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...

105 Commits
0.9.2 ... main

Author SHA1 Message Date
Pbopbo
b3d11582e0 Adds read_sw function to be able to read any size of frames. 2026-04-02 17:27:53 +02:00
Lars Immisch
2a2fa8f742 Update local build instructions wherever setup.py is greppable. 2025-12-10 23:05:14 +01:00
Lars Immisch
acc6f152a3 Adressing TODO in release.yml
Initial test with test.pypi.org was successful
2025-12-01 23:26:37 +01:00
Matteo Bernardini
0d6e0e7bb4 chore: use makefile for reference on building sdist and wheels 2025-12-01 23:26:37 +01:00
Matteo Bernardini
4f3c299e53 fix: drop py2 trove classifier 2025-12-01 23:26:37 +01:00
Matteo Bernardini
e78c145d9e ci: deploy on tags only 2025-12-01 23:26:37 +01:00
Matteo Bernardini
6acae81487 ci: fix permissions 2025-12-01 23:26:37 +01:00
Matteo Bernardini
5eed8adfe5 ci: install ALSA headers 2025-12-01 23:26:37 +01:00
Matteo Bernardini
e3e51cc34d ci: install uv 2025-12-01 23:26:37 +01:00
Matteo Bernardini
f6f53ce92a ci: automatically build docs 2025-12-01 23:26:37 +01:00
Matteo Bernardini
38d0882379 chore: split dependency groups 2025-12-01 23:26:37 +01:00
Matteo Bernardini
6822cffcc8 ci: remove unsupported setting 2025-12-01 23:26:37 +01:00
Matteo Bernardini
8c1e370e04 ci: auto-detect package manager to use 2025-12-01 23:26:37 +01:00
Matteo Bernardini
6d17bc92b7 ci: use legacy license definition to allow building on py3.9 2025-12-01 23:26:37 +01:00
Matteo Bernardini
3e41a9eb97 ci: enforce setuptools version 2025-12-01 23:26:37 +01:00
Matteo Bernardini
df8886dde1 ci: ensure alsa headers are installed 2025-12-01 23:26:37 +01:00
Matteo Bernardini
aac2843f73 chore: stub workflow for automatic docs deployment 2025-12-01 23:26:37 +01:00
Matteo Bernardini
7987986ee4 chore: top-level makefile for common dev tasks 2025-12-01 23:26:37 +01:00
Matteo Bernardini
ae93ddc48e ci: workflow to automate releases on tags 2025-12-01 23:26:37 +01:00
Matteo Bernardini
a48389b750 chore: make pyright happy on tests 2025-12-01 23:26:37 +01:00
Matteo Bernardini
0b8db5f46d fix: PCM constructor stubs not coherent with runtime 2025-12-01 23:26:37 +01:00
Matteo Bernardini
6b2c4c22db fix(docs): add sphinx dep, fix version retrieval 2025-12-01 23:26:37 +01:00
Matteo Bernardini
a7896ee069 fix: stub consistency to runtime 2025-12-01 23:26:37 +01:00
Matteo Bernardini
ed83b3e29b chore(deps): add mypy (for stubtest utility) 2025-12-01 23:26:37 +01:00
Matteo Bernardini
e96f550862 fix: use Final for enum constants 2025-12-01 23:26:37 +01:00
Matteo Bernardini
40f4647d8c fix: missing self in type stubs 2025-12-01 23:26:37 +01:00
Matteo Bernardini
3a8c53851d chore(deps): add pyright for type-checking 2025-12-01 23:26:37 +01:00
Matteo Bernardini
f6770e3a42 fix: add ALSAAudioError to type stubs 2025-12-01 23:26:37 +01:00
Matteo Bernardini
0ac28e9eea fix: include doc/ and examples/ in sdist 2025-12-01 23:26:37 +01:00
Matteo Bernardini
fdc5f3782e chore: modernise packaging.
- use pyproject.toml to define project metadata and setuptools-specific configuration, removing the need for a setup.py file
- move sources in conventional src/ directory
- rework stubs so that they're visible downstream. PEP-561 doesn't support top-level .pyi files, so I made a dummy package instead `alsaaudio-stubs`
2025-12-01 23:26:37 +01:00
Matteo Bernardini
6efef83429 chore: move unit tests to conventional directory 2025-12-01 23:26:37 +01:00
Matteo Bernardini
df5c2f4685 chore: move examples to dedicated directory 2025-12-01 23:26:37 +01:00
Lars Immisch
0b3f1f41c7 Prepare 0.11.0
Bump version and update changelog.
2024-05-30 23:15:27 +02:00
Lars Immisch
3e360b1bb7 Update MANIFEST.in for type hints 2024-05-14 20:56:51 +01:00
Lars Immisch
44ccbf839d Fix sphinx warning 2024-05-07 19:07:47 +02:00
Lars Immisch
2c2e43d3d1 Add type hints & docs 2024-05-07 19:07:47 +02:00
Lars Immisch
a142b70033 Reorder for consistency between alsapcm_methods and code
getchannels/setchannels should be together IMO
2024-05-07 19:07:47 +02:00
Lars Immisch
26ba938e04 Make commands optional 2024-04-19 18:32:38 +01:00
Lars Immisch
f5e9d52c74 Add missing attribute 2024-04-19 13:10:39 +01:00
Ville Viinikka
436c31f9fd Add nominal_bits and physical_bits info
Adds the information discussed in pull request #144
2024-03-13 10:59:11 +01:00
Ville Viinikka
eda913b203 Use correct sample bit width
snd_pcm_hw_params_get_sbits gives the number of significant bits, not
the actual number of bits stored. Change to snd_pcm_format_physical_width.

This fixes a bug where, for example on my hardware:
format = 'S32_LE'
significant bits = 24
physical bits = 32

the program will segfault because the allocated buffer is too small.
2024-02-20 18:31:00 +01:00
Lars Immisch
0aba948277 Whitespace cleanup.
I ended up using Visual Studio Code and did a global regex replace
` +\n` -> `\n` (StackOverflow)
2024-02-20 18:08:09 +01:00
Oswald Buddenhagen
9b7b767594 fix docu typo 2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
db87f2ced5 document new avail() and polldescriptors_revents() PCM functions
amends 43a94b3 and 5221311.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
f179db2d9b de-duplicate PCM.info() documentation
... and move the dumpinfo() docu.

amends 4e098da - clearly, i'm blind.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
420b538321 improve documentation of PCM c'tor and info() method
reformulate and redistribute the information, somewhat inspired by text
provided by Ronald van Elburg in response to issue #110.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
ae5c4aad9b add xrun handling to the examples
it's very primitive, but it shows adequately what can happen and what to
do about it minimally (that is, complain and move on).
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
d23b26b2e5 isine example: fix stereo handling (#42)
while it's usually not actually necessary to generate a stereo signal
(alsa's default plughw device will happily duplicate it for us), we
still do it for demo purposes, just because.

a more realistic demo would actually use numpy, as that's what the
library will most likely be used with, but anyway.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
1d63226e56 isine example: simplify calculations in generate()
i found them a tad hard to follow ...
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
664f81a777 isine example: simplify thread run loop
avoid code duplication.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
eb51d11619 isine example: actually play some tones
the thread in the background actually needs time to do something
sensible. this is most easily achieved by simply sleeping in the
foreground thread.

i addition to the 440 Hz tone, also play 1 kHz, to demonstrate how
the change() function is used.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
2f74e8e8a4 isine example: fix use of deprecated Thread.setDaemon() 2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
6f52de9da0 isine example: remove questionable setting of period size
there is no need to be pedantic about the period size, especially with
a blocking device. what's more, attempting to set it on an already
playing device would error out, and it would be rather counter-
productive to temporarily stop it.
2024-02-06 00:39:15 +01:00
Oswald Buddenhagen
8fb33ddd49 improve write() underrun handling, take 2
we *really* should not paper over underruns, as they require attention.
however, the previous attempt (c2a6b6e) caused an exception to be thrown
(see #130), which was a bit excessive, and was consequently reverted
(438e52e).

