dot.asoundrc

# Seems reasonable quality, low CPU usage
defaults.pcm.rate_converter "speexrate_medium"

# Check that e.g. ut2004 still works, if default is redefined.
# Flash, etc. use this "default" entry.
pcm.!default {
	type plug
	# Would need to always output to all 6 channels, so the dmixer actually works if e.g. 6-channel is attempted to be started, while 2-channel is playing.
    slave.pcm "dmixed"
    hint.description "Dmix Audio"
#    slave.pcm "jack"
#    hint.description "Jack Audio"
}

pcm.dmixed  {
	type asym
	playback.pcm {
		# See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
		type dmix

		# Don't block other users, e.g. the Timidity midi-player daemon
		# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
		ipc_key_add_uid true

		ipc_key 5678293
		ipc_perm 0660
		ipc_gid audio
		# Don't put the rate here! Otherwise it resets the rate & channels set below, as shown by:  cat /proc/asound/card0/pcm0p/sub0/hw_params
		slave {
			# 2 for stereo, 6 for surround51, 8 for surround71
			channels 2
			pcm {
				# mplayer chooses S32_LE, but others usually S16_LE
				#format S32_LE
				format S16_LE

				# 44100 or 48000
				# 44100 for music, 48000 is compatible with most h/w
				#rate 44100
				rate 48000

				# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
				# Maybe helps
				nonblock true

				type hw
				card 1
				device 1
				subdevice 0
			}
			# mplayer2 chooses 1024
			# period_size 512 with buffer_size 16384 stops crackling in xmame
			# 320 breaks flash - https://bbs.archlinux.org/viewtopic.php?id=129458
			#period_size 512
			period_size 1024
			#period_size 512

			# 4096 might make sound crackle
			# mplayer2 chooses 8192. Half-Life 2 chooses 16384.
			# If too large, use CONFIG_SND_HDA_PREALLOC_SIZE=2048
			buffer_size 8192
		}
	}
	# apulse (a cut-down replacement for pulseaudio) needs dsnoop
	# https://bbs.archlinux.org/viewtopic.php?id=187258
	capture.pcm "dsnoop"
}

# If get error "Slave PCM not usable", then need to use plug:
# If get error "Cannot find rate converter", then install libsamplerate and alsa-plugins

# Lunar Linux:  lin ladspa-bs2b
# listplugins
# analyseplugin bs2b
pcm.bs2b {
	type ladspa
	path "/usr/lib/ladspa"
	plugins {
		0 {
			id 4221  # Bauer stereophonic-to-binaural (4221/bs2b)
			input {
				controls [ 700 6 ]
			}
		}
	}
	# http://bbs.archlinux.org/viewtopic.php?id=95582
	slave.pcm "surround20"
}

# speaker-test -D headphones -c 2 -t wav
# audacious uses less CPU when running bs2b as a listed plugin, probably because of samplerate_best
# Posted at http://bbs.archlinux.org/viewtopic.php?pid=626573#p626573
pcm.headphones {
	type rate
	slave {
		pcm "plug:bs2b"
		#pcm "plug:narrower"
		#rate 44100
		rate 48000
	}
	#converter "speexrate_medium"

    hint {
        show on
        description "Headphones"
    }
}
# http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/PerfectSetup/
pcm.pulse {
	type pulse
}

ctl.pulse {
	type pulse
}