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/
+.vscode/

alsaaudio.c

@@ -1,4 +1,4 @@
-/* -*- Mode: C; c-basic-offset: 4; indent-tabs-mode: nil -*- */
+/* -*- Mode: C; c-basic-offset: 4; indent-tabs-mode: t -*- */
 
 /*
  * alsaaudio -- Python interface to ALSA (Advanced Linux Sound Architecture).
@@ -107,7 +107,7 @@ typedef struct {
 
 	// Configurable parameters
 	int channels;
-    int rate;
+	unsigned int rate;
 	int format;
 	snd_pcm_uframes_t periodsize;
 	int framesize;
@@ -370,8 +370,9 @@ static int alsapcm_setup(alsapcm_t *self)
 	   and fill it with configuration space */
 	snd_pcm_hw_params_alloca(&hwparams);
 	res = snd_pcm_hw_params_any(self->handle, hwparams);
-    if (res < 0)
+	if (res < 0) {
 		return res;
+	}
 
 	/* Fill it with default values.
 
@@ -386,10 +387,11 @@ static int alsapcm_setup(alsapcm_t *self)
 								   self->channels);
 
 	dir = 0;
-    snd_pcm_hw_params_set_rate(self->handle, hwparams, self->rate, dir);
-    snd_pcm_hw_params_set_period_size(self->handle, hwparams,
-                                      self->periodsize, dir);
-    snd_pcm_hw_params_set_periods(self->handle, hwparams, 4, 0);
+	unsigned int periods = 4;
+	snd_pcm_hw_params_set_rate_near(self->handle, hwparams, &self->rate, &dir);
+	snd_pcm_hw_params_set_period_size_near(self->handle, hwparams,
+										   &self->periodsize, &dir);
+	snd_pcm_hw_params_set_periods_near(self->handle, hwparams, &periods, &dir);
 
 	/* Write it to the device */
 	res = snd_pcm_hw_params(self->handle, hwparams);
@@ -421,11 +423,16 @@ alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds)
 	char *card = NULL;
 	int cardidx = -1;
 	char hw_device[128];
-    char *kw[] = { "type", "mode", "device", "cardindex", "card", NULL };
+	int rate = 44100;
+	int channels = 2;
+	int format = SND_PCM_FORMAT_S16_LE;
+	int periodsize = 32;
 
-    if (!PyArg_ParseTupleAndKeywords(args, kwds, "|Oisiz", kw,
+	char *kw[] = { "type", "mode", "device", "cardindex", "card", "rate", "channels", "format", "periodsize", NULL };
+
+	if (!PyArg_ParseTupleAndKeywords(args, kwds, "|Oisiziiii", kw,
 									 &pcmtypeobj, &pcmmode, &device,
-                                     &cardidx, &card))
+									 &cardidx, &card, &rate, &channels, &format, &periodsize))
 		return NULL;
 
 	if (cardidx >= 0) {
@@ -469,10 +476,10 @@ alsapcm_new(PyTypeObject *type, PyObject *args, PyObject *kwds)
 	self->handle = 0;
 	self->pcmtype = pcmtype;
 	self->pcmmode = pcmmode;
-    self->channels = 2;
-    self->rate = 44100;
-    self->format = SND_PCM_FORMAT_S16_LE;
-    self->periodsize = 32;
+	self->channels = channels;
+	self->rate = rate;
+	self->format = format;
+	self->periodsize = periodsize;
 
 	res = snd_pcm_open(&(self->handle), device, self->pcmtype,
 					   self->pcmmode);
@@ -687,7 +694,7 @@ alsapcm_getratemaxmin(alsapcm_t *self, PyObject *args)
 		return NULL;
 	}
 
-    unsigned min,max;
+	unsigned min, max;
 	if (snd_pcm_hw_params_get_rate_min(params, &min,NULL)<0) {
 		PyErr_SetString(ALSAAudioError, "Cannot get minimum supported bitrate");
 		return NULL;
@@ -887,6 +894,10 @@ alsapcm_setchannels(alsapcm_t *self, PyObject *args)
 		return NULL;
 	}
 