so instead we make the handling consistent with what we do in read():
return the verbatim -EPIPE in this case. this can be simply ignored, and
the next write will resume the stream, so this is mostly backwards-
compatible (the failing write will be discarded and would need
repeating, but that will just cause a skip after the interruption,
which does not seem particularly relevant).

as a drive-by, again stop using snd_pcm_recover(), as it still just
obfuscates the snd_pcm_prepare() call it does in the end.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen
691c1d9b23 fix return value of PCM.write() on success (#137)
the `else` branch of the return value handling cascade got lost in
commit 438e52e, leading to us returning None on success.

rather than restoring the old code exactly, delay the construction
of the final return code object. this is more consistent with
alsapcm_read() and overall nicer.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen
061c297f4b remove stray snd_pcm_prepare() call from alsapcm_write()
this came from 438e52e, which tried to partially revert c2a6b6e, but
inserted a chunk that actually belonged to alsapcm_drop(). the latter
does not need to be restored, as we now handle SND_PCM_STATE_SETUP prior
to reading/writing.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen
8ff3e169cd unbreak read buffer overrun handling
my commit c2a6b6e broke it big time; we'd now just paper over overruns.
:}

the previous handling was fundamentally correct, needing only two
adjustments:
- to recover from drop()/drain(), we need to call snd_pcm_prepare() when
  the stream state is SND_PCM_STATE_SETUP. notably, we must not do this
  when the state is SND_PCM_STATE_XRUN.
- we should error-check the unlikely case that the recovery from an xrun
  fails.

that way we now have two snd_pcm_prepare() call sites in read(), which
looks a bit messy, but it's actually correct.

as a drive-by, simplify the return value check of snd_pcm_prepare() -
values higher than zero are impossible.
2024-02-05 23:01:30 +01:00
Oswald Buddenhagen
7d9c16618b slightly clarify docu of read() wrt. underrun 2024-02-05 23:01:30 +01:00
Oswald Buddenhagen
16345a139a make alsapcm_read()'s return value preparation clearer
... by nesting the success case into the != -EPIPE block.
2024-02-05 23:01:30 +01:00
Lars Immisch
1c730123eb pre-release updates
- update CHANGES.md
- bump version in setup.py

[Revisionist Note] This is the end of the rewritten branch. The original
history can be found in the branch main-pre-rewrite.
2024-02-02 11:42:30 +01:00
Lars Immisch
0df2e0ee6f Ignore volume events if shairplay-sync is running 2024-02-02 11:36:58 +01:00
Lars Immisch
fe3fbe5376 loopback.py: bugfixes 2024-02-02 11:36:58 +01:00
Lars Immisch
42ca8acbad Add a (naive) loopback implementation (#132)
* WIP
* Open/close the playback device when idle.
  It takes a long time until it's stopped, though.
* open/close logic of playback device
* Fix opening logic, make period size divisible by 6
* Be less verbose in level info
* Extra argument for output mixer card index
  Sometimes, this cannot be deduced from the output device
* Better silence detection
* Run run_after_stop when idle on startup
2024-02-02 11:36:58 +01:00
Lars Immisch
522131123c Add PCM.polldescriptors_revents()
Will be used in the upcoming loopback implementation, but it is
worthwhile regardless.
2024-02-02 11:36:58 +01:00
Lars Immisch
43a94b3c62 Add PCM.avail()
Will be used in the upcoming loopback implementation, but it is
worthwhile regardless.
2024-02-02 11:36:58 +01:00
Lars Immisch
9637703ab5 Fix build (#133)
[Revisionist Note] This is a squashed commit formed from commits
f374adb, 3743cf5, and cd44517, still found in the main-pre-rewrite
branch. It incorporates a suggestion from PR #134.
2024-02-02 11:36:58 +01:00
Lars Immisch
438e52e3fc Restore previous behaviour of calling snd_pcm_prepare in case of XRUN (#131) 2024-02-02 11:36:58 +01:00
Lars Immisch
07ac637b1c Fix memory leaks in PCM.write() error paths on python3 2024-02-02 11:36:58 +01:00
Lars Immisch
bdca4dc061 Small improvement to VolumeForwarder 2024-02-02 11:36:58 +01:00
Lars Immisch
24eef474da Refactor loopback. SCNR.
The Reactor now takes a callable, and the loopback and volume forwarder
are now implemented as callable instances, which seemed the most
Pythonic solution.
2024-02-02 11:36:58 +01:00
Lars Immisch
24d26a5161 Better error logging and comments 2024-02-02 11:36:58 +01:00
Lars Immisch
f62e61f844 Add volume control forwarding
This needs the patches from (probably)
https://lkml.org/lkml/2021/3/1/419. They are already in the raspberry OS
kernel sources and the setup works on an RPi 4.
2024-02-02 11:36:58 +01:00
Lars Immisch
53f4f093e1 mixertest.py: print capture volume 2024-02-02 11:36:58 +01:00
Lars Immisch
82308f32ed Add a naive loopback implementation using select.poll()
It does work, though.
2024-02-02 11:36:58 +01:00
Lars Immisch
39d6acd3ac Handle events in alsamixer_getvolume. Closes #126
This issue can be worked around by calling mixer.handleevents() before
calling mixer.getvolume(), but it makes more sense to handle all events
before returning the volume.
2024-02-02 11:36:58 +01:00
Lars Immisch
c5153db0ac Whitespace fixes
- strip trailing whitespace in several files
- fix some indentation (tabs vs. spaces)
2024-02-02 11:36:58 +01:00
Lars Immisch
f25c8243dc Update changes for release
[Revisionist Note] This commit was originally c6a0c80, still available
on the main-pre-rewrite branch. The 0.10.0 tag used to point to it.
2024-02-02 11:33:22 +01:00
Lars Immisch
073d708bd1 Remove trailing whitespace in CHANGES.md 2024-02-02 09:52:11 +01:00
Oswald Buddenhagen
946694d263 add PCM.state() and associated enum values
in principle, the state is already available from info(), but that's a
rather heavy function for something one might want to query often.

a practical use case might be checking whether a playback stream is done
draining, for example.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
574f78939d add PCM.drain()
for playback, this allows making sure that all written frames are
played, without using an external delay.

in principle, it's also usable for capture, but there isn't really a
practical reason to do so, as simply discarding excess captured frames
has no real cost.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
17d171c1a5 make period count configurable
the period count is just as important for playback latency as the period
size, so it makes no sense to have only one of them configurable.

as a drive-by, fix up the handling of periods in info() & dumpinfo().
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
de2fc3c992 bump (minor) version
we're about to add new features.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
c2a6b6e583 reshuffle XRUN recovery somewhat
perform it prior to invoking read()/write() if necessary, not right
after a failure event. this makes things more uniform and predictable.

we don't use snd_pcm_recover() any more, as we used it only for the
EPIPE case anyway, which boils down to snd_pcm_prepare() exactly.
handling ESTRPIPE as well might be desirable, but that's a separate
consideration.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
da7d04e2fd reduce scope of GIL releases
it's pointless to enclose snd_pcm_close() and snd_pcm_pause(), as these
calls don't sleep.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
1c1af45a7f use data types closer to those of ALSA
this removes lots of casts around snd_pcm_hw_params_get_*() calls

we could go further with that to make the code clean if we enabled all
the warnings, but it doesn't seem worth the effort.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
9b773b48d6 purge pydoc from the source
it's been obsolete for a *long* time, and having it redundantly to the
rst sources is bad hygiene. it still contained some useful info, which
has been transplanted to the rst source in the previous commit.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
b05efa0ad6 add some best practices to the docu
addresses #110, among other things.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
4e098da908 add missing and update incorrect/outdated documentation
for clarity, this includes docs which were previously omitted
(presumably) intentionally, but mark them as comments.

