Prepare 0.11.0

Bump version and update changelog.

.gitignore

@@ -4,6 +4,11 @@ MANIFEST
 doc/gh-pages/
 doc/html/
 doc/doctrees/
+doc/_build/
 gh-pages/
 build/
-dist/
+dist/
+.vscode/
+/__pycache__/
+/pyalsaaudio.egg-info/
+*.raw

CHANGES

@@ -1,70 +0,0 @@
-Version 0.8.2:
-- fix #3 (we cannot get the revision from git for pip installs)
-
-Version 0.8.1:
-- document changes (this file)
-
-Version 0.8:
-- 'PCM()' has new 'device' and 'cardindex' keyword arguments.
-
-  The keyword 'device' allows to select virtual devices, 'cardindex' can be
-  used to select hardware cards by index (as with 'mixers()' and 'Mixer()').
-
-  The 'card' keyword argument is still supported, but deprecated.
-
-  The reason for this change is that the 'card' keyword argument guessed
-  a device name from the card name, but this only works sometimes, and breaks
-  opening virtual devices.
-
-- new function 'pcms()' to list available PCM devices.
-
-- mixers() and Mixer() take an additional 'device' keyword argument.
-  This allows to list or open virtual devices.
-
-- The default behaviour of Mixer() without any arguments has changed.
-  Now Mixer() will try to open the 'default' Mixer instead of the Mixer
-  that is associated with card 0.
-
-- fix a memory leak under Python 3.x
-
-- some more memory leaks in error conditions fixed.
-
-Version 0.7:
-- fixed several memory leaks (patch 3372909), contributed by Erik Kulyk)
-
-
-Version 0.6:
-- mostly reverted patch 2594366: alsapcm_setup did not do complete error
-checking for good reasons; some ALSA functions in alsapcm_setup may fail without
-rendering the device unusable
-
-
-Version 0.5:
-- applied patch 2777035: Fixed setrec method in alsaaudio.c
-  This included a mixertest with more features
-- fixed/applied patch 2594366: alsapcm_setup does not do any error checking
-
-
-Version 0.4:
-- API changes: mixers() and Mixer() now take a card index instead of a
-  card name as optional parameter.
-- Support for Python 3.0
-- Documentation converted to reStructuredText; use Sphinx instead of LaTeX.
-- added cards()
-- added PCM.close()
-- added Mixer.close()
-- added mixer.getenum()
-
-
-Version 0.3:
-- wrapped blocking calls with Py_BEGIN_ALLOW_THREADS/Py_END_ALLOW_THREADS
-- added pause
-
-
-Version 0.2:
-- Many bugfixes related to playback in particular
-- Module documentation in the doc subdirectory
-
-
-Version 0.1:
-- Initial version

CHANGES.md

@@ -0,0 +1,127 @@
+# 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+- Fixed `Mixer.getvolume()` returning outdated value (#126)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)
+  - support for `periods` in the `PCM` constructor.
+  - new functions `PCM.state()`, `PCM.drop()` and `PCM.drain()`
+  - improved underrun/overrun handling
+  - documentation improvements/consolidation (docstrings were removed in favour of online documentation)
+  - more sampling rates
+  - bug fixes
+
+# Version 0.9.2
+- Fix alsamixer_getvolume (#112 from @stephensp)
+
+# Version 0.9.1:
+- Support decibel, percentage, and raw volumes in getvolume, setvolume, and getrange (#109 from @chrisdiamand)
+
+# Version 0.9.0:
+- Added keyword arguments for channels, format, rate and periodsize
+- Deprecated `setchannel`, `setformat`, `setrate` and `setperiodsize`
+
+# Version 0.8.6:
+- Added four methods to the `PCM` class to allow users to get detailed information about the device:
+
+  - `getformats()` returns a dictionary of name / value pairs, one for each of the card's
+    supported formats - e.g. `{"U8": 1, "S16_LE": 2}`,
+  - `getchannels()` returns a list of the supported channel numbers, e.g. `[1, 2]`,
+  - `getrates()` returns supported sample rates for the device, e.g. `[48000]`,
+  - `getratebounds()` returns the device's official minimum and maximum supported
+    sample rates as a tuple, e.g. `(4000, 48000)`.
+
+  (#82 contributed by @jdstmporter)
+
+- Prevent hang on close after capturing audio (#80 contributed by @daym)
+
+# Version 0.8.5:
+- Return an empty string/bytestring when `read()` detects an
+  overrun. Previously the returned data was undefined (contributed by @jcea)
+
+- Unlimited setperiod buffer size when reading frames (contributed by @jcea)
+
+# Version 0.8.4:
+- Fix Python3 API usage broken in 0.8.3
+
+# Version 0.8.3:
+- Add DSD sample formats (contributed by @lintweaker)
+- Add Mixer.handleevents() to acknowledge events identified by select.poll (contributed by @PaulSD)
+- Add functions for listing cards and their names (contributed by @chrisdiamand)
+- Add a method for setting enums (contributed by @chrisdiamand)
+
+# Version 0.8.2:
+- fix #3 (we cannot get the revision from git for pip installs)
+
+# Version 0.8.1:
+- document changes (this file)
+
+# Version 0.8:
+- `PCM()` has new `device` and `cardindex` keyword arguments.
+
+  The keyword `device` allows to select virtual devices, `cardindex` can be
+  used to select hardware cards by index (as with `mixers()` and `Mixer()`).
+
+  The `card` keyword argument is still supported, but deprecated.
+
+  The reason for this change is that the `card` keyword argument guessed
+  a device name from the card name, but this only works sometimes, and breaks
+  opening virtual devices.
+
+- new function `pcms()` to list available PCM devices.
+
+- `mixers()` and `Mixer()` take an additional `device` keyword argument.
+  This allows to list or open virtual devices.
+
+- The default behaviour of `Mixer()` without any arguments has changed.
+  Now Mixer() will try to open the `default` Mixer instead of the Mixer
+  that is associated with card 0.
+
+- fix a memory leak under Python 3.x
+
+- some more memory leaks in error conditions fixed.
+
+# Version 0.7:
+- fixed several memory leaks (patch 3372909), contributed by Erik Kulyk)
+
+# Version 0.6:
+- mostly reverted patch 2594366: alsapcm_setup did not do complete error
+checking for good reasons; some ALSA functions in alsapcm_setup may fail without
+rendering the device unusable
+
+# Version 0.5:
+- applied patch 2777035: Fixed setrec method in alsaaudio.c
+  This included a mixertest with more features
+- fixed/applied patch 2594366: alsapcm_setup does not do any error checking
+
+# Version 0.4:
+- API changes: mixers() and Mixer() now take a card index instead of a
+  card name as optional parameter.
+- Support for Python 3.0
+- Documentation converted to reStructuredText; use Sphinx instead of LaTeX.
+- added `cards()`
+- added `PCM.close()`
+- added `Mixer.close()`
+- added `mixer.getenum()`
+
+# Version 0.3:
+- wrapped blocking calls with `Py_BEGIN_ALLOW_THREADS`/`Py_END_ALLOW_THREADS`
+- added pause
+
+# Version 0.2:
+- Many bugfixes related to playback in particular
+- Module documentation in the doc subdirectory
+
+# Version 0.1:
+- Initial version

MANIFEST.in

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

README.md

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

alsaaudio.pyi

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

doc/README.md

@@ -1,9 +1,32 @@
+# Make a new release
+
+Update the version in setup.py
+
+    pyalsa_version = '0.9.0'
+
+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
 
 The documentation is published through the `gh-pages` branch.
 
 To publish the documentation, you need to clone the `gh-pages` branch of this repository into
-`doc/gh-pages`. In `doc`, do: 
+`doc/gh-pages`. In `doc`, do:
 
     git clone -b gh-pages git@github.com:larsimmisch/pyalsaaudio.git gh-pages
 

doc/conf.py

@@ -1,182 +1,160 @@
 # -*- coding: utf-8 -*-
 #
-# alsaaudio documentation build configuration file, created by
+# alsaaudio documentation documentation build configuration file, created by
-# sphinx-quickstart on Sat Nov 22 00:17:09 2008.
+# sphinx-quickstart on Thu Mar 30 23:52:21 2017.
 #
-# This file is execfile()d with the current directory set to its containing dir.
+# This file is execfile()d with the current directory set to its
+# containing dir.
 #
-# The contents of this file are pickled, so don't put values in the namespace
+# Note that not all possible configuration values are present in this
-# that aren't pickleable (module imports are okay, they're removed automatically).
+# autogenerated file.
 #
-# All configuration values have a default value; values that are commented out
+# All configuration values have a default; values that are commented out
-# serve to show the default value.
+# serve to show the default.
 
