Announcing phasex-0.12.0-pre1:
The pre-release is here. It's stable. It builds cleanly. No build
errors or warnings. No runtime errors or crashes. No glitches.
If no bugs are found in the next week, then this code (with the
addition of some brand-new patches) will become the stable release.
Changes since phasex-0.12.0-beta4:
- Build system overhaul. GCC version detection for better arch-
specific optimization (compiles clean with gcc-3.4.x or higher).
Atom builds fall back to next-best optimizations for GCC < 4.5.x.
New --enable-32bit and --cpu-power-level= flags and better user
variable handling for ./configure. Missing -lpthread in link phase
is fixed.
- Velocity control for the amplifier. Better aftertouch handling.
- Scheduling policy defaults to SCHED_RR, and can be changed in the
settings at runtime.
- Broken ringbuffer code has been fixed. No more glitches (unless you
have actually run out of CPU).
- Theme loading has been fixed. Now properly detects if the nodoka
theme engine is installed, and quietly falls back to an engineless
theme.
- Fullscreen and Maximize interaction have been fixed.
- New command line options: -m (--midi-channel=), -f (--fullscreen),
and -M (--maximize). Loading a patch by name or by program number
on the command line now finds the appropriate patch in the patchbank.
- The lurking patch name corruption bug has been fixed!
- And more. See the ChangeLog for details...
Download phasex and hear for yourself...
http://sysex.net/phasex/
Alternatively, you can utilize the git repositoty:
git clone http://sysex.net/git/phasex.git
Cheers,
--ww
I wrote a scripting engine for a pro audio plugin by embedding a
CPython interpreter. Since audio sequencing hosts use separate threads
for each audio track, loading more than one instance of my plugin
while running scripting code causes contention on Python's GIl and
results in CPU spikes at low latencies.
While CPython is more than capable of running fast enough in the audio
thread for control-rate (MIDI) work, the GIL is killing us. Using
process migration to move calls to CPython to daemon processes would
take less code than forking python itself, but the scripting engine
includes a python extension module that exposes pointers to the C++
classes in the audio engine. This complicates things quite a bit. The
calls look like this:
size = engine.getAudioBufferSize()
engine.loadPatchFile(SOME_PATH)
instrument.loadAudioFile(SOME_PATH_2)
instrument.setVolume(.5)
...where 'engine' and 'instrument' are C++ classes in the audio engine
that I wrapped in a python extension.
I think that the biggest problem for me is tackling the complexity of
managing new real time daemon processes for each audio track, finding
an IPC method, and also implementing a middle-ware layer that would
allow those synchronous calls to the engine be made back to the host.
This is the sort of overhead that one can expect when trying to use
process migration to work around the GIL. I consider myself an
experienced coder, but just thinking about finding a clean, rock-solid
daemon management and IPC mechanism that works on Mac and Windows
freaks me out.
Does anyone have any comments on this topic?
cheers,
-P
Hi all,
the LinuxSampler team is proud to announce LinuxSampler 1.0.0, with
many new features and modules, device drivers and plugin architectures
supported. Available for Linux, Windows and OS X.
See Christian's announcement below:
get it from the usual place:
http://www.linuxsampler.org
support forum:
http://bb.linuxsampler.org
if you wish to support the project financially with your donations and
monthly subscriptions a big thanks in advance on behalf of the
LinuxSampler development team.
see here how to donate:
http://www.linuxsampler.org/donations.html
Feel free to forward this mail to linux-audio-user (I'm not
subscribed), other linux audio related sites or general audio
websites.
Open Octave and LinuxSampler technical overview videos:
http://bb.linuxsampler.org/viewtopic.php?f=6&t=396&sid=b3f8eac9fe3591272785…
thanks,
Benno
Original announcement message:
--------
Hi!
Just to make it official, a new release wave has appeared a (long) while ago:
* linuxsampler 1.0.0
* Fantasia 0.9
* qsampler 0.2.2
* gigedit 0.2.0
* libgig 3.3.0
* jlscp 0.8
* liblscp 0.5.6
Summary of changes:
This is the first release which allows the sampler to be used
as audio host
plugin, namely supporting the standards VST, AU, DSSI and LV2.