+	PyErr_WarnEx(PyExc_DeprecationWarning,
+				 "This function is deprecated. "
+				 "Please use the named parameter `channels` to `PCM()` instead", 1);
+
 	saved = self->channels;
 	self->channels = channels;
 	res = alsapcm_setup(self);
@@ -903,6 +914,8 @@ alsapcm_setchannels(alsapcm_t *self, PyObject *args)
 PyDoc_STRVAR(setchannels_doc,
 "setchannels(numchannels)\n\
 \n\
+Deprecated since 0.9\n\
+\n\
 Used to set the number of capture or playback channels. Common values\n\
 are: 1 = mono, 2 = stereo, and 6 = full 6 channel audio.\n\
 \n\
@@ -914,6 +927,7 @@ alsapcm_setrate(alsapcm_t *self, PyObject *args)
 {
 	int rate, saved;
 	int res;
+
 	if (!PyArg_ParseTuple(args,"i:setrate", &rate))
 		return NULL;
 
@@ -923,6 +937,10 @@ alsapcm_setrate(alsapcm_t *self, PyObject *args)
 		return NULL;
 	}
 
+	PyErr_WarnEx(PyExc_DeprecationWarning,
+				 "This function is deprecated. "
+				 "Please use the named parameter `rate` to `PCM()` instead", 1);
+
 	saved = self->rate;
 	self->rate = rate;
 	res = alsapcm_setup(self);
@@ -939,6 +957,8 @@ alsapcm_setrate(alsapcm_t *self, PyObject *args)
 PyDoc_STRVAR(setrate_doc,
 "setrate(rate)\n\
 \n\
+Deprecated since 0.9\n\
+\n\
 Set the sample rate in Hz for the device. Typical values are\n\
 8000 (telephony), 11025, 44100 (CD), 48000 (DVD audio) and 96000");
 
@@ -948,6 +968,7 @@ alsapcm_setformat(alsapcm_t *self, PyObject *args)
 {
 	int format, saved;
 	int res;
+
 	if (!PyArg_ParseTuple(args,"i:setformat", &format))
 		return NULL;
 				 
@@ -957,6 +978,10 @@ alsapcm_setformat(alsapcm_t *self, PyObject *args)
 		return NULL;
 	}
 
+	PyErr_WarnEx(PyExc_DeprecationWarning,
+				 "This function is deprecated. "
+				 "Please use the named parameter `format` to `PCM()` instead", 1);
+
 	saved = self->format;
 	self->format = format;
 	res = alsapcm_setup(self);
@@ -971,7 +996,9 @@ alsapcm_setformat(alsapcm_t *self, PyObject *args)
 }
 
 PyDoc_STRVAR(setformat_doc,
-"setformat(rate)\n");
+"setformat(rate)\n\
+\n\
+Deprecated since 0.9");
 
 static PyObject *
 alsapcm_setperiodsize(alsapcm_t *self, PyObject *args)
@@ -982,13 +1009,16 @@ alsapcm_setperiodsize(alsapcm_t *self, PyObject *args)
 	if (!PyArg_ParseTuple(args,"i:setperiodsize", &periodsize))
 		return NULL;
 
-    
 	if (!self->handle)
 	{
 		PyErr_SetString(ALSAAudioError, "PCM device is closed");
 		return NULL;
 	}
 
+	PyErr_WarnEx(PyExc_DeprecationWarning,
+				 "This function is deprecated. "
+				 "Please use the named parameter `periodsize` to `PCM()` instead", 1);
+
 	saved = self->periodsize;
 	self->periodsize = periodsize;
 	res = alsapcm_setup(self);
@@ -1009,6 +1039,8 @@ alsapcm_setperiodsize(alsapcm_t *self, PyObject *args)
 PyDoc_STRVAR(setperiodsize_doc,
 "setperiodsize(period) -> int\n\
 \n\
+Deprecated since 0.9\n\
+\n\
 Sets the actual period size in frames. Each write should consist of\n\
 exactly this number of frames, and each read will return this number of\n\
 frames (unless the device is in PCM_NONBLOCK mode, in which case it\n\
@@ -1830,43 +1862,43 @@ alsamixer_switchcap(alsamixer_t *self, PyObject *args)
 	}
 
