Document the release process.

fix generate switch capabilities

.gitignore

@@ -4,6 +4,8 @@ MANIFEST
 doc/gh-pages/
 doc/html/
 doc/doctrees/
+doc/_build/
 gh-pages/
 build/
-dist/
+dist/
+.vscode/

CHANGES

@@ -1,3 +1,32 @@
+Version 0.8.6:
+- Added four methods to the 'PCM' class to allow users to get detailed information about the device:
+
+  - 'getformats()' returns a dictionary of name / value pairs, one for each of the card's
+    supported formats - e.g. '{"U8": 1, "S16_LE": 2}',
+  - 'getchannels()' returns a list of the supported channel numbers, e.g. '[1, 2]',
+  - 'getrates()' returns supported sample rates for the device, e.g. '[48000]',
+  - 'getratebounds()' returns the device's official minimum and maximum supported 
+    sample rates as a tuple, e.g. '(4000, 48000)'.
+  
+  (#82 contributed by @jdstmporter)
+
+- Prevent hang on close after capturing audio (#80 contributed by @daym)
+
+Version 0.8.5:
+- Return an empty string/bytestring when 'read()' detects an 
+  overrun. Previously the returned data was undefined (contributed by @jcea)
+
+- Unlimited setperiod buffer size when reading frames (contributed by @jcea)
+
+Version 0.8.4:
+- Fix Python3 API usage broken in 0.8.3
+
+Version 0.8.3:
+- Add DSD sample formats (contributed by @lintweaker)
+- Add Mixer.handleevents() to acknowledge events identified by select.poll (contributed by @PaulSD)
+- Add functions for listing cards and their names (contributed by @chrisdiamand)
+- Add a method for setting enums (contributed by @chrisdiamand)
+
 Version 0.8.2:
 - fix #3 (we cannot get the revision from git for pip installs)
 

doc/README.md

@@ -1,3 +1,22 @@
+# Make a new release
+
+Update the version in setup.py
+
+    pyalsa_version = '0.9.0'
+
+Commit and push the update.
+
+Create and push a tag naming the version (i.e. 0.9.0):
+
+    git tag 0.9.0
+    git push origin 0.9.0
+
+Upload the package:
+
+    python3 setup.py sdist
+
+Don't forget to update the documentation.
+
 # Publish the documentation
 
 The documentation is published through the `gh-pages` branch.

doc/libalsaaudio.rst

@@ -63,7 +63,6 @@ The :mod:`alsaaudio` module defines functions and classes for using ALSA.
    useful. If you want to see a list of available PCM devices, use :func:`pcms`
    instead.
    
-
 .. function:: mixers(cardindex=-1, device='default')
 
    List the available mixers. The arguments are:
@@ -108,7 +107,7 @@ PCM objects in :mod:`alsaaudio` can play or capture (record) PCM
 sound through speakers or a microphone. The PCM constructor takes the
 following arguments:
 
-.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)
+.. class:: PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, rate=44100, channels=2, format=PCM_FORMAT_S16_LE, periodsize=32, device='default', cardindex=-1)
 
    This class is used to represent a PCM device (either for playback and
    recording). The arguments are:
@@ -117,75 +116,14 @@ following arguments:
      (default).  
    * *mode* - can be either :const:`PCM_NONBLOCK`, or :const:`PCM_NORMAL`
      (default). 
-   * *device* - the name of the PCM device that should be used (for example
-     a value from the output of :func:`pcms`). The default value is
-     ``'default'``.
-   * *cardindex* - the card index. If this argument is given, the device name
-     is constructed as 'hw:*cardindex*' and
-     the `device` keyword argument is ignored.
-     ``0`` is the first hardware sound card.
-
-   This will construct a PCM object with these default settings:
-
-   * Sample format: :const:`PCM_FORMAT_S16_LE`
-   * Rate: 44100 Hz 
-   * Channels: 2
-   * Period size: 32 frames
-
-   *Changed in 0.8:*
+   * *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. 
+     The default value is :const:`PCM_FORMAT_S16_LE`.
    
