodsize.

-.. method:: PCM.write(data: bytes) -> int
+.. method:: PCM.write(data)

   Writes (plays) the sound in data. The length of data *must* be a
   multiple of the frame size, and *should* be exactly the size of a
   period. If less than 'period size' frames are provided, the actual
   playout will not happen until more data is written.

-   If the 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.
+   If the device is not in :const:`PCM_NONBLOCK` mode, this call will block if
+   the kernel buffer is full, and until enough sound has been played
+   to allow the sound data to be buffered. The call always returns the
+   size of the data provided.

   In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a
   return value of zero, if the buffer is full. In this case, the data
-   should be written 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.
+   should be written at a later time.

   Note that this call completing means only that the samples were buffered
   in the kernel, and playout will continue afterwards. Make sure that the
   stream is drained before discarding the PCM handle.

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

   If *enable* is :const:`True`, playback or capture is paused.
   Otherwise, playback/capture is resumed.

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

   Stop the stream and drop residual buffered frames.

   *New in 0.9*

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

   For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
   and then stop the stream. In :const:`PCM_NORMAL` mode,
@@ -417,53 +369,44 @@ PCM objects have the following methods:

   *New in 0.10*

-.. method:: PCM.close() -> None
+.. method:: PCM.close()

   Closes the PCM device.

   For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
   this function blocks until all pending playback is drained.

-.. method:: PCM.polldescriptors() -> list[tuple[int, int]]
+.. method:: PCM.polldescriptors()

   Returns a list of tuples of *(file descriptor, eventmask)* that can be
   used to wait for changes on the PCM with *select.poll*.

-   The *eventmask* value is compatible with `poll.register`__ in the Python
+   The *eventmask* value is compatible with `poll.register`__ in the Python 
   :const:`select` module.

-.. 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])
+.. method:: PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE])

   Set the ALSA timestamp mode on the device. The mode argument can be set to
   either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.

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

   Return the integer value corresponding to the ALSA timestamp mode. The
   return value can be either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.

-.. method:: PCM.set_tstamp_type([type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY]) -> None
+.. method:: PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY])

   Set the ALSA timestamp mode on the device. The type argument
   can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
   :const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.

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

   Return the integer value corresponding to the ALSA timestamp type. The
   return value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
   :const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.

-.. method:: PCM.htimestamp() -> tuple[int, int, int]
+.. method:: PCM.htimestamp()

   Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*.

@@ -492,11 +435,11 @@ 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, if too much data is
+ugly clicking sounds will occur. Conversely, of too much data is
 written to the device, the write function will either block
 (:const:`PCM_NORMAL` mode) or return zero (:const:`PCM_NONBLOCK` mode).

@@ -525,12 +468,12 @@ Mixer Objects

 Mixer objects provides access to the ALSA mixer API.

-.. class:: Mixer(control: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> Mixer
+.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default')

   Arguments are:

   * *control* - specifies which control to manipulate using this mixer
-     object. The list of available controls can be found with the
+     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.
@@ -540,7 +483,7 @@ 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'``.
@@ -552,20 +495,20 @@ Mixer objects provides access to the ALSA mixer API.

 Mixer objects have the following methods:

-.. method:: Mixer.cardname() -> str
+.. method:: Mixer.cardname()

   Return the name of the sound card used by this Mixer object

-.. method:: Mixer.mixer() -> str
+.. method:: Mixer.mixer()

   Return the name of the specific mixer controlled by this object, For example
   ``'Master'`` or ``'PCM'``

-.. method:: Mixer.mixerid() -> int
+.. method:: Mixer.mixerid()

   Return the ID of the ALSA mixer controlled by this object.

-.. method:: Mixer.switchcap() -> int
+.. method:: Mixer.switchcap()

   Returns a list of the switches which are defined by this specific mixer.
   Possible values in this list are:
@@ -577,7 +520,7 @@ 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
   ======================  ================
@@ -585,7 +528,7 @@ Mixer objects have the following methods:
   To manipulate these switches use the :meth:`setrec` or
   :meth:`setmute` methods

-.. method:: Mixer.volumecap() -> int
+.. method:: Mixer.volumecap()

   Returns a list of the volume control capabilities of this
   mixer. Possible values in the list are:
@@ -601,7 +544,7 @@ Mixer objects have the following methods:
   'Joined Capture Volume'   Manipulate sound capture volume for all channels at a time
   ========================  ================

-.. method:: Mixer.getenum() -> tuple[str, list[str]]
+.. method:: Mixer.getenum()

   For enumerated controls, return the currently selected item and  the list of
   items available.
@@ -627,13 +570,13 @@ 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: int) -> None
+.. method:: Mixer.setenum(index)

   For enumerated controls, sets the currently selected item.
   *index* is an index into the list of available enumerated items returned
   by :func:`getenum`.