-import sys, os
+# If extensions (or modules to document with autodoc) are in another directory,
+# add these directories to sys.path here. If the directory is relative to the
+# documentation root, use os.path.abspath to make it absolute, like shown here.
+#
+# import os
+# import sys
+# sys.path.insert(0, os.path.abspath('.'))
 
+import sys
 sys.path.insert(0, '..')
 from setup import pyalsa_version
 
-# If your extensions are in another directory, add it here. If the directory
-# is relative to the documentation root, use os.path.abspath to make it
-# absolute, like shown here.
-#sys.path.append(os.path.abspath('some/directory'))
 
-# General configuration
+# -- General configuration ------------------------------------------------
-# ---------------------
 
-# Add any Sphinx extension module names here, as strings. They can be extensions
+# If your documentation needs a minimal Sphinx version, state it here.
-# coming with Sphinx (named 'sphinx.ext.*') or your custom ones.
+#
+# needs_sphinx = '1.0'
+
+# Add any Sphinx extension module names here, as strings. They can be
+# extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
+# ones.
 extensions = []
 
 # Add any paths that contain templates here, relative to this directory.
-templates_path = ['.templates']
+templates_path = ['_templates']
 
-# The suffix of source filenames.
+# The suffix(es) of source filenames.
+# You can specify multiple suffix as a list of string:
+#
+# source_suffix = ['.rst', '.md']
 source_suffix = '.rst'
 
 # The master toctree document.
 master_doc = 'index'
 
-# General substitutions.
+# General information about the project.
-project = u'alsaaudio'
+project = u'alsaaudio documentation'
-copyright = u'2008-20017, Casper Wilstrup, Lars Immisch'
+copyright = u'2017, Lars Immisch & Casper Wilstrup'
+author = u'Lars Immisch & Casper Wilstrup'
 
-# The default replacements for |version| and |release|, also used in various
+# The version info for the project you're documenting, acts as replacement for
-# other places throughout the built documents.
+# |version| and |release|, also used in various other places throughout the
+# built documents.
 #
 # The short X.Y version.
 version = pyalsa_version
 # The full version, including alpha/beta/rc tags.
-release = pyalsa_version
+release = version
 
-# There are two options for replacing |today|: either, you set today to some
+# The language for content autogenerated by Sphinx. Refer to documentation
-# non-false value, then it is used:
+# for a list of supported languages.
-#today = ''
+#
-# Else, today_fmt is used as the format for a strftime call.
+# This is also used if you do content translation via gettext catalogs.
-today_fmt = '%B %d, %Y'
+# Usually you set "language" from the command line for these cases.
+language = 'en'
 
-# List of documents that shouldn't be included in the build.
+# List of patterns, relative to source directory, that match files and
-#unused_docs = []
+# directories to ignore when looking for source files.
-
+# This patterns also effect to html_static_path and html_extra_path
-# List of directories, relative to source directories, that shouldn't be searched
+exclude_patterns = ['_build', 'Thumbs.db', '.DS_Store']
-# for source files.
-exclude_trees = ['.build']
-
-# The reST default role (used for this markup: `text`) to use for all documents.
-#default_role = None
-
-# If true, '()' will be appended to :func: etc. cross-reference text.
-#add_function_parentheses = True
-
-# If true, the current module name will be prepended to all description
-# unit titles (such as .. function::).
-#add_module_names = True
-
-# If true, sectionauthor and moduleauthor directives will be shown in the
-# output. They are ignored by default.
-#show_authors = False
 
 # The name of the Pygments (syntax highlighting) style to use.
 pygments_style = 'sphinx'
 
+# If true, `todo` and `todoList` produce output, else they produce nothing.
+todo_include_todos = False
 
-# Options for HTML output
-# -----------------------
 
-# The style sheet to use for HTML and HTML Help pages. A file of that name
+# -- Options for HTML output ----------------------------------------------
-# must exist either in Sphinx' static/ path, or in one of the custom paths
-# given in html_static_path.
-html_style = 'default.css'
 
-# The name for this set of Sphinx documents.  If None, it defaults to
+# The theme to use for HTML and HTML Help pages.  See the documentation for
-# "<project> v<release> documentation".
+# a list of builtin themes.
-#html_title = None
+#
+html_theme = 'alabaster'
 
-# A shorter title for the navigation bar.  Default is the same as html_title.
+# Theme options are theme-specific and customize the look and feel of a theme
-#html_short_title = None
+# further.  For a list of options available for each theme, see the
-
+# documentation.
-# The name of an image file (relative to this directory) to place at the top
+#
-# of the sidebar.
+# html_theme_options = {}
-#html_logo = None
-
-# The name of an image file (within the static path) to use as favicon of the
-# docs.  This file should be a Windows icon file (.ico) being 16x16 or 32x32
-# pixels large.
-#html_favicon = None
 
 # Add any paths that contain custom static files (such as style sheets) here,
 # relative to this directory. They are copied after the builtin static files,
 # so a file named "default.css" will overwrite the builtin "default.css".
-html_static_path = ['static']
+html_static_path = ['_static']
 
-# If not '', a 'Last updated on:' timestamp is inserted at every page bottom,
-# using the given strftime format.
-html_last_updated_fmt = '%b %d, %Y'
 
-# If true, SmartyPants will be used to convert quotes and dashes to
+# -- Options for HTMLHelp output ------------------------------------------
-# typographically correct entities.
-#html_use_smartypants = True
-
-# Custom sidebar templates, maps document names to template names.
-#html_sidebars = {}
-
-# Additional templates that should be rendered to pages, maps page names to
-# template names.
-#html_additional_pages = {}
-
-# If false, no module index is generated.
-#html_use_modindex = True
-
-# If false, no index is generated.
-#html_use_index = True
-
-# If true, the index is split into individual pages for each letter.
-#html_split_index = False
-
-# If true, the reST sources are included in the HTML build as _sources/<name>.
-#html_copy_source = True
-
-# If true, an OpenSearch description file will be output, and all pages will
-# contain a <link> tag referring to it.  The value of this option must be the
-# base URL from which the finished HTML is served.
-#html_use_opensearch = ''
-
-# If nonempty, this is the file name suffix for HTML files (e.g. ".xhtml").
-#html_file_suffix = ''
 
 # Output file base name for HTML help builder.
-htmlhelp_basename = 'alsaaudiodoc'
+htmlhelp_basename = 'alsaaudiodocumentationdoc'
 
 
-# Options for LaTeX output
+# -- Options for LaTeX output ---------------------------------------------
-# ------------------------
 
-# The paper size ('letter' or 'a4').
+latex_elements = {
-#latex_paper_size = 'letter'
+    # The paper size ('letterpaper' or 'a4paper').
+    #
+    # 'papersize': 'letterpaper',
 
-# The font size ('10pt', '11pt' or '12pt').
+    # The font size ('10pt', '11pt' or '12pt').
-#latex_font_size = '10pt'
+    #
+    # 'pointsize': '10pt',
+
+    # Additional stuff for the LaTeX preamble.
+    #
+    # 'preamble': '',
+
+    # Latex figure (float) alignment
+    #
+    # 'figure_align': 'htbp',
+}
 
 # Grouping the document tree into LaTeX files. List of tuples
-# (source start file, target name, title, author, document class [howto/manual]).
+# (source start file, target name, title,
+#  author, documentclass [howto, manual, or own class]).
 latex_documents = [
-  ('index', 'alsaaudio.tex', u'alsaaudio Documentation',
+    (master_doc, 'alsaaudiodocumentation.tex', u'alsaaudio documentation Documentation',
-   u'Casper Wilstrup, Lars Immisch', 'manual'),
+     u'Lars Immisch', 'manual'),
 ]
 
-# The name of an image file (relative to this directory) to place at the top of
-# the title page.
-#latex_logo = None
 
-# For "manual" documents, if this is true, then toplevel headings are parts,
+# -- Options for manual page output ---------------------------------------
-# not chapters.
+
-#latex_use_parts = False
+# One entry per manual page. List of tuples
+# (source start file, name, description, authors, manual section).
+man_pages = [
+    (master_doc, 'alsaaudiodocumentation', u'alsaaudio documentation Documentation',
+     [author], 1)
+]
+
+
+# -- Options for Texinfo output -------------------------------------------
+
+# Grouping the document tree into Texinfo files. List of tuples
+# (source start file, target name, title, author,
+#  dir menu entry, description, category)
+texinfo_documents = [
+    (master_doc, 'alsaaudiodocumentation', u'alsaaudio documentation Documentation',
+     author, 'alsaaudiodocumentation', 'One line description of project.',
+     'Miscellaneous'),
+]
 
-# Additional stuff for the LaTeX preamble.
-#latex_preamble = ''
 
