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https://github.com/larsimmisch/pyalsaaudio.git
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196ca87a05
* fix draining/closing, take 2 commit8abf06beintroduced a pause() prior to draining, in an attempt to work around clearly broken pulseaudio client behavior for capture streams (drain() is supposed to imply a stop). but as the workaround was also applied to playback streams, it would cause nasty "clicks", as the stream would (obviously) stop before being resumed for draining. but draining is actually pointless for capture streams, as we're closing right afterwards, so the samples are lost anyway. what's more, destructors are not supposed to wait for anything, so draining in alsapcm_dealloc() was wrong to start with. so we remove it. note that this is a minor behavior change, which is reflected by the adjustment of the playback test to have an explicit close() at the end. finally, close() was also affected by the pulseaudio bug (which was not addressed before), so there we make draining exclusive to playback streams. * fix memory leaks in *_polldescriptors() the calloc'd pollfd arrays were not freed. * fix memory handling in mixer access error paths in case of error, alsamixer_new() would leak the object, while alsamixer_list() might crash due to a null pointer. as a drive-by, make alsamixer_gethandle() `static`. * fix crashes when accessing already closed devices PCM.htimestamp() gets the usual exception emission, Mixer.close() gets a "double invocation" check like PCM.close() has. * fix deprecation warning about PyEval_InitThreads() PyEval_InitThreads is a no-op in since python 3.9. * fix deprecation warning about PyUnicode_AsUnicode() converting to ascii for the purpose of comparison is inefficient. * remove redundant snd_pcm_hw_params_any() call we just called it (and even error-checked it) a few lines above. * add new high-speed samples rates closes #89 (but alsa doesn't support 768khz yet). * drop some pointless comments from the tex => sphinx conversion amends5c2a00655. * remove bogus markup from the documentation the poll objects are linked properly in a different way, and the footnote appears outdated. * unify line spacing in .rst files one empty line, except for high-level sections, which get two. while at it, trim whitespace on otherwise empty lines. * formatting/language fixes in introduction document * improve terminology document mention xruns, and rework the definition of periods: concentrate on relevant information, and remove the misinformation about period size reduction being not that bad (pedantically, an application could run somewhat asynchronously to the interrupts by using some timer, and therefore actually save some of the overhead, but why would one use a small period size in the first place then?). also, language and formatting fixes. * add missing and update incorrect/outdated documentation for clarity, this includes docs which were previously omitted (presumably) intentionally, but mark them as comments. the getrec() and getmute() functions' docs are moved around, so they appear in pairs with their set*() counterparts, like the *volume() ones already did. notably, this also fixes the docu of PCM_FORMAT_U8, which closes #104. * add some best practices to the docu addresses #110, among other things. * purge pydoc from the source it's been obsolete for a *long* time, and having it redundantly to the rst sources is bad hygiene. it still contained some useful info, which has been transplanted to the rst source in the previous commit. * use data types closer to those of ALSA this removes lots of casts around snd_pcm_hw_params_get_*() calls we could go further with that to make the code clean if we enabled all the warnings, but it doesn't seem worth the effort. * reduce scope of GIL releases it's pointless to enclose snd_pcm_close() and snd_pcm_pause(), as these calls don't sleep. * reshuffle XRUN recovery somewhat perform it prior to invoking read()/write() if necessary, not right after a failure event. this makes things more uniform and predictable. we don't use snd_pcm_recover() any more, as we used it only for the EPIPE case anyway, which boils down to snd_pcm_prepare() exactly. handling ESTRPIPE as well might be desirable, but that's a separate consideration. * bump (minor) version we're about to add new features. * make period count configurable the period count is just as important for playback latency as the period size, so it makes no sense to have only one of them configurable. as a drive-by, fix up the handling of periods in info() & dumpinfo(). * add PCM.drain() for playback, this allows making sure that all written frames are played, without using an external delay. in principle, it's also usable for capture, but there isn't really a practical reason to do so, as simply discarding excess captured frames has no real cost. * add PCM.state() and associated enum values in principle, the state is already available from info(), but that's a rather heavy function for something one might want to query often. a practical use case might be checking whether a playback stream is done draining, for example.
54 lines
1.4 KiB
Python
Executable File
54 lines
1.4 KiB
Python
Executable File
#!/usr/bin/env python3
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# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
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## playbacktest.py
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##
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## This is an example of a simple sound playback script.
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##
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## The script opens an ALSA pcm for sound playback. Set
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## various attributes of the device. It then reads data
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## from stdin and writes it to the device.
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##
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## To test it out do the following:
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## python recordtest.py out.raw # talk to the microphone
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## python playbacktest.py out.raw
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from __future__ import print_function
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import sys
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import time
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import getopt
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import alsaaudio
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def usage():
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print('usage: playbacktest.py [-d <device>] <file>', file=sys.stderr)
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sys.exit(2)
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if __name__ == '__main__':
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device = 'default'
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opts, args = getopt.getopt(sys.argv[1:], 'd:')
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for o, a in opts:
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if o == '-d':
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device = a
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if not args:
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usage()
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f = open(args[0], 'rb')
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# Open the device in playback mode in Mono, 44100 Hz, 16 bit little endian frames
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# The period size controls the internal number of frames per period.
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# The significance of this parameter is documented in the ALSA api.
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out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device)
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# Read data from stdin
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data = f.read(320)
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while data:
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out.write(data)
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data = f.read(320)
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out.close()
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