-.. method:: Mixer.getrange(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]
+.. method:: Mixer.getrange(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_RAW)

   Return the volume range of the ALSA mixer controlled by this object.
   The value is a tuple of integers whose meaning is determined by the
@@ -647,7 +590,7 @@ 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: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int
+.. method:: Mixer.getvolume(pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)

   Returns a list with the current volume settings for each channel. The list
   elements are integers whose meaning is determined by the *units* argument.
@@ -660,7 +603,7 @@ 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: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None
+.. method:: Mixer.setvolume(volume, channel=None, pcmtype=PCM_PLAYBACK, units=VOLUME_UNITS_PERCENTAGE)

   Change the current volume settings for this mixer. The *volume* argument
   is an integer whose meaning is determined by the *units* argument.
@@ -677,14 +620,14 @@ 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() -> list[int]
+.. method:: Mixer.getmute()

   Return a list indicating the current mute setting for each channel.
   0 means not muted, 1 means muted.

   This method will fail if the mixer has no playback switch capabilities.

-.. method:: Mixer.setmute(mute: bool, channel: (int | None) = None) -> None
+.. method:: Mixer.setmute(mute, [channel])

   Sets the mute flag to a new value. The *mute* argument is either 0 for not
   muted, or 1 for muted.
@@ -694,14 +637,14 @@ 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.getrec()

   Return a list indicating the current record mute setting for each channel.
   0 means not recording, 1 means recording.

   This method will fail if the mixer has no capture switch capabilities.

-.. method:: Mixer.setrec(capture: int, channel: (int | None) = None) -> None
+.. method:: Mixer.setrec(capture, [channel])

   Sets the capture mute flag to a new value. The *capture* argument
   is either 0 for no capture, or 1 for capture.
@@ -711,21 +654,21 @@ Mixer objects have the following methods:

   This method will fail if the mixer has no capture switch capabilities.

-.. method:: Mixer.polldescriptors() -> list[tuple[int, int]]
+.. method:: Mixer.polldescriptors()

   Returns a list of tuples of *(file descriptor, eventmask)* that can be
   used to wait for changes on the mixer with *select.poll*.

-   The *eventmask* value is compatible with `poll.register`__ in the Python
+   The *eventmask* value is compatible with `poll.register`__ in the Python 
   :const:`select` module.

-.. method:: Mixer.handleevents() -> int
+.. method:: Mixer.handleevents()

   Acknowledge events on the :func:`polldescriptors` file descriptors
   to prevent subsequent polls from returning the same events again.
   Returns the number of events that were acknowledged.

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

   Closes the Mixer device.

@@ -764,7 +707,7 @@ 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::
@@ -779,12 +722,12 @@ 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::

@@ -832,7 +775,7 @@ 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
@@ -841,7 +784,7 @@ 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
@@ -81,12 +81,16 @@ 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:  ---   ::

-   $ pip install .
+   $ python setup.py build
+
+And then as root:  ---   ::
+
+   # python setup.py install


 *******
--- 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.

@@ -85,21 +85,6 @@ 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
@@ -7,7 +7,6 @@
 from __future__ import print_function

 import sys
-import time
 from threading import Thread
 from multiprocessing import Queue

@@ -16,15 +15,14 @@ if sys.version_info[0] < 3:
 else:
    from queue import Empty

-from math import pi, sin, ceil
+from math import pi, sin
 import struct
-import itertools
 import alsaaudio

 sampling_rate = 48000

 format = alsaaudio.PCM_FORMAT_S16_LE
-pack_format = 'h'  # short int, matching S16
+framesize = 2 # bytes per frame for the values above
 channels = 2

 def nearest_frequency(frequency):
@@ -36,28 +34,28 @@ 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 the duration
-    factor = int(ceil(duration * frequency))
-
-    # total number of frames
-    size = cycle_size * factor
+    # number of full cycles we can fit into target_size
+    factor = int(target_size / cycle_size)

+    size = max(int(cycle_size * factor), 1)
+    
    sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \
             for i in range(size)]
-
-    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)
-
+             
+    return struct.pack('%dh' % size, *sine)
+                                                                                  

 class SinePlayer(Thread):
-
+    
    def __init__(self, frequency = 440.0):
-        Thread.__init__(self, daemon=True)
+        Thread.__init__(self)
+        self.setDaemon(True)
        self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
        self.queue = Queue()
        self.change(frequency)
@@ -75,23 +73,22 @@ 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:
-                pass
-            if buffer:
-                if self.device.write(buffer) < 0:
-                    print("Playback buffer underrun! Continuing nonetheless ...")
-
+                if buffer:
+                    self.device.write(buffer)
+                

 isine = SinePlayer()
 isine.start()