-# Documents to append as an appendix to all manuals.
-#latex_appendices = []
 
-# If false, no module index is generated.
-#latex_use_modindex = True

doc/index.rst

@@ -1,21 +1,24 @@
 alsaaudio documentation
-=======================
+===================================================
 
 .. toctree::
    :maxdepth: 2
+   :caption: Contents:
 
    pyalsaaudio
    terminology
    libalsaaudio
 
+Download
+========
 
-Github pages
-=================
-
-* `Project page <https://github.com/larsimmisch/pyalsaaudio/>`_
 * `Download from pypi <https://pypi.python.org/pypi/pyalsaaudio>`_
-* `Bug tracker <https://github.com/larsimmisch/pyalsaaudio/issues>`_
 
+Github
+======
+
+* `Repository <https://github.com/larsimmisch/pyalsaaudio/>`_
+* `Bug tracker <https://github.com/larsimmisch/pyalsaaudio/issues>`_
 
 Indices and tables
 ==================

doc/libalsaaudio.rst

@@ -5,42 +5,19 @@
 .. module:: alsaaudio
    :platform: Linux
 
-
-.. % \declaremodule{builtin}{alsaaudio}	% standard library, in C
-.. % not standard, in C
-
 .. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
 .. moduleauthor:: Lars Immisch <lars@ibp.de>
-                  
-.. % Author of the module code;
-
-
 
 The :mod:`alsaaudio` module defines functions and classes for using ALSA.
 
-.. % ---- 3.1. ----
+.. function:: pcms(pcmtype: int = PCM_PLAYBACK) ->list[str]
-.. % For each function, use a ``funcdesc'' block.  This has exactly two
-.. % parameters (each parameters is contained in a set of curly braces):
-.. % the first parameter is the function name (this automatically
-.. % generates an index entry); the second parameter is the function's
-.. % argument list.  If there are no arguments, use an empty pair of
-.. % curly braces.  If there is more than one argument, separate the
-.. % arguments with backslash-comma.  Optional parts of the parameter
-.. % list are contained in \optional{...} (this generates a set of square
-.. % brackets around its parameter).  Arguments are automatically set in
-.. % italics in the parameter list.  Each argument should be mentioned at
-.. % least once in the description; each usage (even inside \code{...})
-.. % should be enclosed in \var{...}.
-
-
-.. function:: pcms([type=PCM_PLAYBACK])
 
    List available PCM devices by name.
-   
+
    Arguments are:
 
-   * *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
+   * *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
-     (default).  
+     (default).
 
    **Note:**
 
@@ -57,14 +34,19 @@ 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`
    instead.
-   
 
-.. function:: mixers(cardindex=-1, device='default')
+..
+   Omitted by intention due to being superseded by cards():
+
+   .. function:: card_indexes()
+   .. function:: card_name()
+
+.. function:: mixers(cardindex: int = -1, device: str = 'default') -> list[str]
 
    List the available mixers. The arguments are:
 
@@ -73,12 +55,14 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
      the `device` keyword argument is ignored. ``0`` is the first hardware sound
      card.
 
+     **Note:** This should not be used, as it bypasses most of ALSA's configuration.
+
    * *device* - the name of the device on which the mixer resides. The default
      is ``'default'``.
 
    **Note:** For a list of available controls, you can also use ``amixer`` on
    the commandline::
-      
+
       $ amixer
 
    To elaborate the example, calling :func:`mixers` with the argument
@@ -92,12 +76,16 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
       $ amixer -D foo
 
    *Changed in 0.8*:
-   
+
    - The keyword argument `device` is new and can be used to
      select virtual devices. As a result, the default behaviour has subtly
      changed. Since 0.8, this functions returns the mixers for the default
      device, not the mixers for the first card.
 
+.. function:: asoundlib_version() -> str
+
+   Return a Python string containing the ALSA version found.
+
 
 .. _pcm-objects:
 
@@ -108,95 +96,35 @@ 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, 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
+   This class is used to represent a PCM device (either for playback or
-   recording). The arguments are:
+   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).
-   * *device* - the name of the PCM device that should be used (for example
+   * *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``.
-     a value from the output of :func:`pcms`). The default value is
+   * *channels* - the number of channels. The default value is 2 (stereo).
-     ``'default'``.
+   * *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures.
-   * *cardindex* - the card index. If this argument is given, the device name
+     The default value is :const:`PCM_FORMAT_S16_LE`.
-     is constructed as 'hw:*cardindex*' and
-     the `device` keyword argument is ignored.
-     ``0`` is the first hardware sound card.
-
-   This will construct a PCM object with these default settings:
-
-   * Sample format: :const:`PCM_FORMAT_S16_LE`
-   * Rate: 44100 Hz 
-   * Channels: 2
-   * Period size: 32 frames
-
-   *Changed in 0.8:*
-   
-   - The `card` keyword argument is still supported,
-     but deprecated. Please use `device` instead.
-
-   - The keyword argument `cardindex` was added.
-
-   The `card` keyword is deprecated because it guesses the real ALSA
-   name of the card. This was always fragile and broke some legitimate usecases.
-   
-     
-PCM objects have the following methods:
-
-
-.. method:: PCM.pcmtype()
-
-   Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
-   :const:`PCM_PLAYBACK`.
-
-
-.. method:: PCM.pcmmode()
-
-   Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
-   :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
-
-
-.. method:: PCM.cardname()
-
-   Return the name of the sound card used by this PCM object.
-
-
-.. method:: PCM.setchannels(nchannels)
-
-   Used to set the number of capture or playback channels. Common
-   values are: ``1`` = mono, ``2`` = stereo, and ``6`` = full 6 channel audio.
-   Few sound cards support more than 2 channels
-
-
-.. method:: PCM.setrate(rate)
-
-   Set the sample rate in Hz for the device. Typical values are ``8000``
-   (mainly used for telephony), ``16000``, ``44100`` (CD quality),
-   ``48000`` and ``96000``.
-
-
-.. method:: PCM.setformat(format)
-
-   The sound *format* of the device. Sound format controls how the PCM device
-   interpret data for playback, and how data is encoded in captures.
-
-   The following formats are provided by ALSA:
 
    =========================  ===============
-          Format                Description
+        Format                Description
    =========================  ===============
    ``PCM_FORMAT_S8``          Signed 8 bit samples for each channel
-   ``PCM_FORMAT_U8``          Signed 8 bit samples for each channel
+   ``PCM_FORMAT_U8``          Unsigned 8 bit samples for each channel
    ``PCM_FORMAT_S16_LE``      Signed 16 bit samples for each channel Little Endian byte order)
    ``PCM_FORMAT_S16_BE``      Signed 16 bit samples for each channel (Big Endian byte order)
    ``PCM_FORMAT_U16_LE``      Unsigned 16 bit samples for each channel (Little Endian byte order)
    ``PCM_FORMAT_U16_BE``      Unsigned 16 bit samples for each channel (Big Endian byte order)
-   ``PCM_FORMAT_S24_LE``      Signed 24 bit samples for each channel (Little Endian byte order)
+   ``PCM_FORMAT_S24_LE``      Signed 24 bit samples for each channel (Little Endian byte order in 4 bytes)
-   ``PCM_FORMAT_S24_BE``      Signed 24 bit samples for each channel (Big Endian byte order)}
+   ``PCM_FORMAT_S24_BE``      Signed 24 bit samples for each channel (Big Endian byte order in 4 bytes)
-   ``PCM_FORMAT_U24_LE``      Unsigned 24 bit samples for each channel (Little Endian byte order)
+   ``PCM_FORMAT_U24_LE``      Unsigned 24 bit samples for each channel (Little Endian byte order in 4 bytes)
-   ``PCM_FORMAT_U24_BE``      Unsigned 24 bit samples for each channel (Big Endian byte order)
+   ``PCM_FORMAT_U24_BE``      Unsigned 24 bit samples for each channel (Big Endian byte order in 4 bytes)
    ``PCM_FORMAT_S32_LE``      Signed 32 bit samples for each channel (Little Endian byte order)
    ``PCM_FORMAT_S32_BE``      Signed 32 bit samples for each channel (Big Endian byte order)
    ``PCM_FORMAT_U32_LE``      Unsigned 32 bit samples for each channel (Little Endian byte order)
@@ -210,59 +138,365 @@ PCM objects have the following methods:
    ``PCM_FORMAT_IMA_ADPCM``   A 4:1 compressed format defined by the Interactive Multimedia Association.
    ``PCM_FORMAT_MPEG``        MPEG encoded audio?
    ``PCM_FORMAT_GSM``         9600 bits/s constant rate encoding for speech
+   ``PCM_FORMAT_S24_3LE``     Signed 24 bit samples for each channel (Little Endian byte order in 3 bytes)
+   ``PCM_FORMAT_S24_3BE``     Signed 24 bit samples for each channel (Big Endian byte order in 3 bytes)
+   ``PCM_FORMAT_U24_3LE``     Unsigned 24 bit samples for each channel (Little Endian byte order in 3 bytes)
+   ``PCM_FORMAT_U24_3BE``     Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
    =========================  ===============
-   
 
-.. method:: PCM.setperiodsize(period)
+   * *periodsize* - the period size in frames.
+     Make sure you understand :ref:`the meaning of periods <term-period>`.
+     The default value is 32, which is below the actual minimum of most devices,
+     and will therefore likely be larger in practice.
+   * *periods* - the number of periods in the buffer. The default value is 4.
+   * *device* - the name of the PCM device that should be used (for example
+     a value from the output of :func:`pcms`). The default value is
+     ``'default'``.
+   * *cardindex* - the card index. If this argument is given, the device name
+     is constructed as 'hw:*cardindex*' and
+     the `device` keyword argument is ignored.
+     ``0`` is the first hardware sound card.
 