-   - The `card` keyword argument is still supported,
-     but deprecated. Please use `device` instead.
-
-   - The keyword argument `cardindex` was added.
-
-   The `card` keyword is deprecated because it guesses the real ALSA
-   name of the card. This was always fragile and broke some legitimate usecases.
-   
-     
-PCM objects have the following methods:
-
-
-.. method:: PCM.pcmtype()
-
-   Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
-   :const:`PCM_PLAYBACK`.
-
-
-.. method:: PCM.pcmmode()
-
-   Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
-   :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
-
-
-.. method:: PCM.cardname()
-
-   Return the name of the sound card used by this PCM object.
-
-
-.. method:: PCM.setchannels(nchannels)
-
-   Used to set the number of capture or playback channels. Common
-   values are: ``1`` = mono, ``2`` = stereo, and ``6`` = full 6 channel audio.
-   Few sound cards support more than 2 channels
-
-
-.. method:: PCM.setrate(rate)
-
-   Set the sample rate in Hz for the device. Typical values are ``8000``
-   (mainly used for telephony), ``16000``, ``44100`` (CD quality),
-   ``48000`` and ``96000``.
-
-
-.. method:: PCM.setformat(format)
-
-   The sound *format* of the device. Sound format controls how the PCM device
-   interpret data for playback, and how data is encoded in captures.
-
-   The following formats are provided by ALSA:
-
    =========================  ===============
-          Format                Description
+        Format                Description
    =========================  ===============
    ``PCM_FORMAT_S8``          Signed 8 bit samples for each channel
    ``PCM_FORMAT_U8``          Signed 8 bit samples for each channel
@@ -215,15 +153,66 @@ PCM objects have the following methods:
    ``PCM_FORMAT_U24_3LE``     Unsigned 24 bit samples for each channel (Little Endian byte order in 3 bytes)
    ``PCM_FORMAT_U24_3BE``     Unsigned 24 bit samples for each channel (Big Endian byte order in 3 bytes)
    =========================  ===============
+
+   * *periodsize* - the period size in frames. Each write should consist of *periodsize* frames. The default value is 32.
+   * *device* - the name of the PCM device that should be used (for example
+     a value from the output of :func:`pcms`). The default value is
+     ``'default'``.
+   * *cardindex* - the card index. If this argument is given, the device name
+     is constructed as 'hw:*cardindex*' and
+     the `device` keyword argument is ignored.
+     ``0`` is the first hardware sound card.
+
+   This will construct a PCM object with the given settings.
+
+   *Changed in 0.9:*
+
+   - Added the optional named parameters `rate`, `channels`, `format` and `periodsize`.
+
+   *Changed in 0.8:*
    
+   - The `card` keyword argument is still supported,
+     but deprecated. Please use `device` instead.
+
+   - The keyword argument `cardindex` was added.
+
+   The `card` keyword is deprecated because it guesses the real ALSA
+   name of the card. This was always fragile and broke some legitimate usecases.
+   
+     
+PCM objects have the following methods:
+
+.. method:: PCM.pcmtype()
+
+   Returns the type of PCM object. Either :const:`PCM_CAPTURE` or
+   :const:`PCM_PLAYBACK`.
+
+
+.. method:: PCM.pcmmode()
+
+   Return the mode of the PCM object. One of :const:`PCM_NONBLOCK`,
+   :const:`PCM_ASYNC`, or :const:`PCM_NORMAL`
+
+
+.. method:: PCM.cardname()
+
+   Return the name of the sound card used by this PCM object.
+
+.. method:: PCM.setchannels(nchannels)
+
+   .. deprecated:: 0.9 Use the `channels` named argument to :func:`PCM`.
+
+.. method:: PCM.setrate(rate)
+
+   .. deprecated:: 0.9 Use the `rate` named argument to :func:`PCM`.
+
+.. method:: PCM.setformat(format)
+   
+   .. deprecated:: 0.9 Use the `format` named argument to :func:`PCM`.
 
 .. method:: PCM.setperiodsize(period)
 
-   Sets the actual period size in frames. Each write should consist of
-   exactly this number of frames, and each read will return this
-   number of frames (unless the device is in :const:`PCM_NONBLOCK` mode, in
-   which case it may return nothing at all)
-
+   .. deprecated:: 0.9 Use the `periodsize` named argument to :func:`PCM`.
 