the getrec() and getmute() functions' docs are moved around, so they
appear in pairs with their set*() counterparts, like the *volume() ones
already did.

notably, this also fixes the docu of PCM_FORMAT_U8, which closes #104.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
c266d302e0 improve terminology document
mention xruns, and rework the definition of periods: concentrate on
relevant information, and remove the misinformation about period size
reduction being not that bad (pedantically, an application could run
somewhat asynchronously to the interrupts by using some timer, and
therefore actually save some of the overhead, but why would one use a
small period size in the first place then?).

also, language and formatting fixes.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
b094ac096b formatting/language fixes in introduction document 2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
46b91980e0 unify line spacing in .rst files
one empty line, except for high-level sections, which get two.

while at it, trim whitespace on otherwise empty lines.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
9ab4f721d6 remove bogus markup from the documentation
the poll objects are linked properly in a different way, and the
footnote appears outdated.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
a967b7db78 drop some pointless comments from the tex => sphinx conversion
amends 5c2a00655.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
01a444ac21 add new high-speed samples rates
closes #89 (but alsa doesn't support 768khz yet).
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
8bcb7ba626 remove redundant snd_pcm_hw_params_any() call
we just called it (and even error-checked it) a few lines above.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
9dc0fc2fd3 fix deprecation warning about PyUnicode_AsUnicode()
converting to ascii for the purpose of comparison is inefficient.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
4318b63912 fix deprecation warning about PyEval_InitThreads()
PyEval_InitThreads is a no-op in since python 3.9.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
a7b9d617b2 fix crashes when accessing already closed devices
PCM.htimestamp() gets the usual exception emission,
Mixer.close() gets a "double invocation" check like PCM.close() has.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
379fc05b5e fix memory handling in mixer access error paths
in case of error, alsamixer_new() would leak the object, while
alsamixer_list() might crash due to a null pointer.

as a drive-by, make alsamixer_gethandle() `static`.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
dff8ef031f fix memory leaks in *_polldescriptors()
the calloc'd pollfd arrays were not freed.
2023-03-02 00:41:01 +01:00
Oswald Buddenhagen
8ea9470454 fix draining/closing, take 2
commit 8abf06be introduced a pause() prior to draining, in an attempt
to work around clearly broken pulseaudio client behavior for capture
streams (drain() is supposed to imply a stop).

but as the workaround was also applied to playback streams, it would
cause nasty "clicks", as the stream would (obviously) stop before being
resumed for draining.

but draining is actually pointless for capture streams, as we're closing
right afterwards, so the samples are lost anyway.

what's more, destructors are not supposed to wait for anything, so
draining in alsapcm_dealloc() was wrong to start with. so we remove it.
note that this is a minor behavior change, which is reflected by the
adjustment of the playback test to have an explicit close() at the end.

finally, close() was also affected by the pulseaudio bug (which was not
addressed before), so there we make draining exclusive to playback
streams.
2023-03-02 00:35:02 +01:00
Ronald van Elburg
19c9ba3ed9 Fix issue #104 : Update description of PCM_FORMAT_U8: Unsigned 8 bit samples for each channel 2022-11-27 01:56:36 +01:00
Ronald van Elburg
b2f0466dd2 First version documentation PCM.info() method. (#119)
* First version documentation PCM.info() method.

* Add reference to documentation to docstring for PCM.info() method.

* Add extra fields to info dict:
  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
and update documentation accordingly.

Co-authored-by: Ronald van Elburg <Ronald@SoundAppraisal.eu>
2022-11-26 19:08:32 +01:00
Lars Immisch
6317d9addc Extend name on get_enum (#114) 2022-05-23 09:37:40 +02:00
Lars Immisch
2432089759 Allow longer device names. Closes #114 2022-05-22 22:43:55 +02:00
28 changed files with 2354 additions and 942 deletions

40
.github/workflows/docs.yml vendored Normal file
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@@ -0,0 +1,40 @@
on:
workflow_dispatch:
push:
branches:
- main
concurrency:
group: "pages"
cancel-in-progress: false
jobs:
build:
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v5
- name: Install uv
uses: astral-sh/setup-uv@v6
- name: Install ALSA headers
run: sudo apt-get install libasound2-dev
- name: Build HTML documentation
run: make -C doc html
- name: Upload artifacts
uses: actions/upload-pages-artifact@v4
with:
path: doc/html
deploy:
runs-on: ubuntu-latest
needs: build
permissions:
contents: read
pages: write
id-token: write
environment:
name: github-pages
url: ${{ steps.deployment.outputs.page_url }}
steps:
- name: Deploy to GitHub Pages
id: deployment
uses: actions/deploy-pages@v4

55
.github/workflows/release.yml vendored Normal file
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@@ -0,0 +1,55 @@
# Adapted from: https://github.com/pypa/cibuildwheel/blob/main/examples/github-deploy.yml
on:
workflow_dispatch:
push:
tags: "*"
jobs:
build_wheels:
name: Build wheels on ${{ matrix.os }}
runs-on: ${{ matrix.os }}
strategy:
matrix:
os:
- ubuntu-24.04
- ubuntu-24.04-arm
steps:
- uses: actions/checkout@v5
- name: Install uv
uses: astral-sh/setup-uv@v6
- name: Build wheels
run: make build_wheels
- uses: actions/upload-artifact@v4
with:
name: cibw-wheels-${{ matrix.os }}-${{ strategy.job-index }}
path: ./wheelhouse/*.whl
build_sdist:
name: Build source distribution
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v5
- name: Install uv
uses: astral-sh/setup-uv@v6
- name: Build sdist
run: make build_sdist
- uses: actions/upload-artifact@v4
with:
name: cibw-sdist
path: dist/*.tar.gz
upload_pypi:
needs: [build_wheels, build_sdist]
runs-on: ubuntu-latest
environment: pypi
permissions:
id-token: write
steps:
- uses: actions/download-artifact@v5
with:
# unpacks all CIBW artifacts into dist/
pattern: cibw-*
path: dist
merge-multiple: true
- uses: pypa/gh-action-pypi-publish@release/v1

3
.gitignore vendored
View File

@@ -8,7 +8,8 @@ doc/_build/
gh-pages/
build/
dist/
wheelhouse/
.vscode/
/__pycache__/
/pyalsaaudio.egg-info/
*.egg-info/
*.raw

View File

@@ -1,3 +1,27 @@
# 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)

View File

@@ -1,5 +1,2 @@
include *.py
include CHANGES
include TODO
include LICENSE
recursive-include doc *.html *.gif *.png *.css *.py *.rst *.js *.json Makefile
graft doc/
graft examples/

16
Makefile Normal file
View File

@@ -0,0 +1,16 @@
.PHONY: lint test build_wheels build_sdist build
lint:
uv run --group lint pyright; \
uv run --group lint stubtest alsaaudio
test:
uv run tests/test.py
build_wheels:
uvx cibuildwheel==3.1.4
build_sdist:
uv build --sdist
build: build_sdist build_wheels

