Don't use setrate etc. in samples.

Author: larsimmisch-adobe

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 @@ def usage():
 
     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 +10,52 @@
 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
+	print('%d channels, %d sampling rate\n' % (f.getnchannels(),
+											   f.getframerate()))
 
-    device.setperiodsize(periodsize)
-    
-    data = f.readframes(periodsize)
-    while data:
-        # Read data from stdin
-        device.write(data)
-        data = f.readframes(periodsize)
+	periodsize = f.getframerate() // 8
+
+	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'
-
-    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)
+	device = 'default'
 
-    play(device, f)
+	opts, args = getopt.getopt(sys.argv[1:], 'd:')
+	for o, a in opts:
+		if o == '-d':
+			device = a
 
-    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 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'
-
-    opts, args = getopt.getopt(sys.argv[1:], 'd:')
-    for o, a in opts:
-        if o == '-d':
-            device = a
+	device = 'default'
 
-    if not args:
-        usage()
+	opts, args = getopt.getopt(sys.argv[1:], 'd:')
+	for o, a in opts:
+		if o == '-d':
+			device = a
 
-    f = open(args[0], 'wb')
+	if not args:
+		usage()
 
-    # 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)
+	f = open(args[0], 'wb')
 
-    # 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
+	# 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)
 
-    # 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.extension import Extension
 from sys import version
 
-pyalsa_version = '0.8.6'
+pyalsa_version = '0.9.0'
 
 if __name__ == '__main__':
     setup(