 .. method:: PCM.read()
 

doc/pyalsaaudio.rst

@@ -75,7 +75,7 @@ development at the time - and neither are very feature complete.
 I wrote PyAlsaAudio to fill this gap. My long term goal is to have the module
 included in the standard Python library, but that looks currently unlikely.
 
-PyAlsaAudio hass full support for sound capture, playback of sound, as well as
+PyAlsaAudio has full support for sound capture, playback of sound, as well as
 the ALSA Mixer API.
 
 MIDI support is not available, and since I don't own any MIDI hardware, it's

isine.py

@@ -56,10 +56,7 @@ class SinePlayer(Thread):
     def __init__(self, frequency = 440.0):
         Thread.__init__(self)
         self.setDaemon(True)
-        self.device = alsaaudio.PCM()
-        self.device.setchannels(channels)
-        self.device.setformat(format)
-        self.device.setrate(sampling_rate)
+        self.device = alsaaudio.PCM(channels=channels, format=format, rate=sampling_rate)
         self.queue = Queue()
         self.change(frequency)
 

playbacktest.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 ## playbacktest.py
 ##
@@ -38,18 +39,11 @@ if __name__ == '__main__':
 
     f = open(args[0], 'rb')
 
-    # Open the device in playback mode. 
-    out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, device=device)
-
-    # Set attributes: Mono, 44100 Hz, 16 bit little endian frames
-    out.setchannels(1)
-    out.setrate(44100)
-    out.setformat(alsaaudio.PCM_FORMAT_S16_LE)
-
+    # Open the device in playback mode in Mono, 44100 Hz, 16 bit little endian frames
     # The period size controls the internal number of frames per period.
     # The significance of this parameter is documented in the ALSA api.
-    out.setperiodsize(160)
 
+    out = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device)
     # Read data from stdin
     data = f.read(320)
     while data:

playwav.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 # Simple test script that plays (some) wav files
 
@@ -9,57 +10,54 @@ import wave
 import getopt
 import alsaaudio
 
-def play(device, f):    
+def play(device, f):	
 
-    print('%d channels, %d sampling rate\n' % (f.getnchannels(),
-                                               f.getframerate()))
-    # Set attributes
-    device.setchannels(f.getnchannels())
-    device.setrate(f.getframerate())
+	format = None
 
-    # 8bit is unsigned in wav files
-    if f.getsampwidth() == 1:
-        device.setformat(alsaaudio.PCM_FORMAT_U8)
-    # Otherwise we assume signed data, little endian
-    elif f.getsampwidth() == 2:
-        device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
-    elif f.getsampwidth() == 3:
-        device.setformat(alsaaudio.PCM_FORMAT_S24_3LE)
-    elif f.getsampwidth() == 4:
-        device.setformat(alsaaudio.PCM_FORMAT_S32_LE)
-    else:
-        raise ValueError('Unsupported format')
+	# 8bit is unsigned in wav files
+	if f.getsampwidth() == 1:
+		format = alsaaudio.PCM_FORMAT_U8
+	# Otherwise we assume signed data, little endian
+	elif f.getsampwidth() == 2:
+		format = alsaaudio.PCM_FORMAT_S16_LE
+	elif f.getsampwidth() == 3:
+		format = alsaaudio.PCM_FORMAT_S24_3LE
+	elif f.getsampwidth() == 4:
+		format = alsaaudio.PCM_FORMAT_S32_LE
+	else:
+		raise ValueError('Unsupported format')
 