View File

@@ -46,14 +46,9 @@ First, get the sources and change to the source directory:
$ cd pyalsaaudio
```
Then, build:
Then, build and install:
```
$ python setup.py build
```
And install:
```
$ sudo python setup.py install
$ pip install .
```
# Using the API

22
ci/setup_cibuildwheel.sh Executable file
View File

@@ -0,0 +1,22 @@
#!/bin/sh
# NOTE: ensure to use the correct package manager depending on the manylinux
# image used.
if which yum; then
yum install -y alsa-lib-devel
exit
fi
if which apk; then
apk add alsa-lib-dev
exit
fi
if which apt-get; then
apt-get install -y libasound2-dev
exit
fi
echo "Cannot install ALSA headers! No package manager detected"
exit 1

View File

@@ -3,7 +3,7 @@
# You can set these variables from the command line.
SPHINXOPTS =
SPHINXBUILD = sphinx-build
SPHINXBUILD = uv run --group doc sphinx-build
PAPER =
# Internal variables.
@@ -70,9 +70,3 @@ linkcheck:
@echo
@echo "Link check complete; look for any errors in the above output " \
"or in linkcheck/output.txt."
gh-pages: html
cd gh-pages; git pull --rebase; cd ..; cp -r ./html/* gh-pages
publish: gh-pages
cd gh-pages; git commit -a; git push; cd ..

View File

@@ -1,28 +1,18 @@
# Make a new release
Update the version in setup.py
Create and push a tag naming the version (i.e. 0.11.1):
pyalsa_version = '0.9.0'
git tag 0.11.1
git push origin 0.11.1
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.
This should trigger a build via a github actions and publish pre-built binaries to pypi.org
# Publish the documentation
All commits to main should trigger a rebuild of the documentation.
## Historical background
The documentation is published through the `gh-pages` branch.
To publish the documentation, you need to clone the `gh-pages` branch of this repository into

View File

@@ -20,9 +20,8 @@
# import sys
# sys.path.insert(0, os.path.abspath('.'))
import sys
sys.path.insert(0, '..')
from setup import pyalsa_version
from importlib.metadata import version
pyalsa_version = version("pyalsaaudio")
# -- General configuration ------------------------------------------------
@@ -67,7 +66,7 @@ release = version
#
# This is also used if you do content translation via gettext catalogs.
# Usually you set "language" from the command line for these cases.
language = None
language = 'en'
# List of patterns, relative to source directory, that match files and
# directories to ignore when looking for source files.

View File

@@ -1,8 +1,3 @@
.. alsaaudio documentation documentation master file, created by
sphinx-quickstart on Thu Mar 30 23:52:21 2017.
You can adapt this file completely to your liking, but it should at least
contain the root `toctree` directive.
alsaaudio documentation
===================================================
@@ -19,14 +14,12 @@ Download
* `Download from pypi <https://pypi.python.org/pypi/pyalsaaudio>`_
Github
======
* `Repository <https://github.com/larsimmisch/pyalsaaudio/>`_
* `Bug tracker <https://github.com/larsimmisch/pyalsaaudio/issues>`_
Indices and tables
==================
@@ -34,5 +27,3 @@ Indices and tables
* :ref:`modindex`
* :ref:`search`

View File

@@ -5,35 +5,12 @@
.. module:: alsaaudio
:platform: Linux
.. % \declaremodule{builtin}{alsaaudio} % standard library, in C
.. % not standard, in C
.. moduleauthor:: Casper Wilstrup <cwi@aves.dk>
.. moduleauthor:: Lars Immisch <lars@ibp.de>
.. % Author of the module code;
The :mod:`alsaaudio` module defines functions and classes for using ALSA.
.. % ---- 3.1. ----
.. % For each function, use a ``funcdesc'' block. This has exactly two
.. % parameters (each parameters is contained in a set of curly braces):
.. % the first parameter is the function name (this automatically
.. % generates an index entry); the second parameter is the function's
.. % argument list. If there are no arguments, use an empty pair of
.. % curly braces. If there is more than one argument, separate the
.. % arguments with backslash-comma. Optional parts of the parameter
.. % list are contained in \optional{...} (this generates a set of square
.. % brackets around its parameter). Arguments are automatically set in
.. % italics in the parameter list. Each argument should be mentioned at
.. % least once in the description; each usage (even inside \code{...})
.. % should be enclosed in \var{...}.
.. function:: pcms(pcmtype=PCM_PLAYBACK)
.. function:: pcms(pcmtype: int = PCM_PLAYBACK) ->list[str]
List available PCM devices by name.
@@ -57,13 +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:
@@ -72,6 +55,8 @@ 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'``.
@@ -97,10 +82,11 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
changed. Since 0.8, this functions returns the mixers for the default
device, not the mixers for the first card.
.. function:: asoundlib_version()
.. function:: asoundlib_version() -> str
Return a Python string containing the ALSA version found.
.. _pcm-objects:
PCM Objects
@@ -110,10 +96,12 @@ PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
sound through speakers or a microphone. The PCM constructor takes the
following arguments:
.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, device='default', cardindex=-1)
.. class:: PCM(type: int = PCM_PLAYBACK, mode: int = PCM_NORMAL, rate: int = 44100, channels: int = 2,
format: int = PCM_FORMAT_S16_LE, periodsize: int = 32, periods: int = 4,
device: str = 'default', cardindex: int = -1) -> PCM
This class is used to represent a PCM device (either for playback and
recording). The arguments are:
This class is used to represent a PCM device (either for playback or
recording). The constructor's arguments are:
* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default).
@@ -128,7 +116,7 @@ following arguments:
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)
@@ -156,7 +144,11 @@ following arguments:
``PCM_FORMAT_U24_3BE`` Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
========================= ===============
* *periodsize* - the period size in frames. Each write should consist of *periodsize* frames. The default value is 32.
* *periodsize* - the period size in frames.
Make sure you understand :ref:`the meaning of periods <term-period>`.
The default value is 32, which is below the actual minimum of most devices,
and will therefore likely be larger in practice.
* *periods* - the number of periods in the buffer. The default value is 4.
* *device* - the name of the PCM device that should be used (for example
a value from the output of :func:`pcms`). The default value is
``'default'``.
@@ -165,7 +157,21 @@ following arguments:
the `device` keyword argument is ignored.
``0`` is the first hardware sound card.
This will construct a PCM object with the given settings.
**Note:** This should not be used, as it bypasses most of ALSA's configuration.
The defaults mentioned above are values passed by :mod:alsaaudio
to ALSA, not anything internal to ALSA.
**Note:** For default and non-default values alike, there is no
guarantee that a PCM device supports the requested configuration,
and ALSA may pick realizable values which it believes to be closest
to the request. Therefore, after creating a PCM object, it is
necessary to verify whether its realized configuration is acceptable.
The :func:info method can be used to query it.
*Changed in 0.10:*
- Added the optional named parameter `periods`.
*Changed in 0.9:*
@@ -181,42 +187,168 @@ following arguments:
The `card` keyword is deprecated because it guesses the real ALSA
name of the card. This was always fragile and broke some legitimate usecases.
PCM objects have the following methods:
.. method:: PCM.pcmtype()
.. 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()
.. method:: PCM.pcmmode() -> int
Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
:const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
.. method:: PCM.cardname()
.. method:: PCM.cardname() -> string
Return the name of the sound card used by this PCM object.
