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Whitespace cleanup.

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I ended up using Visual Studio Code and did a global regex replace
` +\n` -> `\n` (StackOverflow)
This commit is contained in:
Lars Immisch
2024-02-16 16:17:19 +01:00
parent 9b7b767594
commit 0aba948277
8 changed files with 40 additions and 40 deletions
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6
README.md
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@@ -2,7 +2,7 @@
For documentation, see http://larsimmisch.github.io/pyalsaaudio/ 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) > Maintainer: Lars Immisch (lars@ibp.de)
This package contains wrappers for accessing the 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 $ git clone https://github.com/larsimmisch/pyalsaaudio.git
$ cd pyalsaaudio $ cd pyalsaaudio
``` ```
Then, build: Then, build:
``` ```
$ python setup.py build $ python setup.py build
``` ```
And install: And install:
``` ```
$ sudo python setup.py install $ sudo python setup.py install

2
doc/README.md
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@@ -26,7 +26,7 @@ Don't forget to update the documentation.
The documentation is published through the `gh-pages` branch. 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 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 git clone -b gh-pages git@github.com:larsimmisch/pyalsaaudio.git gh-pages

42
doc/libalsaaudio.rst
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@@ -17,7 +17,7 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
Arguments are: Arguments are:
* *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK` * *pcmtype* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default). (default).
**Note:** **Note:**
@@ -102,12 +102,12 @@ following arguments:
recording). The constructor's arguments are: recording). The constructor's arguments are:
* *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK` * *type* - can be either :const:`PCM_CAPTURE` or :const:`PCM_PLAYBACK`
(default). (default).
* *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL` * *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
(default). (default).
* *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``. * *rate* - the sampling rate in Hz. Typical values are ``8000`` (mainly used for telephony), ``16000``, ``44100`` (default), ``48000`` and ``96000``.
* *channels* - the number of channels. The default value is 2 (stereo). * *channels* - the number of channels. The default value is 2 (stereo).
* *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures. * *format* - the data format. This controls how the PCM device interprets data for playback, and how data is encoded in captures.
The default value is :const:`PCM_FORMAT_S16_LE`. The default value is :const:`PCM_FORMAT_S16_LE`.
========================= =============== ========================= ===============
@@ -206,19 +206,19 @@ PCM objects have the following methods:
descriptions are prefixed with "hw:" below. descriptions are prefixed with "hw:" below.
=========================== ============================= ================================================================== =========================== ============================= ==================================================================
Key Description (Reference) Type Key Description (Reference) Type
=========================== ============================= ================================================================== =========================== ============================= ==================================================================
name PCM():device string name PCM():device string
card_no *index of card* integer (negative indicates device not associable with a card) card_no *index of card* integer (negative indicates device not associable with a card)
device_no *index of PCM device* integer device_no *index of PCM device* integer
subdevice_no *index of PCM subdevice* integer subdevice_no *index of PCM subdevice* integer
state *name of PCM state* string state *name of PCM state* string
access_type *name of PCM access type* string access_type *name of PCM access type* string
(call value) type PCM():type integer (call value) type PCM():type integer
(call value) type_name PCM():type string (call value) type_name PCM():type string
(call value) mode PCM():mode integer (call value) mode PCM():mode integer
(call value) mode_name PCM():mode string (call value) mode_name PCM():mode string
format PCM():format integer format PCM():format integer
format_name PCM():format string format_name PCM():format string
format_description PCM():format string format_description PCM():format string
subformat_name *name of PCM subformat* string subformat_name *name of PCM subformat* string
@@ -241,7 +241,7 @@ PCM objects have the following methods:
can_mmap_sample_resolution *hw: sample-resol. mmap* 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_pause *hw: pause* boolean (True: hardware supported)
can_resume *hw: resume* boolean (True: hardware supported) can_resume *hw: resume* boolean (True: hardware supported)
can_sync_start *hw: synchronized start* boolean (True: hardware supported) can_sync_start *hw: synchronized start* boolean (True: hardware supported)
=========================== ============================= ================================================================== =========================== ============================= ==================================================================
The italicized descriptions give a summary of the "full" description The italicized descriptions give a summary of the "full" description
@@ -346,7 +346,7 @@ PCM objects have the following methods:
In :const:`PCM_NORMAL` mode, this function blocks until a full period is In :const:`PCM_NORMAL` mode, this function blocks until a full period is
available, and then returns a tuple (length,data) where *length* is available, and then returns a tuple (length,data) where *length* is
the number of frames of captured data, and *data* is the captured 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. periodsize\*framesize bytes.