-   Sets the actual period size in frames. Each write should consist of
+     **Note:** This should not be used, as it bypasses most of ALSA's configuration.
-   exactly this number of frames, and each read will return this
-   number of frames (unless the device is in :const:`PCM_NONBLOCK` mode, in
-   which case it may return nothing at all)
 
+   The defaults mentioned above are values passed by :mod:alsaaudio
+   to ALSA, not anything internal to ALSA.
 
-.. method:: PCM.read()
+   **Note:** For default and non-default values alike, there is no
+   guarantee that a PCM device supports the requested configuration,
+   and ALSA may pick realizable values which it believes to be closest
+   to the request. Therefore, after creating a PCM object, it is
+   necessary to verify whether its realized configuration is acceptable.
+   The :func:info method can be used to query it.
+
+   *Changed in 0.10:*
+
+   - Added the optional named parameter `periods`.
+
+   *Changed in 0.9:*
+
+   - Added the optional named parameters `rate`, `channels`, `format` and `periodsize`.
+
+   *Changed in 0.8:*
+
+   - The `card` keyword argument is still supported,
+     but deprecated. Please use `device` instead.
+
+   - The keyword argument `cardindex` was added.
+
+   The `card` keyword is deprecated because it guesses the real ALSA
+   name of the card. This was always fragile and broke some legitimate usecases.
+
+PCM objects have the following methods:
+
+.. method:: PCM.info() -> dict
+
+   Returns a dictionary containing the configuration of a PCM device.
+
+   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.
+
+   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() -> int
+
+   Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
+   :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
+
+.. method:: PCM.cardname() -> string
+
+   Return the name of the sound card used by this PCM object.
+
+..
+   Omitted by intention due to not really fitting the c'tor-based setup concept:
+
+   .. method:: PCM.getchannels()
+
+   Returns list of the device's supported channel counts.
+
+   .. method:: PCM.getratebounds()
+
+   Returns the card's minimum and maximum supported sample rates as
+   a tuple of integers.
+
+   .. method:: PCM.getrates()
+
+   Returns the sample rates supported by the device.
+   The returned value can be of one of the following, depending on
+   the card's properties:
+   * Card supports only a single rate: returns the rate
+   * Card supports a continuous range of rates: returns a tuple of
+     the range's lower and upper bounds (inclusive)
+   * Card supports a collection of well-known rates: returns a list of
+     the supported rates
+
+   .. method:: PCM.getformats()
+
+   Returns a dictionary of supported format codes (integers) keyed by
+   their standard ALSA names (strings).
+
+.. method:: PCM.setchannels(nchannels: int) -> int
+
+   .. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
+
+.. method:: PCM.setrate(rate: int) -> int
+
+   .. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
+
+.. method:: PCM.setformat(format: int) -> int
+
+   .. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
+
+.. method:: PCM.setperiodsize(period: int) -> int
+
+   .. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
+
+.. method:: PCM.state() -> int
+
+   Returs the current state of the stream, which can be one of
+   :const:`PCM_STATE_OPEN` (this should not actually happen),
+   :const:`PCM_STATE_SETUP` (after :func:`drop` or :func:`drain`),
+   :const:`PCM_STATE_PREPARED` (after construction),
+   :const:`PCM_STATE_RUNNING`,
+   :const:`PCM_STATE_XRUN`,
+   :const:`PCM_STATE_DRAINING`,
+   :const:`PCM_STATE_PAUSED`,
+   :const:`PCM_STATE_SUSPENDED`, and
+   :const:`PCM_STATE_DISCONNECTED`.
+
+   *New in 0.10*
+
+.. method:: PCM.avail() -> int
+
+   For :const:`PCM_PLAYBACK` PCM objects, returns the number of writable
+   (that is, free) frames in the buffer.
+
+   For :const:`PCM_CAPTURE` PCM objects, returns the number of readable
+   (that is, filled) frames in the buffer.
+
+   An attempt to read/write more frames than indicated will block (in
+   :const:`PCM_NORMAL` mode) or fail and return zero (in
+   :const:`PCM_NONBLOCK` mode).
+
+   *New in 0.11*
+
+.. method:: PCM.read() -> tuple[int, bytes]
 
    In :const:`PCM_NORMAL` mode, this function blocks until a full period is
    available, and then returns a tuple (length,data) where *length* is
    the number of frames of captured data, and *data* is the captured
-   sound frames as a string. The length of the returned data will be 
+   sound frames as a string. The length of the returned data will be
    periodsize\*framesize bytes.
 
    In :const:`PCM_NONBLOCK` mode, the call will not block, but will return
    ``(0,'')`` if no new period has become available since the last
    call to read.
 
+   In case of a buffer overrun, this function will return the negative
+   size :const:`-EPIPE`, and no data is read.
+   This indicates that data was lost. To resume capturing, just call read
+   again, but note that the stream was already corrupted.
+   To avoid the problem in the future, try using a larger period size
+   and/or more periods, at the cost of higher latency.
 
-.. method:: PCM.write(data)
+.. method:: PCM.write(data: bytes) -> int
 
    Writes (plays) the sound in data. The length of data *must* be a
    multiple of the frame size, and *should* be exactly the size of a
    period. If less than 'period size' frames are provided, the actual
    playout will not happen until more data is written.
 
-   If the device is not in :const:`PCM_NONBLOCK` mode, this call will block if
+   If the data was successfully written, the call returns the size of the
-   the kernel buffer is full, and until enough sound has been played
+   data *in frames*.
-   to allow the sound data to be buffered. The call always returns the
+
-   size of the data provided.
+   If the device is not in :const:`PCM_NONBLOCK` mode, this call will block
+   if the kernel buffer is full, and until enough sound has been played
+   to allow the sound data to be buffered.
 
    In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a
    return value of zero, if the buffer is full. In this case, the data
-   should be written at a later time.
+   should be written again at a later time.
 
+   In case of a buffer underrun, this function will return the negative
+   size :const:`-EPIPE`, and no data is written.
+   At this point, the playback was already corrupted. If you want to play
+   the data nonetheless, call write again with the same data.
+   To avoid the problem in the future, try using a larger period size
+   and/or more periods, at the cost of higher latency.
 
-.. method:: PCM.pause([enable=True])
+   Note that this call completing means only that the samples were buffered
+   in the kernel, and playout will continue afterwards. Make sure that the
+   stream is drained before discarding the PCM handle.
+
+.. method:: PCM.pause([enable: int = True]) -> int
 
    If *enable* is :const:`True`, playback or capture is paused.
    Otherwise, playback/capture is resumed.
 