.. method:: PCM.setchannels(nchannels)
..
Omitted by intention due to not really fitting the c'tor-based setup concept:
.. method:: PCM.getchannels()
Returns list of the device's supported channel counts.
.. method:: PCM.getratebounds()
Returns the card's minimum and maximum supported sample rates as
a tuple of integers.
.. method:: PCM.getrates()
Returns the sample rates supported by the device.
The returned value can be of one of the following, depending on
the card's properties:
* Card supports only a single rate: returns the rate
* Card supports a continuous range of rates: returns a tuple of
the range's lower and upper bounds (inclusive)
* Card supports a collection of well-known rates: returns a list of
the supported rates
.. method:: PCM.getformats()
Returns a dictionary of supported format codes (integers) keyed by
their standard ALSA names (strings).
.. method:: PCM.setchannels(nchannels: int) -> int
.. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
.. method:: PCM.setrate(rate)
.. method:: PCM.setrate(rate: int) -> int
.. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
.. method:: PCM.setformat(format)
.. method:: PCM.setformat(format: int) -> int
.. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
.. method:: PCM.setperiodsize(period)
.. method:: PCM.setperiodsize(period: int) -> int
.. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
.. method:: PCM.read()
.. method:: PCM.state() -> int
Returs the current state of the stream, which can be one of
:const:`PCM_STATE_OPEN` (this should not actually happen),
:const:`PCM_STATE_SETUP` (after :func:`drop` or :func:`drain`),
:const:`PCM_STATE_PREPARED` (after construction),
:const:`PCM_STATE_RUNNING`,
:const:`PCM_STATE_XRUN`,
:const:`PCM_STATE_DRAINING`,
:const:`PCM_STATE_PAUSED`,
:const:`PCM_STATE_SUSPENDED`, and
:const:`PCM_STATE_DISCONNECTED`.
*New in 0.10*
.. method:: PCM.avail() -> int
For :const:`PCM_PLAYBACK` PCM objects, returns the number of writable
(that is, free) frames in the buffer.
For :const:`PCM_CAPTURE` PCM objects, returns the number of readable
(that is, filled) frames in the buffer.
An attempt to read/write more frames than indicated will block (in
:const:`PCM_NORMAL` mode) or fail and return zero (in
:const:`PCM_NONBLOCK` mode).
*New in 0.11*
.. method:: PCM.read() -> tuple[int, bytes]
In :const:`PCM_NORMAL` mode, this function blocks until a full period is
available, and then returns a tuple (length,data) where *length* is
@@ -228,64 +360,110 @@ PCM objects have the following methods:
``(0,'')`` if no new period has become available since the last
call to read.
In case of an overrun, this function will return a negative size: :const:`-EPIPE`.
This indicates that data was lost, even if the operation itself succeeded.
Try using a larger periodsize.
In case of a buffer overrun, this function will return the negative
size :const:`-EPIPE`, and no data is read.
This indicates that data was lost. To resume capturing, just call read
again, but note that the stream was already corrupted.
To avoid the problem in the future, try using a larger period size
and/or more periods, at the cost of higher latency.
.. method:: PCM.write(data)
.. method:: PCM.write(data: bytes) -> int
Writes (plays) the sound in data. The length of data *must* be a
multiple of the frame size, and *should* be exactly the size of a
period. If less than 'period size' frames are provided, the actual
playout will not happen until more data is written.
If the device is not in :const:`PCM_NONBLOCK` mode, this call will block if
the kernel buffer is full, and until enough sound has been played
to allow the sound data to be buffered. The call always returns the
size of the data provided.
If the data was successfully written, the call returns the size of the
data *in frames*.
If the device is not in :const:`PCM_NONBLOCK` mode, this call will block
if the kernel buffer is full, and until enough sound has been played
to allow the sound data to be buffered.
In :const:`PCM_NONBLOCK` mode, the call will return immediately, with a
return value of zero, if the buffer is full. In this case, the data
should be written at a later time.
should be written again at a later time.
In case of a buffer underrun, this function will return the negative
size :const:`-EPIPE`, and no data is written.
At this point, the playback was already corrupted. If you want to play
the data nonetheless, call write again with the same data.
To avoid the problem in the future, try using a larger period size
and/or more periods, at the cost of higher latency.
.. method:: PCM.pause([enable=True])
Note that this call completing means only that the samples were buffered
in the kernel, and playout will continue afterwards. Make sure that the
stream is drained before discarding the PCM handle.
.. method:: PCM.pause([enable: int = True]) -> int
If *enable* is :const:`True`, playback or capture is paused.
Otherwise, playback/capture is resumed.
.. method:: PCM.drop() -> int
.. method:: PCM.polldescriptors()
Stop the stream and drop residual buffered frames.
Returns a tuple of *(file descriptor, eventmask)* that can be used to
wait for changes on the PCM with *select.poll*.
*New in 0.9*
.. method:: PCM.drain() -> int
For :const:`PCM_PLAYBACK` PCM objects, play residual buffered frames
and then stop the stream. In :const:`PCM_NORMAL` mode,
this function blocks until all pending playback is drained.
For :const:`PCM_CAPTURE` PCM objects, this function is not very useful.
*New in 0.10*
.. method:: PCM.close() -> None
Closes the PCM device.
For :const:`PCM_PLAYBACK` PCM objects in :const:`PCM_NORMAL` mode,
this function blocks until all pending playback is drained.
.. method:: PCM.polldescriptors() -> list[tuple[int, int]]
Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the PCM with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module.
.. method:: PCM.set_tstamp_mode([mode=PCM_TSTAMP_ENABLE])
.. method:: PCM.polldescriptors_revents(descriptors: list[tuple[int, int]]) -> int
Processes the descriptor list returned by :func:`polldescriptors` after
using it with *select.poll*, and returns a single *eventmask* value that
is meaningful for deciding whether :func:`read` or :func:`write` should
be called.
*New in 0.11*
.. method:: PCM.set_tstamp_mode([mode: int = PCM_TSTAMP_ENABLE])
Set the ALSA timestamp mode on the device. The mode argument can be set to
either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
.. method:: PCM.get_tstamp_mode()
.. method:: PCM.get_tstamp_mode() -> int
Return the integer value corresponding to the ALSA timestamp mode. The
return value can be either :const:`PCM_TSTAMP_NONE` or :const:`PCM_TSTAMP_ENABLE`.
.. method:: PCM.set_tstamp_type([type=PCM_TSTAMP_TYPE_GETTIMEOFDAY])
.. method:: PCM.set_tstamp_type([type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY]) -> None
Set the ALSA timestamp mode on the device. The type argument
can be set to either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
.. method:: PCM.get_tstamp_type()
.. method:: PCM.get_tstamp_type() -> int
Return the integer value corresponding to the ALSA timestamp type. The
return value can be either :const:`PCM_TSTAMP_TYPE_GETTIMEOFDAY`,
:const:`PCM_TSTAMP_TYPE_MONOTONIC` or :const:`PCM_TSTAMP_TYPE_MONOTONIC_RAW`.
.. method:: PCM.htimestamp()
.. method:: PCM.htimestamp() -> tuple[int, int, int]
Return a Python tuple *(seconds, nanoseconds, frames_available_in_buffer)*.
@@ -312,16 +490,13 @@ PCM objects have the following methods:
update.
================================= ===========================================
__ poll_objects_
**A few hints on using PCM devices for playback**
The most common reason for problems with playback of PCM audio is that writes
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).
@@ -350,8 +525,7 @@ Mixer Objects
Mixer objects provides access to the ALSA mixer API.
.. class:: Mixer(control='Master', id=0, cardindex=-1, device='default')