In :const:`PCM_NONBLOCK` mode, the call will not block, but will return In :const:`PCM_NONBLOCK` mode, the call will not block, but will return
@@ -422,7 +422,7 @@ PCM objects have the following methods:
Returns a list of tuples of *(file descriptor, eventmask)* that can be Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the PCM with *select.poll*. used to wait for changes on the PCM with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module. :const:`select` module.
.. method:: PCM.polldescriptors_revents(descriptors) .. method:: PCM.polldescriptors_revents(descriptors)
@@ -485,7 +485,7 @@ PCM objects have the following methods:
**A few hints on using PCM devices for playback** **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. 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 If too little data is written to the device, it will underrun, and
@@ -523,7 +523,7 @@ Mixer objects provides access to the ALSA mixer API.
Arguments are: Arguments are:
* *control* - specifies which control to manipulate using this mixer * *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 :mod:`alsaaudio`.\ :func:`mixers` function. The default value is
``'Master'`` - other common controls may be ``'Master Mono'``, ``'PCM'``, ``'Master'`` - other common controls may be ``'Master Mono'``, ``'PCM'``,
``'Line'``, etc. ``'Line'``, etc.
@@ -533,7 +533,7 @@ Mixer objects provides access to the ALSA mixer API.
* *cardindex* - specifies which card should be used. If this argument * *cardindex* - specifies which card should be used. If this argument
is given, the device name is constructed like this: 'hw:*cardindex*' and is given, the device name is constructed like this: 'hw:*cardindex*' and
the `device` keyword argument is ignored. ``0`` is the 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 * *device* - the name of the device on which the mixer resides. The default
value is ``'default'``. value is ``'default'``.
@@ -570,7 +570,7 @@ Mixer objects have the following methods:
'Joined Mute' This mixer can mute all channels at the same time 'Joined Mute' This mixer can mute all channels at the same time
'Playback Mute' This mixer can mute the playback output 'Playback Mute' This mixer can mute the playback output
'Joined Playback Mute' Mute playback for all channels at the same time} '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} 'Joined Capture Mute' Mute sound capture for all channels at a time}
'Capture Exclusive' Not quite sure what this is 'Capture Exclusive' Not quite sure what this is
====================== ================ ====================== ================
@@ -709,7 +709,7 @@ Mixer objects have the following methods:
Returns a list of tuples of *(file descriptor, eventmask)* that can be Returns a list of tuples of *(file descriptor, eventmask)* that can be
used to wait for changes on the mixer with *select.poll*. used to wait for changes on the mixer with *select.poll*.
The *eventmask* value is compatible with `poll.register`__ in the Python The *eventmask* value is compatible with `poll.register`__ in the Python
:const:`select` module. :const:`select` module.
.. method:: Mixer.handleevents() .. method:: Mixer.handleevents()
@@ -757,7 +757,7 @@ The following example are provided:
* `playbacktest.py` * `playbacktest.py`
* `mixertest.py` * `mixertest.py`
All examples (except `mixertest.py`) accept the commandline option All examples (except `mixertest.py`) accept the commandline option
*-c <cardname>*. *-c <cardname>*.