+.. method:: PCM.drop() -> int
+
+   Stop the stream and drop residual buffered frames.
+
+   *New in 0.9*
+
+.. method:: PCM.drain() -> int
+
+   For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
+   and then stop the stream. In :const:`PCM_NORMAL` mode,
+   this function blocks until all pending playback is drained.
+
+   For :const:`PCM_CAPTURE` PCM objects, this function is not very useful.
+
+   *New in 0.10*
+
+.. method:: PCM.close() -> None
+
+   Closes the PCM device.
+
+   For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
+   this function blocks until all pending playback is drained.
+
+.. method:: PCM.polldescriptors() -> list[tuple[int, int]]
+
+   Returns a list of tuples of *(file descriptor, eventmask)* that can be
+   used to wait for changes on the PCM with *select.poll*.
+
+   The *eventmask* value is compatible with `poll.register`__ in the Python
+   :const:`select` module.
+
+.. method:: PCM.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() -> 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: 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() -> 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() -> tuple[int, int, int]
+
+   Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*.
+
+   The type of output is controlled by the tstamp_type, as described in the table below.
+
+   =================================  ===========================================
+            Timestamp Type                             Description
+   =================================  ===========================================
+   ``PCM_TSTAMP_TYPE_GETTIMEOFDAY``   System-wide realtime clock with seconds
+                                      since epoch.
+   ``PCM_TSTAMP_TYPE_MONOTONIC``      Monotonic time from an unspecified starting
+                                      time. Progress is NTP synchronized.
+   ``PCM_TSTAMP_TYPE_MONOTONIC_RAW``  Monotonic time from an unspecified starting
+                                      time using only the system clock.
+   =================================  ===========================================
+
+   The timestamp mode is controlled by the tstamp_mode, as described in the table below.
+
+   =================================  ===========================================
+            Timestamp Mode                             Description
+   =================================  ===========================================
+   ``PCM_TSTAMP_NONE``                No timestamp.
+   ``PCM_TSTAMP_ENABLE``              Update timestamp at every hardware position
+                                      update.
+   =================================  ===========================================
+
 **A few hints on using PCM devices for playback**
 
-The most common reason for problems with playback of PCM audio is that writes 
+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).
 
@@ -291,13 +525,12 @@ 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.
@@ -307,35 +540,32 @@ Mixer objects provides access to the ALSA mixer API.
    * *cardindex* - specifies which card should be used. If this argument
      is given, the device name is constructed like this: 'hw:*cardindex*' and
      the `device` keyword argument is ignored. ``0`` is the
-     first sound card. 
+     first sound card.
 
    * *device* - the name of the device on which the mixer resides. The default
      value is ``'default'``.
-   
+
    *Changed in 0.8*:
-   
+
    - The keyword argument `device` is new and can be used to select virtual
      devices.
-  
+
 Mixer objects have the following methods:
 
-.. method:: Mixer.cardname()
+.. method:: Mixer.cardname() -> str
 
    Return the name of the sound card used by this Mixer object
 
-
+.. method:: Mixer.mixer() -> 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:
@@ -347,7 +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
    ======================  ================
@@ -355,8 +585,7 @@ 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:
@@ -371,8 +600,8 @@ Mixer objects have the following methods:
    'Capture Volume'          Manipulate sound capture volume
    'Joined Capture Volume'   Manipulate sound capture volume for all channels at a time
    ========================  ================
-   
+
-.. method:: Mixer.getenum()
+.. method:: Mixer.getenum() -> tuple[str, list[str]]
 
    For enumerated controls, return the currently selected item and  the list of
    items available.
@@ -398,59 +627,64 @@ Mixer objects have the following methods:
    This method will return an empty tuple if the mixer is not an  enumerated
    control.
 
+.. method:: Mixer.setenum(index: int) -> None
 
-.. method:: Mixer.getmute()
+   For enumerated controls, sets the currently selected item.
+   *index* is an index into the list of available enumerated items returned
+   by :func:`getenum`.
 
-   Return a list indicating the current mute setting for each
+.. method:: Mixer.getrange(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]
-   channel. 0 means not muted, 1 means muted.
-
-   This method will fail if the mixer has no playback switch capabilities.
-
-
-.. method:: Mixer.getrange([direction])
 
    Return the volume range of the ALSA mixer controlled by this object.
+   The value is a tuple of integers whose meaning is determined by the
+   *units* argument.
 
-   The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
+   The optional *pcmtype* argument can be either :const:`PCM_PLAYBACK` or
    :const:`PCM_CAPTURE`, which is relevant if the mixer can control both
    playback and capture volume.  The default value is :const:`PCM_PLAYBACK`
    if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
 
+   The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
+   :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
 
-.. method:: Mixer.getrec()
+.. method:: Mixer.getvolume(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> 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.getvolume([direction])
 
    Returns a list with the current volume settings for each channel. The list
-   elements are integer percentages.
+   elements are integers whose meaning is determined by the *units* argument.
 
-   The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
+   The optional *pcmtype* argument can be either :const:`PCM_PLAYBACK` or
    :const:`PCM_CAPTURE`, which is relevant if the mixer can control both
    playback and capture volume. The default value is :const:`PCM_PLAYBACK`
    if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
 
+   The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
+   :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
 
-.. method:: Mixer.setvolume(volume, [channel], [direction])
+.. method:: Mixer.setvolume(volume: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None
 
    Change the current volume settings for this mixer. The *volume* argument
-   controls the new volume setting as an integer percentage.
+   is an integer whose meaning is determined by the *units* argument.
 
    If the optional argument *channel* is present, the volume is set
    only for this channel. This assumes that the mixer can control the
    volume for the channels independently.
 
-   The optional *direction* argument can be either :const:`PCM_PLAYBACK` or
+   The optional *pcmtype* argument can be either :const:`PCM_PLAYBACK` or
    :const:`PCM_CAPTURE`, which is relevant if the mixer can control both
    playback and capture volume. The default value is :const:`PCM_PLAYBACK`
    if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
 
-.. method:: Mixer.setmute(mute, [channel])
+   The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
+   :const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
+
+.. method:: Mixer.getmute() -> list[int]
+
+   Return a list indicating the current mute setting for each channel.
+   0 means not muted, 1 means muted.
+
+   This method will fail if the mixer has no playback switch capabilities.
+
+.. method:: Mixer.setmute(mute: bool, channel: (int | None) = None) -> None
 
    Sets the mute flag to a new value. The *mute* argument is either 0 for not
    muted, or 1 for muted.
@@ -460,8 +694,14 @@ Mixer objects have the following methods:
 
    This method will fail if the mixer has no playback mute capabilities
 
+.. method:: Mixer.getrec() -> list[int]
 
-.. method:: Mixer.setrec(capture, [channel])
+   Return a list indicating the current record mute setting for each channel.
+   0 means not recording, 1 means recording.
+
+   This method will fail if the mixer has no capture switch capabilities.
+
+.. method:: Mixer.setrec(capture: int, channel: (int | None) = None) -> None
 
    Sets the capture mute flag to a new value. The *capture* argument
    is either 0 for no capture, or 1 for capture.
@@ -471,17 +711,24 @@ Mixer objects have the following methods:
 
    This method will fail if the mixer has no capture switch capabilities.
 
-.. method:: Mixer.polldescriptors()
+.. method:: Mixer.polldescriptors() -> list[tuple[int, int]]
 
-   Returns a tuple of (file descriptor, eventmask) that can be used to
+   Returns a list of tuples of *(file descriptor, eventmask)* that can be
-   wait for changes on the mixer with *select.poll*.
+   used to wait for changes on the mixer with *select.poll*.
 
-.. method:: Mixer.handleevents()
+   The *eventmask* value is compatible with `poll.register`__ in the Python
+   :const:`select` module.
 
-   Acknowledge events on the *polldescriptors* file descriptors
+.. method:: Mixer.handleevents() -> int
+
+   Acknowledge events on the :func:`polldescriptors` file descriptors
    to prevent subsequent polls from returning the same events again.
    Returns the number of events that were acknowledged.
 
+.. method:: Mixer.close() -> None
+
+   Closes the Mixer device.
+
 **A rant on the ALSA Mixer API**
 
 The ALSA mixer API is extremely complicated - and hardly documented at all.
@@ -504,8 +751,6 @@ Unfortunately, I'm not able to create such a HOWTO myself, since I only
 understand half of the API, and that which I do understand has come from a
 painful trial and error process.
 
-.. % ==== 4. ====
-
 
 .. _pcm-example:
 
@@ -519,7 +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::
@@ -534,12 +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::
 
@@ -547,6 +792,7 @@ To test PCM playback (on your default soundcard), run::
 
 recordtest.py and playbacktest.py
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
 **recordtest.py** and **playbacktest.py** will record and play a raw
 sound file in CD quality.
 
@@ -568,7 +814,7 @@ Without arguments, **mixertest.py** will list all available *controls* on the
 default soundcard.
 
 The output might look like this::
-   
+
   $ ./mixertest.py
   Available mixer controls:
      'Master'
@@ -586,7 +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
@@ -595,7 +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
@@ -616,7 +862,3 @@ argument::
   Capabilities: Playback Volume Playback Mute
   Channel 0 volume: 61%
   Channel 1 volume: 61%
-
-.. rubric:: Footnotes
-
-.. [#f1]   ALSA also allows ``PCM_ASYNC``, but this is not supported yet.

doc/pyalsaaudio.rst

@@ -7,33 +7,19 @@ Introduction
 
 .. |release| replace:: version
 
-.. % At minimum, give your name and an email address.  You can include a
-.. % snail-mail address if you like.
-
-.. % This makes the Abstract go on a separate page in the HTML version;
-.. % if a copyright notice is used, it should go immediately after this.
-.. % 
-
-
 .. _front:
 
 This software is licensed under the PSF license - the same one used by the
 majority of the python distribution. Basically you can use it for anything you
 wish (even commercial purposes). There is no warranty whatsoever.
 
