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

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.
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

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

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
default in Linux kernel 2.6 and later. If you are using kernel version 2.4 you
may need to install the ALSA patches yourself - although most distributions
may need to install the ALSA patches yourself - although most distributions
ship with ALSA kernels.
To install, execute the following: --- ::

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.
Sample
PCM audio, whether it is input or output, consists of *samples*.
PCM audio, whether it is input or output, consists of *samples*.
A single sample represents the amplitude of one channel of sound
at a certain point in time. A lot of individual samples are
necessary to represent actual sound; for CD audio, 44100 samples
@@ -22,8 +22,8 @@ Sample
loudest signal that can be reproduced.
Frame
A frame consists of exactly one sample per channel. If there is only one
channel (Mono sound) a frame is simply a single sample. If the sound is
A frame consists of exactly one sample per channel. If there is only one
channel (Mono sound) a frame is simply a single sample. If the sound is
stereo, each frame consists of two samples, etc.
Frame size
@@ -33,7 +33,7 @@ Frame size
is 48 bytes.
Rate
PCM sound consists of a flow of sound frames. The sound rate controls how
PCM sound consists of a flow of sound frames. The sound rate controls how
often the current frame is replaced. For example, a rate of 8000 Hz
means that a new frame is played or captured 8000 times per second.

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

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

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