The sampler's
limits for max. voices & disk streams can now be altered at runtime by
frontends, no need to recompile the sampler anymore. The Mac
version now also
supports CoreAudio as audio driver. The Windows version finally
supports the
sampler's instruments DB feature as well, however expect it still to be
unstable at this point. Along to the already existing JACK
audio driver, Jack
MIDI support has been added in this release. The sampler allows
frontends now
basic MIDI control, that is to monitor incoming MIDI data on MIDI input
devices and sampler channels and to send note-on and note-off
MIDI events to
sampler channels, which allows frontends to provide a virtual
MIDI keyboard
to the user. Besides these major changes there were countless
bugfixes and
optimizations.
Many of you probably saw that most of the files already appeared more than 2
months ago on the webserver. Reason for this late announcement is that some
binaries took quite some additional time. So we wanted to wait until all
binaries of this new release were ready as well, which is now finally the
case.
Of course you get everything from the usual place:
http://www.linuxsampler.org
CU
Christian
Hi,
I'm impressed by my recent discovery: non-daw and especially non-sequencer:
http://non.tuxfamily.org/
I love the design of the sequencer, it's very intuitive, userfriendly
and fast! Respect for that!
But I couldn't contact the author. So I was wondering, maybe he is on
this (lad) list?
I really like to know what his plans are with the apps and if he needs
support in some kind of way. Does he continue to work on it? (If not
that would be a real pity, cause this is not just another sequencer...
And then we might need some talented devs to fork it...)
Ok, I hope I will get some sign of life from the author soon...
Kind regards,
\rooz
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Hash: SHA1
Hi,
I'm just curious what your long-time experiences with these
gui-libraries are.
Considering to use one of these two but can't really decide.
But I do not want to switch in a year or two...
Thanks for your advices!
Christian
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iEYEARECAAYFAkp/wE0ACgkQVC26eJ+o0+2UQACePLcVXkXSwPygZrC1sQnDUWC0
hk0AmgNmgru7KzOfYCkGppoqldsam5GH
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Hi guys,
I'm looking to delve into the world of plugins (for both synth and fx),
and of the free API's I find LADSPA, DSSI, and LV2. But the picture is a
little confusing.
It appears that LV2 is "current," that DSSI is deprecated, and that LADSPA
would be deprecated if it weren't so widely adopted. However, LV2 is slow
in being adopted. Is this developer resistance... or is it just too new?
Also, I can't find any applications that will host all 3, nor even
LV2+DSSI. Is there a technical reason for this?
Thanks in advance,
Gabriel
Problem solved by a contributor on LAU.
He sent me a /var/lib/asound.state file from a working Audigy2
installation and, what do you know, it worked.
There is a file that describes the state of the sound card in
/var/lib/alsa/asound.state. When the original (bad) version was replaced
with one from a working Audigy2 sound cared (good version) the card
started working.
The replacement asound.state file has 227 controls vs 216 for the oroginal.
The following 14 switch controls existed only in the good file.
The Master Playback Switch only existed in the good file. If it exists
on the sound card and is muted by default, there would be no sound output.
There are 3 possible duplicated controls.
control.1 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Front Playback Switch'
value.0 true
value.1 true
control.2 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Surround Playback Switch'
value.0 true
value.1 true
}
control.3 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Center Playback Switch'
value true
}
control.4 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'LFE Playback Switch'
value true
}
control.5 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Side Playback Switch'
value.0 true
value.1 true
control.7 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'CD Playback Switch'
value.0 true
value.1 true
control.9 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Line Playback Switch'
value.0 true
value.1 true
control.11 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Mic Playback Switch'
value.0 false
value.1 false
control.12 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Front Mic Playback Switch'
value.0 true
value.1 true
control.13 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Headphone Playback Switch'
value.0 true
value.1 true
control.15 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Capture Switch'
value.0 true
value.1 true
control.20 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'IEC958 Playback Switch'
value false
control.21 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'IEC958 Default PCM Playback Switch'
value true
control.22 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Master Playback Switch'
value true
-----------------
The following controls existed in both files.