-.. % Copyright statement should go here, if needed.
-
-.. % The abstract should be a paragraph or two long, and describe the
-.. % scope of the document.
-
-
 .. topic:: Abstract
 
    This package contains wrappers for accessing the ALSA API from Python. It is
    currently fairly complete for PCM devices and Mixer access. MIDI sequencer
    support is low on our priority list, but volunteers are welcome.
 
-   If you find bugs in the wrappers please use thegithub issue tracker. 
+   If you find bugs in the wrappers please use the github issue tracker.
    Please don't send bug reports regarding ALSA specifically. There are several
    bugs in this API, and those should be reported to the ALSA team - not me.
 
@@ -64,8 +50,8 @@ More information about ALSA may be found on the project homepage
 ALSA and Python
 ===============
 
-The older Linux sound API (OSS) which is now deprecated is well supported from
+The older Linux sound API (OSS) -- which is now deprecated -- is well supported
-the standard Python library, through the ossaudiodev module. No native ALSA
+by the standard Python library, through the ossaudiodev module. No native ALSA
 support exists in the standard library.
 
 There are a few other "ALSA for Python" projects available, including at least
@@ -75,7 +61,7 @@ development at the time - and neither are very feature complete.
 I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module
 included in the standard Python library, but that looks currently unlikely.
 
-PyAlsaAudio hass full support for sound capture, playback of sound, as well as
+PyAlsaAudio has full support for sound capture, playback of sound, as well as
 the ALSA Mixer API.
 
 MIDI support is not available, and since I don't own any MIDI hardware, it's
@@ -95,7 +81,7 @@ and need the ALSA headers for compilation.  Verify that you have
 
 Naturally you also need to use a kernel with proper ALSA support. This is the
 default in Linux kernel 2.6 and later. If you are using kernel version 2.4 you
-may need to install the ALSA patches yourself - although most distributions 
+may need to install the ALSA patches yourself - although most distributions
 ship with ALSA kernels.
 
 To install, execute the following:  ---   ::
@@ -106,29 +92,37 @@ And then as root:  ---   ::
 
    # python setup.py install
 
+
 *******
 Testing
 *******
 
-First of all, run::
+Make sure that :code:`aplay` plays a file through the soundcard you want, then
-   
+try::
-   $ python test.py
 
-This is a small test suite that mostly performs consistency tests. If
+  $ python playwav.py <filename.wav>
-it fails, please file a `bug report
+
-<https://github.com/larsimmisch/pyalsaaudio/issues>`_.
+If :code:`aplay` needs a device argument, like
+:code:`aplay -D hw:CARD=sndrpihifiberry,DEV=0`, use::
+
+   $ python playwav.py -d hw:CARD=sndrpihifiberry,DEV=0 <filename.wav>
 
 To test PCM recordings (on your default soundcard), verify your
 microphone works, then do::
 
-   $ python recordtest.py <filename>
+   $ python recordtest.py -d <device> <filename>
 
 Speak into the microphone, and interrupt the recording at any time
 with ``Ctl-C``.
 
 Play back the recording with::
 
-   $ python playbacktest.py <filename>
+   $ python playbacktest.py -d <device> <filename>
 
+There is a minimal test suite in :code:`test.py`, but it is
+a bit dependent on the ALSA configuration and may fail without indicating
+a real problem.
 
+If you find bugs/problems, please file a `bug report
+<https://github.com/larsimmisch/pyalsaaudio/issues>`_.
 

doc/terminology.rst

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

isine.py

@@ -6,40 +6,59 @@
 
 from __future__ import print_function
 
+import sys
+import time
 from threading import Thread
-from queue import Queue, Empty
+from multiprocessing import Queue
-from math import pi, sin
+
+if sys.version_info[0] < 3:
+    from Queue import Empty
+else:
+    from queue import Empty
+
+from math import pi, sin, ceil
 import struct
+import itertools
 import alsaaudio
 
-sampling_rate = 44100
+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 digitize(s):
+def nearest_frequency(frequency):
-    if s > 1.0 or s < -1.0:
+    # calculate the nearest frequency where the wave form fits into the buffer
-        raise ValueError
+    # in other words, select f so that sampling_rate/f is an integer
-        
+    return float(sampling_rate)/int(sampling_rate/frequency)
-    return struct.pack('h', int(s * 32767)) 
 
-def generate(frequency):
+def generate(frequency, duration = 0.125):
-    # generate a buffer with a sine wave of frequency
+    # generate a buffer with a sine wave of `frequency`
-    size = int(sampling_rate / frequency)
+    # that is approximately `duration` seconds long
-    buffer = bytes()
+
-    for i in range(size):
+    # the length of a full sine wave at the frequency
-        buffer = buffer + digitize(sin(i/(2 * pi)))
+    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
+
+    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 buffer
 
 class SinePlayer(Thread):
-    
+
     def __init__(self, frequency = 440.0):
-        Thread.__init__(self)
+        Thread.__init__(self, daemon=True)
-        self.setDaemon(True)
+        self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
-        self.device = alsaaudio.PCM()
-        self.device.setchannels(1)
-        self.device.setformat(format)
-        self.device.setrate(sampling_rate)
         self.queue = Queue()
         self.change(frequency)
 
@@ -47,35 +66,32 @@ class SinePlayer(Thread):
         '''This is called outside of the player thread'''
         # we generate the buffer in the calling thread for less
         # latency when switching frequencies
-        
 
-        # More than 100 writes/s are pushing it - play multiple buffers
+        if frequency > sampling_rate / 2:
-        # for higher frequencies
+            raise ValueError('maximum frequency is %d' % (sampling_rate / 2))
 
-        factor = int(frequency/100.0)
+        f = nearest_frequency(frequency)
-        if factor == 0:
+        print('nearest frequency: %f' % f)
-            factor = 1
+
-        
+        buf = generate(f)
-        buf = generate(frequency) * factor
+        self.queue.put(buf)
-        print('factor: %d, frames: %d' % (factor, len(buf) / framesize))
 
-        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:
+                pass
-                    self.device.write(buffer)
+            if buffer:
-                
+                if self.device.write(buffer) < 0:
+                    print("Playback buffer underrun! Continuing nonetheless ...")
+
 
 isine = SinePlayer()
 isine.start()
 
-def change(f):
+time.sleep(1)
-    isine.change(f)
+isine.change(1000)
-    
+time.sleep(1)
+

loopback.py

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

mixertest.py

@@ -43,12 +43,42 @@ def show_mixer(name, kwargs):
         sys.exit(1)
 
     print("Mixer name: '%s'" % mixer.mixer())
-    print("Capabilities: %s %s" % (' '.join(mixer.volumecap()),
+    volcap = mixer.volumecap()
+    print("Capabilities: %s %s" % (' '.join(volcap),
                                    ' '.join(mixer.switchcap())))
+
+    if "Volume" in volcap or "Joined Volume" in volcap or "Playback Volume" in volcap:
+        pmin, pmax = mixer.getrange(alsaaudio.PCM_PLAYBACK)
+        pmin_keyword, pmax_keyword = mixer.getrange(pcmtype=alsaaudio.PCM_PLAYBACK, units=alsaaudio.VOLUME_UNITS_RAW)
+        pmin_default, pmax_default = mixer.getrange()
+        assert pmin == pmin_keyword
+        assert pmax == pmax_keyword
+        assert pmin == pmin_default
+        assert pmax == pmax_default
+        print("Raw playback volume range {}-{}".format(pmin, pmax))
+        pmin_dB, pmax_dB = mixer.getrange(units=alsaaudio.VOLUME_UNITS_DB)
+        print("dB playback volume range {}-{}".format(pmin_dB / 100.0, pmax_dB / 100.0))
+
+    if "Capture Volume" in volcap or "Joined Capture Volume" in volcap:
+        # Check that `getrange` works with keyword and positional arguments
+        cmin, cmax = mixer.getrange(alsaaudio.PCM_CAPTURE)
+        cmin_keyword, cmax_keyword = mixer.getrange(pcmtype=alsaaudio.PCM_CAPTURE, units=alsaaudio.VOLUME_UNITS_RAW)
+        assert cmin == cmin_keyword
+        assert cmax == cmax_keyword
+        print("Raw capture volume range {}-{}".format(cmin, cmax))
+        cmin_dB, cmax_dB = mixer.getrange(pcmtype=alsaaudio.PCM_CAPTURE, units=alsaaudio.VOLUME_UNITS_DB)
+        print("dB capture volume range {}-{}".format(cmin_dB / 100.0, cmax_dB / 100.0))
+
     volumes = mixer.getvolume()
+    volumes_dB = mixer.getvolume(units=alsaaudio.VOLUME_UNITS_DB)
     for i in range(len(volumes)):
-        print("Channel %i volume: %i%%" % (i,volumes[i]))
+        print("Channel %i playback volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
-        
+
+    volumes = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE)
+    volumes_dB = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE, units=alsaaudio.VOLUME_UNITS_DB)
+    for i in range(len(volumes)):
+        print("Channel %i capture volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
+
     try:
         mutes = mixer.getmute()
         for i in range(len(mutes)):
@@ -88,7 +118,7 @@ def set_mixer(name, args, kwargs):
             mixer.setmute(1, channel)
         else:
             mixer.setmute(0, channel)
-        
+
     elif args in ['rec','unrec']:
         # Enable/disable recording
         if args == 'rec':

playbacktest.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 ## playbacktest.py
 ##
@@ -38,22 +39,16 @@ if __name__ == '__main__':
 
     f = open(args[0], 'rb')
 