-time.sleep(1)
-isine.change(1000)
-time.sleep(1)
-
+def change(f):
+    isine.change(f)
+    
--- a/examples/loopback.py
+++ b/examples/loopback.py
@@ -70,13 +70,8 @@ class Loopback(object):
 		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 = run_after_stop.split(' ')
+		self.run_before_start = run_before_start.split(' ')
 		self.run_after_stop_did_run = False

 		self.waitBeforeOpen = False
@@ -234,7 +229,6 @@ 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
--- 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
@@ -47,8 +47,7 @@ if __name__ == '__main__':
    # Read data from stdin
    data = f.read(320)
    while data:
-        if out.write(data) < 0:
-            print("Playback buffer underrun! Continuing nonetheless ...")
+        out.write(data)
        data = f.read(320)
    out.close()
-
+        
--- a/examples/playwav.py
+++ b/examples/playwav.py
@@ -39,8 +39,7 @@ def play(device, f):
 	data = f.readframes(periodsize)
 	while data:
 		# Read data from stdin
-		if device.write(data) < 0:
-			print("Playback buffer underrun! Continuing nonetheless ...")
+		device.write(data)
 		data = f.readframes(periodsize)


--- a/pyproject.toml
+++ b/pyproject.toml
@@ -1,70 +0,0 @@
-[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/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
@@ -59,8 +59,6 @@ if __name__ == '__main__':
 		# Read data from device
 		l, data = inp.read()

-		if l < 0:
-			print("Capture buffer overrun! Continuing nonetheless ...")
-		elif l:
+		if l:
 			f.write(data)
 			time.sleep(.001)
--- a/setup.py
+++ b/setup.py
@@ -0,0 +1,40 @@
+#!/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
@@ -1,154 +0,0 @@
-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/tests/test.py
+++ b/tests/test.py
@@ -13,16 +13,14 @@ 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: MethodsType = [
+PCMMethods = [
 	('pcmtype', None),
 	('pcmmode', None),
 	('cardname', None)
 ]

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

-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')),
-]
+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')),
+				]