-    periodsize = f.getframerate() / 8
+	periodsize = f.getframerate() // 8
 
-    device.setperiodsize(periodsize)
-    
-    data = f.readframes(periodsize)
-    while data:
-        # Read data from stdin
-        device.write(data)
-        data = f.readframes(periodsize)
+	print('%d channels, %d sampling rate, format %d, periodsize %d\n' % (f.getnchannels(),
+																		 f.getframerate(),
+																		 format,
+																		 periodsize))
+
+	device = alsaaudio.PCM(channels=f.getnchannels(), rate=f.getframerate(), format=format, periodsize=periodsize, device=device)
+	
+	data = f.readframes(periodsize)
+	while data:
+		# Read data from stdin
+		device.write(data)
+		data = f.readframes(periodsize)
 
 
 def usage():
-    print('usage: playwav.py [-d <device>] <file>', file=sys.stderr)
-    sys.exit(2)
+	print('usage: playwav.py [-d <device>] <file>', file=sys.stderr)
+	sys.exit(2)
 
 if __name__ == '__main__':
 
-    device = 'default'
+	device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'd:')
-    for o, a in opts:
-        if o == '-d':
-            device = a
+	opts, args = getopt.getopt(sys.argv[1:], 'd:')
+	for o, a in opts:
+		if o == '-d':
+			device = a
 
-    if not args:
-        usage()
-        
-    f = wave.open(args[0], 'rb')
-    device = alsaaudio.PCM(device=device)
-
-    play(device, f)
-
-    f.close()
+	if not args:
+		usage()
+		
+	with wave.open(args[0], 'rb') as f:
+		play(device, f)

recordtest.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 ## recordtest.py
 ##
@@ -22,48 +23,42 @@ import getopt
 import alsaaudio
 
 def usage():
-    print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr)
-    sys.exit(2)
+	print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr)
+	sys.exit(2)
 
 if __name__ == '__main__':
 
-    device = 'default'
+	device = 'default'
 
-    opts, args = getopt.getopt(sys.argv[1:], 'd:')
-    for o, a in opts:
-        if o == '-d':
-            device = a
+	opts, args = getopt.getopt(sys.argv[1:], 'd:')
+	for o, a in opts:
+		if o == '-d':
+			device = a
 
-    if not args:
-        usage()
+	if not args:
+		usage()
 
-    f = open(args[0], 'wb')
+	f = open(args[0], 'wb')
 
-    # Open the device in nonblocking capture mode. The last argument could
-    # just as well have been zero for blocking mode. Then we could have
-    # left out the sleep call in the bottom of the loop
-    inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, device=device)
+	# 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.
+	# For our purposes, it is suficcient to know that reads from the device
+	# will return this many frames. Each frame being 2 bytes long.
+	# This means that the reads below will return either 320 bytes of data
+	# or 0 bytes of data. The latter is possible because we are in nonblocking
+	# mode.
+	inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, 
+		channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, 
+		periodsize=160, device=device)
 
-    # Set attributes: Mono, 44100 Hz, 16 bit little endian samples
-    inp.setchannels(1)
-    inp.setrate(44100)
-    inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
-
-    # The period size controls the internal number of frames per period.
-    # The significance of this parameter is documented in the ALSA api.
-    # For our purposes, it is suficcient to know that reads from the device
-    # will return this many frames. Each frame being 2 bytes long.
-    # This means that the reads below will return either 320 bytes of data
-    # or 0 bytes of data. The latter is possible because we are in nonblocking
-    # mode.
-    inp.setperiodsize(160)
-    
-    loops = 1000000
-    while loops > 0:
-        loops -= 1
-        # Read data from device
-        l, data = inp.read()
-      
-        if l:
-            f.write(data)
-            time.sleep(.001)
+	loops = 1000000
+	while loops > 0:
+		loops -= 1
+		# Read data from device
+		l, data = inp.read()
+	  
+		if l:
+			f.write(data)
+			time.sleep(.001)

setup.py

@@ -8,7 +8,7 @@ from setuptools import setup
 from setuptools.extension import Extension
 from sys import version
 
-pyalsa_version = '0.8.4'
+pyalsa_version = '0.9.0'
 
 if __name__ == '__main__':
     setup(

test.py

@@ -1,4 +1,5 @@
-#!/usr/bin/env python
+#!/usr/bin/env python3
+# -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*-
 
 # These are internal tests. They shouldn't fail, but they don't cover all
 # of the ALSA API. Most importantly PCM.read and PCM.write are missing.
@@ -12,125 +13,148 @@ import alsaaudio
 import warnings
 