-    # Open the device in playback mode. 
+    # Open the device in playback mode in Mono, 44100 Hz, 16 bit little endian frames
-    out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, device=device)
-
-    # Set attributes: Mono, 44100 Hz, 16 bit little endian frames
-    out.setchannels(1)
-    out.setrate(44100)
-    out.setformat(alsaaudio.PCM_FORMAT_S16_LE)
-
     # The period size controls the internal number of frames per period.
     # The significance of this parameter is documented in the ALSA api.
-    out.setperiodsize(160)
 
+    out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device)
     # Read data from stdin
     data = f.read(320)
     while data:
-        out.write(data)
+        if out.write(data) < 0:
+            print("Playback buffer underrun! Continuing nonetheless ...")
         data = f.read(320)
-            
+    out.close()
-        
+

playwav.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 # Simple test script that plays (some) wav files
 
@@ -9,57 +10,55 @@ import wave
 import getopt
 import alsaaudio
 
-def play(device, f):    
+def play(device, f):	
 
-    print('%d channels, %d sampling rate\n' % (f.getnchannels(),
+	format = None
-                                               f.getframerate()))
-    # Set attributes
-    device.setchannels(f.getnchannels())
-    device.setrate(f.getframerate())
 
-    # 8bit is unsigned in wav files
+	# 8bit is unsigned in wav files
-    if f.getsampwidth() == 1:
+	if f.getsampwidth() == 1:
-        device.setformat(alsaaudio.PCM_FORMAT_U8)
+		format = alsaaudio.PCM_FORMAT_U8
-    # Otherwise we assume signed data, little endian
+	# Otherwise we assume signed data, little endian
-    elif f.getsampwidth() == 2:
+	elif f.getsampwidth() == 2:
-        device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
+		format = alsaaudio.PCM_FORMAT_S16_LE
-    elif f.getsampwidth() == 3:
+	elif f.getsampwidth() == 3:
-        device.setformat(alsaaudio.PCM_FORMAT_S24_LE)
+		format = alsaaudio.PCM_FORMAT_S24_3LE
-    elif f.getsampwidth() == 4:
+	elif f.getsampwidth() == 4:
-        device.setformat(alsaaudio.PCM_FORMAT_S32_LE)
+		format = alsaaudio.PCM_FORMAT_S32_LE
-    else:
+	else:
-        raise ValueError('Unsupported format')
+		raise ValueError('Unsupported format')
 
-    periodsize = f.getframerate() / 8
+	periodsize = f.getframerate() // 8
 
-    device.setperiodsize(periodsize)
+	print('%d channels, %d sampling rate, format %d, periodsize %d\n' % (f.getnchannels(),
-    
+																		 f.getframerate(),
-    data = f.readframes(periodsize)
+																		 format,
-    while data:
+																		 periodsize))
-        # Read data from stdin
+
-        device.write(data)
+	device = alsaaudio.PCM(channels=f.getnchannels(), rate=f.getframerate(), format=format, periodsize=periodsize, device=device)
-        data = f.readframes(periodsize)
+	
+	data = f.readframes(periodsize)
+	while data:
+		# Read data from stdin
+		if device.write(data) < 0:
+			print("Playback buffer underrun! Continuing nonetheless ...")
+		data = f.readframes(periodsize)
 
 
 def usage():
-    print('usage: playwav.py [-d <device>] <file>', file=sys.stderr)
+	print('usage: playwav.py [-d <device>] <file>', file=sys.stderr)
-    sys.exit(2)
+	sys.exit(2)
 
 if __name__ == '__main__':
 
-    device = 'default'
+	device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'd:')
+	opts, args = getopt.getopt(sys.argv[1:], 'd:')
-    for o, a in opts:
+	for o, a in opts:
-        if o == '-d':
+		if o == '-d':
-            device = a
+			device = a
 
-    if not args:
+	if not args:
-        usage()
+		usage()
-        
+		
-    f = wave.open(args[0], 'rb')
+	with wave.open(args[0], 'rb') as f:
-    device = alsaaudio.PCM(device=device)
+		play(device, f)
-
-    play(device, f)
-
-    f.close()

recordtest.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 ## recordtest.py
 ##
@@ -22,48 +23,44 @@ import getopt
 import alsaaudio
 
 def usage():
-    print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr)
+	print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr)
-    sys.exit(2)
+	sys.exit(2)
 
 if __name__ == '__main__':
 
-    device = 'default'
+	device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'd:')
+	opts, args = getopt.getopt(sys.argv[1:], 'd:')
-    for o, a in opts:
+	for o, a in opts:
-        if o == '-d':
+		if o == '-d':
-            device = a
+			device = a
 
-    if not args:
+	if not args:
-        usage()
+		usage()
 
-    f = open(args[0], 'wb')
+	f = open(args[0], 'wb')
 
-    # Open the device in nonblocking capture mode. The last argument could
+	# Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz
-    # just as well have been zero for blocking mode. Then we could have
+	# and 16 bit little endian samples
-    # left out the sleep call in the bottom of the loop
+	# The period size controls the internal number of frames per period.
-    inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, device=device)
+	# The significance of this parameter is documented in the ALSA api.
+	# For our purposes, it is suficcient to know that reads from the device
+	# will return this many frames. Each frame being 2 bytes long.
+	# 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,
+		periodsize=160, device=device)
 
-    # Set attributes: Mono, 44100 Hz, 16 bit little endian samples
+	loops = 1000000
-    inp.setchannels(1)
+	while loops > 0:
-    inp.setrate(44100)
+		loops -= 1
-    inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
+		# Read data from device
+		l, data = inp.read()
 
-    # The period size controls the internal number of frames per period.
+		if l < 0:
-    # The significance of this parameter is documented in the ALSA api.
+			print("Capture buffer overrun! Continuing nonetheless ...")
-    # For our purposes, it is suficcient to know that reads from the device
+		elif l:
-    # will return this many frames. Each frame being 2 bytes long.
+			f.write(data)
-    # This means that the reads below will return either 320 bytes of data
+			time.sleep(.001)
-    # or 0 bytes of data. The latter is possible because we are in nonblocking
-    # mode.
-    inp.setperiodsize(160)
-    
-    loops = 1000000
-    while loops > 0:
-        loops -= 1
-        # Read data from device
-        l, data = inp.read()
-      
-        if l:
-            f.write(data)
-            time.sleep(.001)

setup.py

@@ -8,7 +8,7 @@ from setuptools import setup
 from setuptools.extension import Extension
 from sys import version
 
-pyalsa_version = '0.8.4'
+pyalsa_version = '0.11.0'
 
 if __name__ == '__main__':
     setup(
@@ -29,12 +29,12 @@ if __name__ == '__main__':
             'License :: OSI Approved :: Python Software Foundation License',
             'Operating System :: POSIX :: Linux',
             'Programming Language :: Python :: 2',
-            'Programming Language :: Python :: 3',    
+            'Programming Language :: Python :: 3',
             'Topic :: Multimedia :: Sound/Audio',
             'Topic :: Multimedia :: Sound/Audio :: Mixers',
             'Topic :: Multimedia :: Sound/Audio :: Players',
             'Topic :: Multimedia :: Sound/Audio :: Capture/Recording',
             ],
-        ext_modules=[Extension('alsaaudio',['alsaaudio.c'], 
+        ext_modules=[Extension('alsaaudio',['alsaaudio.c'],
                                libraries=['asound'])]
     )

test.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 # These are internal tests. They shouldn't fail, but they don't cover all
 # of the ALSA API. Most importantly PCM.read and PCM.write are missing.
@@ -10,127 +11,177 @@
 import unittest
 import alsaaudio
 import warnings
+from contextlib import closing
 