.. class:: Mixer(control: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> Mixer
Arguments are:
@@ -378,23 +552,20 @@ Mixer objects provides access to the ALSA mixer API.
Mixer objects have the following methods:
.. method:: Mixer.cardname()
.. method:: Mixer.cardname() -> str
Return the name of the sound card used by this Mixer object
.. method:: Mixer.mixer()
.. method:: Mixer.mixer() -> str
Return the name of the specific mixer controlled by this object, For example
``'Master'`` or ``'PCM'``
.. method:: Mixer.mixerid()
.. method:: Mixer.mixerid() -> int
Return the ID of the ALSA mixer controlled by this object.
.. method:: Mixer.switchcap()
.. method:: Mixer.switchcap() -> int
Returns a list of the switches which are defined by this specific mixer.
Possible values in this list are:
@@ -414,8 +585,7 @@ Mixer objects have the following methods:
To manipulate these switches use the :meth:`setrec` or
:meth:`setmute` methods
.. method:: Mixer.volumecap()
.. method:: Mixer.volumecap() -> int
Returns a list of the volume control capabilities of this
mixer. Possible values in the list are:
@@ -431,7 +601,7 @@ Mixer objects have the following methods:
'Joined Capture Volume' Manipulate sound capture volume for all channels at a time
======================== ================
.. method:: Mixer.getenum()
.. method:: Mixer.getenum() -> tuple[str, list[str]]
For enumerated controls, return the currently selected item and the list of
items available.
@@ -457,48 +627,43 @@ Mixer objects have the following methods:
This method will return an empty tuple if the mixer is not an enumerated
control.
.. method:: Mixer.setenum(index: int) -> None
.. method:: Mixer.getmute()
For enumerated controls, sets the currently selected item.
*index* is an index into the list of available enumerated items returned
by :func:`getenum`.
Return a list indicating the current mute setting for each
channel. 0 means not muted, 1 means muted.
This method will fail if the mixer has no playback switch capabilities.
.. method:: Mixer.getrange(pcmtype=PCM_PLAYBACK)
.. method:: Mixer.getrange(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]
Return the volume range of the ALSA mixer controlled by this object.
The value is a tuple of integers whose meaning is determined by the
*units* argument.
The optional *pcmtype* argument can be either :const:`PCM_PLAYBACK` or
:const:`PCM_CAPTURE`, which is relevant if the mixer can control both
playback and capture volume. The default value is :const:`PCM_PLAYBACK`
if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.getrec()
Return a list indicating the current record mute setting for each channel. 0
means not recording, 1 means recording.
This method will fail if the mixer has no capture switch capabilities.
.. method:: Mixer.getvolume(pcmtype=PCM_PLAYBACK)
.. method:: Mixer.getvolume(pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int
Returns a list with the current volume settings for each channel. The list
elements are integer percentages.
elements are integers whose meaning is determined by the *units* argument.
The optional *pcmtype* argument can be either :const:`PCM_PLAYBACK` or
:const:`PCM_CAPTURE`, which is relevant if the mixer can control both
playback and capture volume. The default value is :const:`PCM_PLAYBACK`
if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.setvolume(volume, channel=None, pcmtype=PCM_PLAYBACK)
.. method:: Mixer.setvolume(volume: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None
Change the current volume settings for this mixer. The *volume* argument
controls the new volume setting as an integer percentage.
is an integer whose meaning is determined by the *units* argument.
If the optional argument *channel* is present, the volume is set
only for this channel. This assumes that the mixer can control the
@@ -509,7 +674,17 @@ Mixer objects have the following methods:
playback and capture volume. The default value is :const:`PCM_PLAYBACK`
if the mixer has playback channels, otherwise it is :const:`PCM_CAPTURE`.
.. method:: Mixer.setmute(mute, [channel])
The optional *units* argument can be one of :const:`VOLUME_UNITS_PERCENTAGE`,
:const:`VOLUME_UNITS_RAW`, or :const:`VOLUME_UNITS_DB`.
.. method:: Mixer.getmute() -> list[int]
Return a list indicating the current mute setting for each channel.
0 means not muted, 1 means muted.
This method will fail if the mixer has no playback switch capabilities.
.. method:: Mixer.setmute(mute: bool, channel: (int | None) = None) -> None
Sets the mute flag to a new value. The *mute* argument is either 0 for not
muted, or 1 for muted.
@@ -519,8 +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.
@@ -530,22 +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
wait for changes on the mixer with *select.poll*.
Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the mixer with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module.
__ poll_objects_
.. method:: Mixer.handleevents() -> int
.. method:: Mixer.handleevents()
Acknowledge events on the *polldescriptors* file descriptors
Acknowledge events on the :func:`polldescriptors` file descriptors
to prevent subsequent polls from returning the same events again.
Returns the number of events that were acknowledged.
.. method:: Mixer.close() -> None
Closes the Mixer device.
**A rant on the ALSA Mixer API**
The ALSA mixer API is extremely complicated - and hardly documented at all.
@@ -568,8 +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:
@@ -611,6 +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.
@@ -680,9 +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.
.. _poll_objects: http://docs.python.org/library/select.html#poll-objects

View File

@@ -7,26 +7,12 @@ 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
@@ -64,8 +50,8 @@ More information about ALSA may be found on the project homepage
ALSA and Python
===============
The older Linux sound API (OSS) which is now deprecated is well supported from
the standard Python library, through the ossaudiodev module. No native ALSA
The older Linux sound API (OSS) -- which is now deprecated -- is well supported
by the standard Python library, through the ossaudiodev module. No native ALSA
support exists in the standard library.
There are a few other "ALSA for Python" projects available, including at least
@@ -100,11 +86,8 @@ ship with ALSA kernels.
To install, execute the following: --- ::
$ python setup.py build
$ pip install .
And then as root: --- ::
# python setup.py install
*******
Testing

View File

@@ -19,7 +19,7 @@ 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
@@ -29,8 +29,8 @@ Frame
Frame size
This is the size in bytes of each frame. This can vary a lot: if each sample
is 8 bits, and we're handling mono sound, the frame size is one byte.
Similarly in 6 channel audio with 64 bit floating point samples, the frame
size is 48 bytes
For six channel audio with 64 bit floating point samples, the frame size
is 48 bytes.
Rate
PCM sound consists of a flow of sound frames. The sound rate controls how
@@ -38,7 +38,7 @@ Rate
means that a new frame is played or captured 8000 times per second.
Data rate
This is the number of bytes, which must be recorded or provided per
This is the number of bytes which must be consumed or provided per
second at a certain frame size and rate.
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
@@ -46,29 +46,60 @@ Data rate
At the other end of the scale, 96000 Hz, 6 channel sound with 64
bit (8 bytes) samples has a data rate of 96000 \* 6 \* 8 = 4608
kb/s (almost 5 MB sound data per second)
kb/s (almost 5 MB sound data per second).
If the data rate requirement is not met, an overrun (on capture) or
underrun (on playback) occurs; the term "xrun" is used to refer to
either event.
.. _term-period:
Period
When the hardware processes data this is done in chunks of frames. The time
interval between each processing (A/D or D/A conversion) is known
as the period.