 	result = PyList_New(0);
-    if (self->volume_cap & MIXER_CAP_SWITCH)
+	if (self->switch_cap & MIXER_CAP_SWITCH)
 	{
 		item = PyUnicode_FromString("Mute");
 		PyList_Append(result, item);
 		Py_DECREF(item);
 	}
-    if (self->volume_cap & MIXER_CAP_SWITCH_JOINED)
+	if (self->switch_cap & MIXER_CAP_SWITCH_JOINED)
 	{
 		item = PyUnicode_FromString("Joined Mute");
 		PyList_Append(result, item);
 		Py_DECREF(item);
 	}
-    if (self->volume_cap & MIXER_CAP_PSWITCH)
+	if (self->switch_cap & MIXER_CAP_PSWITCH)
 	{
 		item = PyUnicode_FromString("Playback Mute");
 		PyList_Append(result, item);
 		Py_DECREF(item);
 	}
-    if (self->volume_cap & MIXER_CAP_PSWITCH_JOINED)
+	if (self->switch_cap & MIXER_CAP_PSWITCH_JOINED)
 	{
 		item = PyUnicode_FromString("Joined Playback Mute");
 		PyList_Append(result, item);
 		Py_DECREF(item);
 	}
-    if (self->volume_cap & MIXER_CAP_CSWITCH)
+	if (self->switch_cap & MIXER_CAP_CSWITCH)
 	{
 		item = PyUnicode_FromString("Capture Mute");
 		PyList_Append(result, item);
 		Py_DECREF(item);
 	}
-    if (self->volume_cap & MIXER_CAP_CSWITCH_JOINED)
+	if (self->switch_cap & MIXER_CAP_CSWITCH_JOINED)
 	{
 		item = PyUnicode_FromString("Joined Capture Mute");
 		PyList_Append(result, item);
 		Py_DECREF(item);
 	}
-    if (self->volume_cap & MIXER_CAP_CSWITCH_EXCLUSIVE)
+	if (self->switch_cap & MIXER_CAP_CSWITCH_EXCLUSIVE)
 	{
 		item = PyUnicode_FromString("Capture Exclusive");
 		PyList_Append(result, item);

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:
@@ -117,72 +116,11 @@ 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:*
-   
-   - 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:
+   * *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`.
    
    =========================  ===============
         Format                Description
@@ -216,14 +154,65 @@ PCM objects have the following methods:
    ``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()
 

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
 
@@ -11,28 +12,29 @@ import alsaaudio
 
 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)
+		format = alsaaudio.PCM_FORMAT_U8
 	# Otherwise we assume signed data, little endian
 	elif f.getsampwidth() == 2:
-        device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
+		format = alsaaudio.PCM_FORMAT_S16_LE
 	elif f.getsampwidth() == 3:
-        device.setformat(alsaaudio.PCM_FORMAT_S24_3LE)
+		format = alsaaudio.PCM_FORMAT_S24_3LE
 	elif f.getsampwidth() == 4:
-        device.setformat(alsaaudio.PCM_FORMAT_S32_LE)
+		format = alsaaudio.PCM_FORMAT_S32_LE
 	else:
 		raise ValueError('Unsupported format')
 
 	periodsize = f.getframerate() // 8
 
-    device.setperiodsize(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:
@@ -57,9 +59,5 @@ if __name__ == '__main__':
 	if not args:
 		usage()
 		
-    f = wave.open(args[0], 'rb')
-    device = alsaaudio.PCM(device=device)
-
+	with wave.open(args[0], 'rb') as f:
 		play(device, f)
-
-    f.close()

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
 ##
@@ -39,16 +40,8 @@ if __name__ == '__main__':
 
 	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)
-
-    # Set attributes: Mono, 44100 Hz, 16 bit little endian samples
-    inp.setchannels(1)
-    inp.setrate(44100)
-    inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
-
+	# 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
@@ -56,7 +49,9 @@ 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.setperiodsize(160)
+	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
 	while loops > 0:

setup.py

@@ -8,7 +8,7 @@ from setuptools import setup
 from setuptools.extension import Extension
 from sys import version
 
-pyalsa_version = '0.8.6'
+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.
@@ -12,13 +13,18 @@ import alsaaudio
 import warnings
 
 # we can't test read and write well - these are tested otherwise
-PCMMethods = [('pcmtype', None),
+PCMMethods = [
+	('pcmtype', None),
 	('pcmmode', None),
-              ('cardname', None),
+	('cardname', None)
+]
+
+PCMDeprecatedMethods = [
 	('setchannels', (2,)), 
 	('setrate', (44100,)),
 	('setformat', (alsaaudio.PCM_FORMAT_S8,)),
-              ('setperiodsize', (320,))]
+	('setperiodsize', (320,))
+]
 
 # A clever test would look at the Mixer capabilities and selectively run the
 # omitted tests, but I am too tired for that.
@@ -129,8 +135,26 @@ class PCMTest(unittest.TestCase):
 				pass
 				
 			# Verify we got a DepreciationWarning
-            assert len(w) == 1
-            assert issubclass(w[-1].category, DeprecationWarning)
+			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")
 
 if __name__ == '__main__':
 	unittest.main()