 # we can't test read and write well - these are tested otherwise
-PCMMethods = [('pcmtype', None),
+PCMMethods = [
-              ('pcmmode', None),
+	('pcmtype', None),
-              ('cardname', None),
+	('pcmmode', None),
-              ('setchannels', (2,)), 
+	('cardname', None)
-              ('setrate', (44100,)),
+]
-              ('setformat', (alsaaudio.PCM_FORMAT_S8,)),
+
-              ('setperiodsize', (320,))]
+PCMDeprecatedMethods = [
+	('setchannels', (2,)),
+	('setrate', (44100,)),
+	('setformat', (alsaaudio.PCM_FORMAT_S8,)),
+	('setperiodsize', (320,))
+]
 
 # 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),
+				('mixer', None),
-                ('mixerid', None),
+				('mixerid', None),
-                ('switchcap', None),
+				('switchcap', None),
-                ('volumecap', None),
+				('volumecap', None),
-                ('getvolume', None),
+				('getvolume', None),
-                ('getrange', None),
+				('getrange', None),
-                ('getenum', None),
+				('getenum', None),
-#                ('getmute', None),
+#				('getmute', None),
-#                ('getrec', None),
+#				('getrec', None),
-#                ('setvolume', (60,)),
+#				('setvolume', (60,)),
-#                ('setmute', (0,))
+#				('setmute', (0,))
-#                ('setrec', (0')),
+#				('setrec', (0')),
-                ]
+				]
 
 class MixerTest(unittest.TestCase):
-    """Test Mixer objects"""
+	"""Test Mixer objects"""
 
-    def testMixer(self):
+	def testMixer(self):
-        """Open the default Mixers and the Mixers on every card"""
+		"""Open the default Mixers and the Mixers on every card"""
-        
-        for c in alsaaudio.card_indexes():
-            mixers = alsaaudio.mixers(cardindex=c)
-            
-            for m in mixers:
-                mixer = alsaaudio.Mixer(m, cardindex=c)
-                mixer.close()
 
-    def testMixerAll(self):
+		for c in alsaaudio.card_indexes():
-        "Run common Mixer methods on an open object"
+			mixers = alsaaudio.mixers(cardindex=c)
 
-        mixers = alsaaudio.mixers()
+			for m in mixers:
-        mixer = alsaaudio.Mixer(mixers[0])
+				mixer = alsaaudio.Mixer(m, cardindex=c)
+				mixer.close()
 
-        for m, a in MixerMethods:
+	def testMixerAll(self):
-            f = alsaaudio.Mixer.__dict__[m]
+		"Run common Mixer methods on an open object"
-            if a is None:
-                f(mixer)
-            else:
-                f(mixer, *a)
 
-        mixer.close()
+		mixers = alsaaudio.mixers()
+		mixer = alsaaudio.Mixer(mixers[0])
 
-    def testMixerClose(self):
+		for m, a in MixerMethods:
-        """Run common Mixer methods on a closed object and verify it raises an 
+			f = alsaaudio.Mixer.__dict__[m]
-        error"""
+			if a is None:
+				f(mixer)
+			else:
+				f(mixer, *a)
 
-        mixers = alsaaudio.mixers()
+		mixer.close()
-        mixer = alsaaudio.Mixer(mixers[0])
-        mixer.close()
 
-        for m, a in MixerMethods:
+	def testMixerClose(self):
-            f = alsaaudio.Mixer.__dict__[m]
+		"""Run common Mixer methods on a closed object and verify it raises an
-            if a is None:
+		error"""
-                self.assertRaises(alsaaudio.ALSAAudioError, f, mixer)
+
-            else:
+		mixers = alsaaudio.mixers()
-                self.assertRaises(alsaaudio.ALSAAudioError, f, mixer, *a)
+		mixer = alsaaudio.Mixer(mixers[0])
+		mixer.close()
+
+		for m, a in MixerMethods:
+			f = alsaaudio.Mixer.__dict__[m]
+			if a is None:
+				self.assertRaises(alsaaudio.ALSAAudioError, f, mixer)
+			else:
+				self.assertRaises(alsaaudio.ALSAAudioError, f, mixer, *a)
 
 class PCMTest(unittest.TestCase):
-    """Test PCM objects"""
+	"""Test PCM objects"""
 
-    def testPCM(self):
+	def testPCM(self):
-        "Open a PCM object on every card"
+		"Open a PCM object on every card"
 
-        for c in alsaaudio.card_indexes():
+		for c in alsaaudio.card_indexes():
-            pcm = alsaaudio.PCM(cardindex=c)
+			pcm = alsaaudio.PCM(cardindex=c)
-            pcm.close()
+			pcm.close()
 
-    def testPCMAll(self):
+	def testPCMAll(self):
-        "Run all PCM methods on an open object"
+		"Run all PCM methods on an open object"
 
-        pcm = alsaaudio.PCM()
+		pcm = alsaaudio.PCM()
 
-        for m, a in PCMMethods:
+		for m, a in PCMMethods:
-            f = alsaaudio.PCM.__dict__[m]
+			f = alsaaudio.PCM.__dict__[m]
-            if a is None:
+			if a is None:
-                f(pcm)
+				f(pcm)
-            else:
+			else:
-                f(pcm, *a)
+				f(pcm, *a)
 
-        pcm.close()
+		pcm.close()
 
-    def testPCMClose(self):
+	def testPCMClose(self):
-        "Run all PCM methods on a closed object and verify it raises an error"
+		"Run all PCM methods on a closed object and verify it raises an error"
 
-        pcm = alsaaudio.PCM()
+		pcm = alsaaudio.PCM()
-        pcm.close()
+		pcm.close()
 
-        for m, a in PCMMethods:
+		for m, a in PCMMethods:
-            f = alsaaudio.PCM.__dict__[m]
+			f = alsaaudio.PCM.__dict__[m]
-            if a is None:
+			if a is None:
-                self.assertRaises(alsaaudio.ALSAAudioError, f, pcm)
+				self.assertRaises(alsaaudio.ALSAAudioError, f, pcm)
-            else:
+			else:
-                self.assertRaises(alsaaudio.ALSAAudioError, f, pcm, *a)
+				self.assertRaises(alsaaudio.ALSAAudioError, f, pcm, *a)
+
+	def testPCMDeprecated(self):
+		with warnings.catch_warnings(record=True) as w:
+			# Cause all warnings to always be triggered.
+			warnings.simplefilter("always")
+
+			try:
+				pcm = alsaaudio.PCM(card='default')
+			except alsaaudio.ALSAAudioError:
+				pass
+
+			# Verify we got a DepreciationWarning
+			self.assertEqual(len(w), 1, "PCM(card='default') expected a warning" )
+			self.assertTrue(issubclass(w[-1].category, DeprecationWarning), "PCM(card='default') expected a DeprecationWarning")
+
+		for m, a in PCMDeprecatedMethods:
+			with warnings.catch_warnings(record=True) as w:
+				# Cause all warnings to always be triggered.
+				warnings.simplefilter("always")
+
+				pcm = alsaaudio.PCM()
+
+				f = alsaaudio.PCM.__dict__[m]
+				if a is None:
+					f(pcm)
+				else:
+					f(pcm, *a)
+
+				# Verify we got a DepreciationWarning
+				method = "%s%s" % (m, str(a))
+				self.assertEqual(len(w), 1, method + " expected a warning")
+				self.assertTrue(issubclass(w[-1].category, DeprecationWarning), method + " expected a DeprecationWarning")
+
+class PollDescriptorArgsTest(unittest.TestCase):
+	'''Test invalid args for polldescriptors_revents (takes a list of tuples of 2 integers)'''
+	def testArgsNoList(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents('foo')
+
+	def testArgsListButNoTuples(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents(['foo', 1])
+
+	def testArgsListButInvalidTuples(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents([('foo', 'bar')])
+
+	def testArgsListTupleWrongLength(self):
+		with closing(alsaaudio.PCM()) as pcm:
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents([(1, )])
+
+			with self.assertRaises(TypeError):
+				pcm.polldescriptors_revents([(1, 2, 3)])
 
-    def testPCMDeprecated(self):
-        with warnings.catch_warnings(record=True) as w:
-            # Cause all warnings to always be triggered.
-            warnings.simplefilter("always")
 
-            try:
-                pcm = alsaaudio.PCM(card='default')
-            except alsaaudio.ALSAAudioError:
-                pass
-                
-            # Verify we got a DepreciationWarning
-            assert len(w) == 1
-            assert issubclass(w[-1].category, DeprecationWarning)
-                
 if __name__ == '__main__':
-    unittest.main()
+	unittest.main()