The size of the period has direct implication on the latency of the
sound input or output. For low-latency the period size should be
very small, while low CPU resource usage would usually demand
larger period sizes. With ALSA, the CPU utilization is not impacted
much by the period size, since the kernel layer buffers multiple
periods internally, so each period generates an interrupt and a
memory copy, but userspace can be slower and read or write multiple
periods at the same time.
The CPU processes sample data in chunks of frames, so-called periods
(also called fragments by some systems). The operating system kernel's
sample buffer must hold at least two periods (at any given time, one
is processed by the sound hardware, and one by the CPU).
The completion of a *period* triggers a CPU interrupt, which causes
processing and context switching overhead. Therefore, a smaller period
size causes higher CPU resource usage at a given data rate.
A bigger size of the *buffer* improves the system's resilience to xruns.
The buffer being split into a bigger number of smaller periods also does
that, as it allows it to be drained / topped up sooner.
On the other hand, a bigger size of the *buffer* also increases the
playback latency, that is, the time it takes for a frame from being
sent out by the application to being actually audible.
Similarly, a bigger *period* size increases the capture latency.
The trade-off between latency, xrun resilience, and resource usage
must be made depending on the application.
Period size
This is the size of each period in Hz. *Not bytes, but Hz!.* In
:mod:`alsaaudio` the period size is set directly, and it is
This is the size of each period in frames. *Not bytes, but frames!*
In :mod:`alsaaudio` the period size is set directly, and it is
therefore important to understand the significance of this
number. If the period size is configured to for example 32,
each write should contain exactly 32 frames of sound data, and each
read will return either 32 frames of data or nothing at all.
.. _term-sample-size:
Sample size
Each sample takes *physical_bits* of space. *nominal_bits* tells
how many least significant bits are used. This is the bit depth
in the format name and sometimes called just *sample bits* or
*format width*. *significant_bits* tells how many most significant
bits of the *nominal_bits* are used by the sample. This can be thought
of as the *sample resolution*. This is visualized as follows::
MSB |00000000 XXXXXXXX XXXXXXXX 00000000| LSB
|--significant--|
|---------nominal---------|
|-------------physical--------------|
Once you understand these concepts, you will be ready to use the PCM API. Read
on.

View File

@@ -7,6 +7,7 @@
from __future__ import print_function
import sys
import time
from threading import Thread
from multiprocessing import Queue
@@ -15,14 +16,15 @@ if sys.version_info[0] < 3:
else:
from queue import Empty
from math import pi, sin
from math import pi, sin, ceil
import struct
import itertools
import alsaaudio
sampling_rate = 48000
format = alsaaudio.PCM_FORMAT_S16_LE
framesize = 2 # bytes per frame for the values above
pack_format = 'h' # short int, matching S16
channels = 2
def nearest_frequency(frequency):
@@ -34,28 +36,28 @@ def generate(frequency, duration = 0.125):
# generate a buffer with a sine wave of `frequency`
# that is approximately `duration` seconds long
# the buffersize we approximately want
target_size = int(sampling_rate * channels * duration)
# the length of a full sine wave at the frequency
cycle_size = int(sampling_rate / frequency)
# number of full cycles we can fit into target_size
factor = int(target_size / cycle_size)
# number of full cycles we can fit into the duration
factor = int(ceil(duration * frequency))
size = max(int(cycle_size * factor), 1)
# total number of frames
size = cycle_size * factor
sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \
for i in range(size)]
return struct.pack('%dh' % size, *sine)
if channels > 1:
sine = list(itertools.chain.from_iterable(itertools.repeat(x, channels) for x in sine))
return struct.pack(str(size * channels) + pack_format, *sine)
class SinePlayer(Thread):
def __init__(self, frequency = 440.0):
Thread.__init__(self)
self.setDaemon(True)
Thread.__init__(self, daemon=True)
self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
self.queue = Queue()
self.change(frequency)
@@ -79,16 +81,17 @@ class SinePlayer(Thread):
while True:
try:
buffer = self.queue.get(False)
self.device.setperiodsize(int(len(buffer) / framesize))
self.device.write(buffer)
except Empty:
pass
if buffer:
self.device.write(buffer)
if self.device.write(buffer) < 0:
print("Playback buffer underrun! Continuing nonetheless ...")
isine = SinePlayer()
isine.start()
def change(f):
isine.change(f)
time.sleep(1)
isine.change(1000)
time.sleep(1)

403
examples/loopback.py Normal file
View File

@@ -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()

View File

@@ -72,7 +72,12 @@ def show_mixer(name, kwargs):
volumes = mixer.getvolume()
volumes_dB = mixer.getvolume(units=alsaaudio.VOLUME_UNITS_DB)
for i in range(len(volumes)):
print("Channel %i volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
print("Channel %i playback volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
volumes = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE)
volumes_dB = mixer.getvolume(pcmtype=alsaaudio.PCM_CAPTURE, units=alsaaudio.VOLUME_UNITS_DB)
for i in range(len(volumes)):
print("Channel %i capture volume: %i%% (%.1f dB)" % (i, volumes[i], volumes_dB[i] / 100.0))
try:
mutes = mixer.getmute()

View File

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

View File

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

View File

@@ -59,6 +59,8 @@ if __name__ == '__main__':
# Read data from device
l, data = inp.read()
if l:
if l < 0:
print("Capture buffer overrun! Continuing nonetheless ...")
elif l:
f.write(data)
time.sleep(.001)

70
pyproject.toml Normal file
View File

@@ -0,0 +1,70 @@
[project]
name = "pyalsaaudio"
version = "0.11.0"
description = "ALSA bindings"
readme.text = """
This package contains wrappers for accessing the ALSA API from Python.
It is fairly complete for PCM devices and Mixer access.
"""
readme.content-type = "text/plain"
license.file = "LICENSE"
# NOTE: license as a table has been deprecated by PEP-639, however support in
# setuptools has been introduced in v77.0 which is not installable on Python
# 3.9. Since this project doesn't constraint the minimum python version, we have
# to stick to the legacy `license` definition for now.
# license="PSF-2.0"
# license-files=["LICENSE"]
classifiers = [
"Development Status :: 4 - Beta",
"Intended Audience :: Developers",
"Operating System :: POSIX :: Linux",
"Programming Language :: Python :: 3",
"Topic :: Multimedia :: Sound/Audio",
"Topic :: Multimedia :: Sound/Audio :: Mixers",
"Topic :: Multimedia :: Sound/Audio :: Players",
"Topic :: Multimedia :: Sound/Audio :: Capture/Recording",
]
[[project.authors]]
name = "Casper Wilstrup"
email = "cwi@aves.dk"
[[project.maintainers]]
name = "Lars Immisch"
email = "lars@ibp.de"
[project.urls]
homepage = "http://larsimmisch.github.io/pyalsaaudio/"
source = "https://github.com/larsimmisch/pyalsaaudio"
documentation = "https://larsimmisch.github.io/pyalsaaudio/"
issues = "https://github.com/larsimmisch/pyalsaaudio/issues"
[build-system]
requires = ["setuptools"]
build-backend = "setuptools.build_meta"
[tool.setuptools]
ext-modules = [
{ name = "alsaaudio", sources = ["src/alsaaudio.c"], libraries = ["asound"] }
]
[tool.cibuildwheel]
build-frontend = "build[uv]"
before-all = "ci/setup_cibuildwheel.sh"
[dependency-groups]
lint = [
"mypy>=1.17.1",
"pyright>=1.1.403",
]
doc = [
"sphinx>=8.1.3",
]

View File

@@ -1,40 +0,0 @@
#!/usr/bin/env python
'''This package contains wrappers for accessing the ALSA API from Python.
It is fairly complete for PCM devices and Mixer access.
'''
from setuptools import setup
from setuptools.extension import Extension