 # we can't test read and write well - these are tested otherwise
-PCMMethods = [('pcmtype', None),
-              ('pcmmode', None),
-              ('cardname', None),
-              ('setchannels', (2,)), 
-              ('setrate', (44100,)),
-              ('setformat', (alsaaudio.PCM_FORMAT_S8,)),
-              ('setperiodsize', (320,))]
+PCMMethods = [
+	('pcmtype', None),
+	('pcmmode', None),
+	('cardname', None)
+]
+
+PCMDeprecatedMethods = [
+	('setchannels', (2,)), 
+	('setrate', (44100,)),
+	('setformat', (alsaaudio.PCM_FORMAT_S8,)),
+	('setperiodsize', (320,))
+]
 
 # A clever test would look at the Mixer capabilities and selectively run the
 # omitted tests, but I am too tired for that.
 
 MixerMethods = [('cardname', None),
-                ('mixer', None),
-                ('mixerid', None),
-                ('switchcap', None),
-                ('volumecap', None),
-                ('getvolume', None),
-                ('getrange', None),
-                ('getenum', None),
-#                ('getmute', None),
-#                ('getrec', None),
-#                ('setvolume', (60,)),
-#                ('setmute', (0,))
-#                ('setrec', (0')),
-                ]
+				('mixer', None),
+				('mixerid', None),
+				('switchcap', None),
+				('volumecap', None),
+				('getvolume', None),
+				('getrange', None),
+				('getenum', None),
+#				('getmute', None),
+#				('getrec', None),
+#				('setvolume', (60,)),
+#				('setmute', (0,))
+#				('setrec', (0')),
+				]
 
 class MixerTest(unittest.TestCase):
-    """Test Mixer objects"""
+	"""Test Mixer objects"""
 
-    def testMixer(self):
-        """Open the default Mixers and the Mixers on every card"""
-        
-        for c in alsaaudio.card_indexes():
-            mixers = alsaaudio.mixers(cardindex=c)
-            
-            for m in mixers:
-                mixer = alsaaudio.Mixer(m, cardindex=c)
-                mixer.close()
+	def testMixer(self):
+		"""Open the default Mixers and the Mixers on every card"""
+		
+		for c in alsaaudio.card_indexes():
+			mixers = alsaaudio.mixers(cardindex=c)
+			
+			for m in mixers:
+				mixer = alsaaudio.Mixer(m, cardindex=c)
+				mixer.close()
 
-    def testMixerAll(self):
-        "Run common Mixer methods on an open object"
+	def testMixerAll(self):
+		"Run common Mixer methods on an open object"
 
-        mixers = alsaaudio.mixers()
-        mixer = alsaaudio.Mixer(mixers[0])
+		mixers = alsaaudio.mixers()
+		mixer = alsaaudio.Mixer(mixers[0])
 
-        for m, a in MixerMethods:
-            f = alsaaudio.Mixer.__dict__[m]
-            if a is None:
-                f(mixer)
-            else:
-                f(mixer, *a)
+		for m, a in MixerMethods:
+			f = alsaaudio.Mixer.__dict__[m]
+			if a is None:
+				f(mixer)
+			else:
+				f(mixer, *a)
 
-        mixer.close()
+		mixer.close()
 
-    def testMixerClose(self):
-        """Run common Mixer methods on a closed object and verify it raises an 
-        error"""
+	def testMixerClose(self):
+		"""Run common Mixer methods on a closed object and verify it raises an 
+		error"""
 
-        mixers = alsaaudio.mixers()
-        mixer = alsaaudio.Mixer(mixers[0])
-        mixer.close()
+		mixers = alsaaudio.mixers()
+		mixer = alsaaudio.Mixer(mixers[0])
+		mixer.close()
 
-        for m, a in MixerMethods:
-            f = alsaaudio.Mixer.__dict__[m]
-            if a is None:
-                self.assertRaises(alsaaudio.ALSAAudioError, f, mixer)
-            else:
-                self.assertRaises(alsaaudio.ALSAAudioError, f, mixer, *a)
+		for m, a in MixerMethods:
+			f = alsaaudio.Mixer.__dict__[m]
+			if a is None:
+				self.assertRaises(alsaaudio.ALSAAudioError, f, mixer)
+			else:
+				self.assertRaises(alsaaudio.ALSAAudioError, f, mixer, *a)
 
 class PCMTest(unittest.TestCase):
-    """Test PCM objects"""
+	"""Test PCM objects"""
 