To determine a valid card name, use the commandline ALSA player:: To determine a valid card name, use the commandline ALSA player::
@@ -772,12 +772,12 @@ or::
>>> alsaaudio.pcms() >>> alsaaudio.pcms()
mixertest.py accepts the commandline options *-d <device>* and mixertest.py accepts the commandline options *-d <device>* and
*-c <cardindex>*. *-c <cardindex>*.
playwav.py playwav.py
~~~~~~~~~~ ~~~~~~~~~~
**playwav.py** plays a wav file. **playwav.py** plays a wav file.
To test PCM playback (on your default soundcard), run:: To test PCM playback (on your default soundcard), run::
@@ -825,7 +825,7 @@ The output might look like this::
'Mix' 'Mix'
'Mix Mono' '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:: that control; for example::
$ ./mixertest.py Master $ ./mixertest.py Master
@@ -834,7 +834,7 @@ that control; for example::
Channel 0 volume: 61% Channel 0 volume: 61%
Channel 1 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:: parameter on the mixer::
$ ./mixertest.py Master mute $ ./mixertest.py Master mute

2
doc/pyalsaaudio.rst
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@@ -81,7 +81,7 @@ and need the ALSA headers for compilation. Verify that you have
Naturally you also need to use a kernel with proper ALSA support. This is the 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 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. ship with ALSA kernels.
To install, execute the following: --- :: To install, execute the following: --- ::

8
doc/terminology.rst
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@@ -6,7 +6,7 @@ In order to use PCM devices it is useful to be familiar with some concepts and
terminology. terminology.
Sample 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 A single sample represents the amplitude of one channel of sound
at a certain point in time. A lot of individual samples are at a certain point in time. A lot of individual samples are
necessary to represent actual sound; for CD audio, 44100 samples necessary to represent actual sound; for CD audio, 44100 samples
@@ -22,8 +22,8 @@ Sample
loudest signal that can be reproduced. loudest signal that can be reproduced.
Frame 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. stereo, each frame consists of two samples, etc.
Frame size Frame size
@@ -33,7 +33,7 @@ Frame size
is 48 bytes. is 48 bytes.
Rate 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 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. means that a new frame is played or captured 8000 times per second.

12
isine.py
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@@ -44,18 +44,18 @@ def generate(frequency, duration = 0.125):
# total number of frames # total number of frames
size = cycle_size * factor size = cycle_size * factor
sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \ sine = [ int(32767 * sin(2 * pi * frequency * i / sampling_rate)) \
for i in range(size)] for i in range(size)]
if channels > 1: if channels > 1:
sine = list(itertools.chain.from_iterable(itertools.repeat(x, channels) for x in sine)) sine = list(itertools.chain.from_iterable(itertools.repeat(x, channels) for x in sine))
return struct.pack(str(size * channels) + pack_format, *sine) return struct.pack(str(size * channels) + pack_format, *sine)
class SinePlayer(Thread): class SinePlayer(Thread):
def __init__(self, frequency = 440.0): def __init__(self, frequency = 440.0):
Thread.__init__(self, daemon=True) Thread.__init__(self, daemon=True)
self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate) self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
@@ -75,7 +75,7 @@ class SinePlayer(Thread):
buf = generate(f) buf = generate(f)
self.queue.put(buf) self.queue.put(buf)
def run(self): def run(self):
buffer = None buffer = None
while True: while True:
@@ -86,7 +86,7 @@ class SinePlayer(Thread):
if buffer: if buffer:
if self.device.write(buffer) < 0: if self.device.write(buffer) < 0:
print("Playback buffer underrun! Continuing nonetheless ...") print("Playback buffer underrun! Continuing nonetheless ...")
isine = SinePlayer() isine = SinePlayer()
isine.start() isine.start()

2
playbacktest.py
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@@ -51,4 +51,4 @@ if __name__ == '__main__':
print("Playback buffer underrun! Continuing nonetheless ...") print("Playback buffer underrun! Continuing nonetheless ...")
data = f.read(320) data = f.read(320)
out.close() out.close()

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