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

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

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

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

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


 if __name__ == '__main__':
--- a/uv.lock
+++ b/uv.lock
@@ -1,534 +0,0 @@
-version = 1
-revision = 3
-requires-python = ">=3.10"
-resolution-markers = [
-    "python_full_version >= '3.11'",
-    "python_full_version < '3.11'",
-]
-
-[[package]]
-name = "alabaster"
-version = "1.0.0"
-source = { registry = "https://pypi.org/simple" }
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Author	SHA1	Message	Date
Oswald Buddenhagen	d0625934bb	Merge tag '0.10.0' the tag was erroneously "orphaned". fixes #140. Conflicts: CHANGES.md	2024-01-31 20:35:28 +01:00
Lars Immisch	788fcb7c69	Ignore volume events if shairplay-sync is running	2023-11-18 15:41:30 +00:00
Lars Immisch	345260241f	loopback.py: bugfixes Retroactively add to CHANGES.md	2023-10-24 19:17:58 +01:00
Lars Immisch	67adbf0524	Add a (naive) loopback implementation (#132 ) * WIP * Implement alsapcm_polldescriptors_revents and alspcm_avail * Open/close the playback device when idle. It takes a long time until it's stopped, though. * open/close logic of playback device * Fix opening logic, make period size divisible by 6 * Be less verbose in level info * Extra argument for output mixer card index Sometimes, this cannot be deduced from the output device * Better silence detection * Run run_after_stop when idle on startup	2023-09-12 21:04:02 +02:00
Lars Immisch	cd44517ed5	Improve scope (like in #134 )	2023-09-04 13:47:20 +01:00
Lars Immisch	3743cf5bd5	Py_BEGIN_ALLOW_THREADS was in the wrong scope.	2023-09-04 13:45:57 +01:00
Lars Immisch	f374adb269	Fix issue #133	2023-09-04 13:43:34 +01:00
Lars Immisch	7cc1dadcb2	Restore previous behaviour of calling snd_pcm_prepare in case of XRUN (#131 )	2023-08-31 16:25:42 +02:00
Lars Immisch	d3bb44517d	Small improvement to VolumeForwarder	2023-06-01 15:29:45 +01:00
Lars Immisch	c3dba90ff2	Refactor. SCNR. The Reactor now takes a callable, and the loopback and volume forwarder are now implemented as callable instances, which seemed the most Pythonic solution.	2023-06-01 15:16:42 +01:00
Lars Immisch	e3ec2d1628	Better error logging and comments	2023-05-31 20:00:16 +01:00
Lars Immisch	a66cf02a4c	Add volume control forwarding. This needs the patches from (probably) https://lkml.org/lkml/2021/3/1/419. They are already in the raspberry OS kernel sources and the setup works on a RPi 4	2023-05-30 16:21:53 +01:00
Lars Immisch	d5e81f2604	mixertest.py: print capture volume	2023-05-30 14:26:56 +01:00
Lars Immisch	8f7e8fbdfe	Add a naive loopback implementation using select.poll() It does work, though.	2023-05-30 14:07:40 +01:00
Lars Immisch	dc3d6c6183	Handle events in alsamixer_getvolume. Closes #126 This issue can be worked around by calling mixer.handleevents() before calling mixer.getvolume(), but it makes more sense to handle all events before returning the volume.	2023-05-12 02:43:51 +01:00
Lars Immisch	c6a0c8016d	Update changes for release	2023-04-16 00:05:55 +02:00
Oswald Buddenhagen	196ca87a05	assorted improvements (#123 ) * fix draining/closing, take 2 commit `8abf06be` introduced a pause() prior to draining, in an attempt to work around clearly broken pulseaudio client behavior for capture streams (drain() is supposed to imply a stop). but as the workaround was also applied to playback streams, it would cause nasty "clicks", as the stream would (obviously) stop before being resumed for draining. but draining is actually pointless for capture streams, as we're closing right afterwards, so the samples are lost anyway. what's more, destructors are not supposed to wait for anything, so draining in alsapcm_dealloc() was wrong to start with. so we remove it. note that this is a minor behavior change, which is reflected by the adjustment of the playback test to have an explicit close() at the end. finally, close() was also affected by the pulseaudio bug (which was not addressed before), so there we make draining exclusive to playback streams. * fix memory leaks in _polldescriptors() the calloc'd pollfd arrays were not freed. fix memory handling in mixer access error paths in case of error, alsamixer_new() would leak the object, while alsamixer_list() might crash due to a null pointer. as a drive-by, make alsamixer_gethandle() `static`. * fix crashes when accessing already closed devices PCM.htimestamp() gets the usual exception emission, Mixer.close() gets a "double invocation" check like PCM.close() has. * fix deprecation warning about PyEval_InitThreads() PyEval_InitThreads is a no-op in since python 3.9. * fix deprecation warning about PyUnicode_AsUnicode() converting to ascii for the purpose of comparison is inefficient. * remove redundant snd_pcm_hw_params_any() call we just called it (and even error-checked it) a few lines above. * add new high-speed samples rates closes #89 (but alsa doesn't support 768khz yet). * drop some pointless comments from the tex => sphinx conversion amends `5c2a00655`. * remove bogus markup from the documentation the poll objects are linked properly in a different way, and the footnote appears outdated. * unify line spacing in .rst files one empty line, except for high-level sections, which get two. while at it, trim whitespace on otherwise empty lines. * formatting/language fixes in introduction document * improve terminology document mention xruns, and rework the definition of periods: concentrate on relevant information, and remove the misinformation about period size reduction being not that bad (pedantically, an application could run somewhat asynchronously to the interrupts by using some timer, and therefore actually save some of the overhead, but why would one use a small period size in the first place then?). also, language and formatting fixes. * add missing and update incorrect/outdated documentation for clarity, this includes docs which were previously omitted (presumably) intentionally, but mark them as comments. the getrec() and getmute() functions' docs are moved around, so they appear in pairs with their set() counterparts, like the volume() ones already did. notably, this also fixes the docu of PCM_FORMAT_U8, which closes #104. * add some best practices to the docu addresses #110, among other things. * purge pydoc from the source it's been obsolete for a long time, and having it redundantly to the rst sources is bad hygiene. it still contained some useful info, which has been transplanted to the rst source in the previous commit. * use data types closer to those of ALSA this removes lots of casts around snd_pcm_hw_params_get_() calls we could go further with that to make the code clean if we enabled all the warnings, but it doesn't seem worth the effort. reduce scope of GIL releases it's pointless to enclose snd_pcm_close() and snd_pcm_pause(), as these calls don't sleep. * reshuffle XRUN recovery somewhat perform it prior to invoking read()/write() if necessary, not right after a failure event. this makes things more uniform and predictable. we don't use snd_pcm_recover() any more, as we used it only for the EPIPE case anyway, which boils down to snd_pcm_prepare() exactly. handling ESTRPIPE as well might be desirable, but that's a separate consideration. * bump (minor) version we're about to add new features. * make period count configurable the period count is just as important for playback latency as the period size, so it makes no sense to have only one of them configurable. as a drive-by, fix up the handling of periods in info() & dumpinfo(). * add PCM.drain() for playback, this allows making sure that all written frames are played, without using an external delay. in principle, it's also usable for capture, but there isn't really a practical reason to do so, as simply discarding excess captured frames has no real cost. * add PCM.state() and associated enum values in principle, the state is already available from info(), but that's a rather heavy function for something one might want to query often. a practical use case might be checking whether a playback stream is done draining, for example.	2023-04-15 21:45:32 +02:00