-    def testPCM(self):
-        "Open a PCM object on every card"
+	def testPCM(self):
+		"Open a PCM object on every card"
 
-        for c in alsaaudio.card_indexes():
-            pcm = alsaaudio.PCM(cardindex=c)
-            pcm.close()
+		for c in alsaaudio.card_indexes():
+			pcm = alsaaudio.PCM(cardindex=c)
+			pcm.close()
 
-    def testPCMAll(self):
-        "Run all PCM methods on an open object"
+	def testPCMAll(self):
+		"Run all PCM methods on an open object"
 
-        pcm = alsaaudio.PCM()
+		pcm = alsaaudio.PCM()
 
-        for m, a in PCMMethods:
-            f = alsaaudio.PCM.__dict__[m]
-            if a is None:
-                f(pcm)
-            else:
-                f(pcm, *a)
+		for m, a in PCMMethods:
+			f = alsaaudio.PCM.__dict__[m]
+			if a is None:
+				f(pcm)
+			else:
+				f(pcm, *a)
 
-        pcm.close()
+		pcm.close()
 
-    def testPCMClose(self):
-        "Run all PCM methods on a closed object and verify it raises an error"
+	def testPCMClose(self):
+		"Run all PCM methods on a closed object and verify it raises an error"
 
-        pcm = alsaaudio.PCM()
-        pcm.close()
+		pcm = alsaaudio.PCM()
+		pcm.close()
 
-        for m, a in PCMMethods:
-            f = alsaaudio.PCM.__dict__[m]
-            if a is None:
-                self.assertRaises(alsaaudio.ALSAAudioError, f, pcm)
-            else:
-                self.assertRaises(alsaaudio.ALSAAudioError, f, pcm, *a)
+		for m, a in PCMMethods:
+			f = alsaaudio.PCM.__dict__[m]
+			if a is None:
+				self.assertRaises(alsaaudio.ALSAAudioError, f, pcm)
+			else:
+				self.assertRaises(alsaaudio.ALSAAudioError, f, pcm, *a)
 
-    def testPCMDeprecated(self):
-        with warnings.catch_warnings(record=True) as w:
-            # Cause all warnings to always be triggered.
-            warnings.simplefilter("always")
+	def testPCMDeprecated(self):
+		with warnings.catch_warnings(record=True) as w:
+			# Cause all warnings to always be triggered.
+			warnings.simplefilter("always")
+
+			try:
+				pcm = alsaaudio.PCM(card='default')
+			except alsaaudio.ALSAAudioError:
+				pass
+				
+			# Verify we got a DepreciationWarning
+			self.assertEqual(len(w), 1, "PCM(card='default') expected a warning" )
+			self.assertTrue(issubclass(w[-1].category, DeprecationWarning), "PCM(card='default') expected a DeprecationWarning")
+
+		for m, a in PCMDeprecatedMethods:
+			with warnings.catch_warnings(record=True) as w:
+				# Cause all warnings to always be triggered.
+				warnings.simplefilter("always")
+
+				pcm = alsaaudio.PCM()
+
+				f = alsaaudio.PCM.__dict__[m]
+				if a is None:
+					f(pcm)
+				else:
+					f(pcm, *a)
+
+				# Verify we got a DepreciationWarning
+				method = "%s%s" % (m, str(a))
+				self.assertEqual(len(w), 1, method + " expected a warning")
+				self.assertTrue(issubclass(w[-1].category, DeprecationWarning), method + " expected a DeprecationWarning")
 
-            try:
-                pcm = alsaaudio.PCM(card='default')
-            except alsaaudio.ALSAAudioError:
-                pass
-                
-            # Verify we got a DepreciationWarning
-            assert len(w) == 1
-            assert issubclass(w[-1].category, DeprecationWarning)
-                
 if __name__ == '__main__':
-    unittest.main()
+	unittest.main()