from sys import version
pyalsa_version = '0.9.2'
if __name__ == '__main__':
setup(
name = 'pyalsaaudio',
version = pyalsa_version,
description = 'ALSA bindings',
long_description = __doc__,
author = 'Casper Wilstrup',
author_email='cwi@aves.dk',
maintainer = 'Lars Immisch',
maintainer_email = 'lars@ibp.de',
license='PSF',
platforms=['posix'],
url='http://larsimmisch.github.io/pyalsaaudio/',
classifiers = [
'Development Status :: 4 - Beta',
'Intended Audience :: Developers',
'License :: OSI Approved :: Python Software Foundation License',
'Operating System :: POSIX :: Linux',
'Programming Language :: Python :: 2',
'Programming Language :: Python :: 3',
'Topic :: Multimedia :: Sound/Audio',
'Topic :: Multimedia :: Sound/Audio :: Mixers',
'Topic :: Multimedia :: Sound/Audio :: Players',
'Topic :: Multimedia :: Sound/Audio :: Capture/Recording',
],
ext_modules=[Extension('alsaaudio',['alsaaudio.c'],
libraries=['asound'])]
)

View File

@@ -0,0 +1,154 @@
from typing import Final, final
PCM_PLAYBACK: Final[int]
PCM_CAPTURE: Final[int]
PCM_NORMAL: Final[int]
PCM_NONBLOCK: Final[int]
PCM_ASYNC: Final[int]
PCM_FORMAT_S8: Final[int]
PCM_FORMAT_U8: Final[int]
PCM_FORMAT_S16_LE: Final[int]
PCM_FORMAT_S16_BE: Final[int]
PCM_FORMAT_U16_LE: Final[int]
PCM_FORMAT_U16_BE: Final[int]
PCM_FORMAT_S24_LE: Final[int]
PCM_FORMAT_S24_BE: Final[int]
PCM_FORMAT_U24_LE: Final[int]
PCM_FORMAT_U24_BE: Final[int]
PCM_FORMAT_S32_LE: Final[int]
PCM_FORMAT_S32_BE: Final[int]
PCM_FORMAT_U32_LE: Final[int]
PCM_FORMAT_U32_BE: Final[int]
PCM_FORMAT_FLOAT_LE: Final[int]
PCM_FORMAT_FLOAT_BE: Final[int]
PCM_FORMAT_FLOAT64_LE: Final[int]
PCM_FORMAT_FLOAT64_BE: Final[int]
PCM_FORMAT_MU_LAW: Final[int]
PCM_FORMAT_A_LAW: Final[int]
PCM_FORMAT_IMA_ADPCM: Final[int]
PCM_FORMAT_MPEG: Final[int]
PCM_FORMAT_GSM: Final[int]
PCM_FORMAT_S24_3LE: Final[int]
PCM_FORMAT_S24_3BE: Final[int]
PCM_FORMAT_U24_3LE: Final[int]
PCM_FORMAT_U24_3BE: Final[int]
PCM_TSTAMP_NONE: Final[int]
PCM_TSTAMP_ENABLE: Final[int]
PCM_TSTAMP_TYPE_GETTIMEOFDAY: Final[int]
PCM_TSTAMP_TYPE_MONOTONIC: Final[int]
PCM_TSTAMP_TYPE_MONOTONIC_RAW: Final[int]
PCM_FORMAT_DSD_U8: Final[int]
PCM_FORMAT_DSD_U16_LE: Final[int]
PCM_FORMAT_DSD_U32_LE: Final[int]
PCM_FORMAT_DSD_U32_BE: Final[int]
PCM_STATE_OPEN: Final[int]
PCM_STATE_SETUP: Final[int]
PCM_STATE_PREPARED: Final[int]
PCM_STATE_RUNNING: Final[int]
PCM_STATE_XRUN: Final[int]
PCM_STATE_DRAINING: Final[int]
PCM_STATE_PAUSED: Final[int]
PCM_STATE_SUSPENDED: Final[int]
PCM_STATE_DISCONNECTED: Final[int]
MIXER_CHANNEL_ALL: Final[int]
# NOTE: Omit for now - use case unknown (see alsaaudio.c)
#MIXER_SCHN_UNKNOWN: Final[int]
#MIXER_SCHN_FRONT_LEFT: Final[int]
#MIXER_SCHN_FRONT_RIGHT: Final[int]
#MIXER_SCHN_REAR_LEFT: Final[int]
#MIXER_SCHN_REAR_RIGHT: Final[int]
#MIXER_SCHN_FRONT_CENTER: Final[int]
#MIXER_SCHN_WOOFER: Final[int]
#MIXER_SCHN_SIDE_LEFT: Final[int]
#MIXER_SCHN_SIDE_RIGHT: Final[int]
#MIXER_SCHN_REAR_CENTER: Final[int]
#MIXER_SCHN_MONO: Final[int]
VOLUME_UNITS_PERCENTAGE: Final[int]
VOLUME_UNITS_RAW: Final[int]
VOLUME_UNITS_DB: Final[int]
def pcms(pcmtype: int) -> list[str]: ...
def cards() -> list[str]: ...
def mixers(cardindex: int = -1, device: str = 'default') -> list[str]: ...
def asoundlib_version() -> str: ...
def card_indexes() -> list[int]: ...
def card_name(index: int): ...
_DEPRECATED = ...
@final
class PCM:
def __init__(
self,
type: int = PCM_PLAYBACK,
mode: int = PCM_NORMAL,
device: str = "default",
cardindex: int = -1,
card: str = _DEPRECATED,
rate: int = 44100,
channels: int = 2,
format: int = PCM_FORMAT_S16_LE,
periodsize: int = 32,
periods: int = 4,
) -> None: ...
def close(self) -> None: ...
def dumpinfo(self) -> None: ...
def info(self) -> dict: ...
def state(self) -> int: ...
def htimestamp(self) -> tuple[int, int, int]: ...
def set_tstamp_mode(self, mode: int = PCM_TSTAMP_ENABLE) -> None: ...
def get_tstamp_mode(self) -> int: ...
def set_tstamp_type(self, type: int = PCM_TSTAMP_TYPE_GETTIMEOFDAY) -> None: ...
def get_tstamp_type(self) -> int: ...
def getformats(self) -> dict: ...
def getratebounds(self) -> tuple[int, int]: ...
def getrates(self) -> int | tuple[int, int] | list[int]: ...
def getchannels(self) -> list[int]: ...
def setchannels(self, nchannels: int) -> None: ...
def pcmtype(self) -> int: ...
def pcmmode(self) -> int: ...
def cardname(self) -> str: ...
def setrate(self, rate: int) -> None: ...
def setformat(self, format: int) -> int: ...
def setperiodsize(self, period: int) -> int: ...
def read(self) -> tuple[int, bytes]: ...
def write(self, data: bytes) -> int: ...
def avail(self) -> int: ...
def pause(self, enable: bool = True) -> int: ...
def drop(self) -> int: ...
def drain(self) -> int: ...
def polldescriptors(self) -> list[tuple[int, int]]: ...
def polldescriptors_revents(self, descriptors: list[tuple[int, int]]) -> int: ...
@final
class Mixer:
def __init__(self, control: str = 'Master', id: int = 0, cardindex: int = -1, device: str = 'default') -> None: ...
def cardname(self) -> str: ...
def close(self) -> None: ...
def mixer(self) -> str: ...
def mixerid(self) -> int: ...
def switchcap(self) -> int: ...
def volumecap(self) -> int: ...
def getvolume(self, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE) -> int: ...
def getrange(self, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_RAW) -> tuple[int, int]: ...
def getenum(self) -> tuple[str, list[str]]: ...
def getmute(self) -> list[int]: ...
def getrec(self) -> list[int]: ...
def setvolume(self, volume: int, pcmtype: int = PCM_PLAYBACK, units: int = VOLUME_UNITS_PERCENTAGE, channel: (int | None) = None) -> None: ...
def setenum(self, index: int) -> None: ...
def setmute(self, mute: bool, channel: (int | None) = None) -> None: ...
def setrec(self, capture: int, channel: (int | None) = None) -> None: ...
def polldescriptors(self) -> list[tuple[int, int]]: ...
def handleevents(self) -> int: ...
class ALSAAudioError(Exception): ...

File diff suppressed because it is too large Load Diff

View File

@@ -11,15 +11,18 @@
import unittest
import alsaaudio
import warnings
from contextlib import closing
MethodsType = list[tuple[str, tuple[int, ...] | None]]
# we can't test read and write well - these are tested otherwise
PCMMethods = [
PCMMethods: MethodsType = [
('pcmtype', None),
('pcmmode', None),
('cardname', None)
]
PCMDeprecatedMethods = [
PCMDeprecatedMethods: MethodsType = [
('setchannels', (2,)),
('setrate', (44100,)),
('setformat', (alsaaudio.PCM_FORMAT_S8,)),
@@ -29,7 +32,8 @@ PCMDeprecatedMethods = [
# A clever test would look at the Mixer capabilities and selectively run the
# omitted tests, but I am too tired for that.
MixerMethods = [('cardname', None),
MixerMethods: MethodsType = [
('cardname', None),
('mixer', None),
('mixerid', None),
('switchcap', None),
@@ -156,5 +160,31 @@ class PCMTest(unittest.TestCase):
self.assertEqual(len(w), 1, method + " expected a warning")
self.assertTrue(issubclass(w[-1].category, DeprecationWarning), method + " expected a DeprecationWarning")
class PollDescriptorArgsTest(unittest.TestCase):
'''Test invalid args for polldescriptors_revents (takes a list of tuples of 2 integers)'''
def testArgsNoList(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents('foo') # pyright: ignore[reportArgumentType]
def testArgsListButNoTuples(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents(['foo', 1]) # pyright: ignore[reportArgumentType]
def testArgsListButInvalidTuples(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents([('foo', 'bar')]) # pyright: ignore[reportArgumentType]
def testArgsListTupleWrongLength(self):
with closing(alsaaudio.PCM()) as pcm:
with self.assertRaises(TypeError):
pcm.polldescriptors_revents([(1, )]) # pyright: ignore[reportArgumentType]
with self.assertRaises(TypeError):
pcm.polldescriptors_revents([(1, 2, 3)]) # pyright: ignore[reportArgumentType]
if __name__ == '__main__':
unittest.main()

534
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