Bad file, Good file
Control 29 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'Tone Control - Switch'
value.0 false
value.1 falseiface MIXER
Bad file, Good file
control.31 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 2
iface MIXER
name 'IEC958 Optical Raw Playback Switch'
value.0 false
value.1 false
Bad file, Good file NB also see control 13, an apparent duplicate
control.35 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Headphone Playback Switch'
value true
Bad file, Good file
control.39 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'PC Speaker Playback Switch'
value true
Bad file, Good file
control.41 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Phone Playback Switch'
value true
Bad file, Good file NB also see control 9, a possible duplicate
control.46 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Line Playback Switch'
value true
Bad file, Good file NB also see control 7, a possible duplicate.
control.48 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'CD Playback Switch'
value true
Bad file, Good file
control.52 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name 'Aux Playback Switch'
value true
Bad file, Good file
control.60 {
comment.access 'read write'
comment.type BOOLEAN
comment.count 1
iface MIXER
name '3D Control - Switch'
value false
--
David Morrell
Web site: www.davidmorrell.ozeweb.net (when I get it on air again :)
<http://www.davidmorrell.ozeweb.net>
Email: dsmorrell56(a)dodo.com.au <mailto:dsmorrell56@dodo.com.au>
Ph: 0408 842 955 / 03 6343 5131
Hi all, my name is David, I'm new to the list. It was suggested that I try here by someone on the LAU list as they think I may have found a bug in the emu10k1 driver for Creative Audigy sound cards.
Sound works in Windows XP and worked in Ubuntu 8.04. It stopped after a
clean installation of Ubuntu Studio 9.04. It is not known if recording
worked or if it is now broken. Hopefully sorting out playback will
lead to a quick solution for recording, if one is needed.
Summary of Test Results
--------------------------------
Testing followed this understanding of the way sound playback is handled
in Linux. Sound;
- is generated by apps or test utilities such as Hydrogen or aplay
- may pass through a sound server such as PulseAudio or Jack
- goes to ALSA, which translates it for communication to sound cards
- is converted from digital to analogue, amplified and fed to speakers
by the sound card.
The system has a Creative Labs Audigy2 sound card.
It shares an IRQ with a video card and an onboard USB controller.
Video and USB work properly in Linux. Video, USB and sound work in
Windows.
However, while BIOS reports these items on IRQ10, Linux (using cat
/proc/interrupts) thinks they are on IRQ18. Is there any significance in
this? It seems unlikely as video and USB work properly.
ALSA appears to talk to the card – it is possible to set mixer controls
using one mixer interface and read their state using an unrelated
interface. Apps and test utilities appear able to send data to the
card. To back this up, the kernel modules required to form the driver
all appear to be loaded. When ALSA is forced to reload them, a pop
and brief low level white noise comes from the speaker.
This is similar to what can be heard during system boot.
Possibly ALSA is passing control data to the card but not PCM audio
data. I don't know if this is so or how to check.
Re-installing ALSA had no effect. A full uninstall – reinstall was not
done as many other packages would have been affected because of
dependencies.
Detailed information about the driver was reviewed, but it is beyond
my technical competence to understand if or how it applies to my card
and then to act on it (“Matrix:Module-emu10k1” at
http://www.alsa-project.org/main/index.php/Matrix:Module-emu10k1).
Now for the test results in detail.
Thanks to Dave Phillips for his excellent series "Troubleshooting Linux
Audio" at
http://www.linuxjournal.com and to many other sources.
Application & Test Utility Output
--------------------------------
the Hyrdogen drum machine can be seen producing output on its VU meters.
From the command line: aplay -vv ak4744.wav shows an apparently
successful result;
Playing WAVE 'ak4744.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo Plug PCM: Hardware PCM card 0 'Audigy 2 Value [SB0400]' device 0
subdevice 0
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 16384
period_size : 4096
period_time : 92879
tstamp_mode : NONE
period_step : 1
avail_min : 4096
period_event : 0
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
Test tone from Sound → Preferences → Test output appears to be
successful to most sound pipelines;
- Audigy 2 Multichannel Playback (ALSA) OK
- Audigy 2 Multichannel Capture / PT Playback (ALSA) Fails with
error message “gconfaudiosink: cound not open audio device for playback”
(also describes the audio sources as a 512 Hz tone and the pipeline as
“audioconvert ! Audioresample ! Gconfaudiosink”).
- Audigy 2 ADC Capture / Standard PCM Playback (ALSA) OK
- Audigy 2 ADC Capture / Standard PCM Playback (OSS) 2 are OK, the
third apparently identical one fails
- ALSA OK
- OSS OK
Sound Server
--------------------------------
PulseAudio and JACK were both on the sytesm. To simplify the problem;
JACK was not started;
PulseAudio was completely uninstalled, the system was restarted and it
was
verified that no persistent processes still existed for PulseAudio
(ps -e | grep “pulse”)
Drivers (ALSA)
--------------------------------
All volume controls were set to 100% and all mute controls were set to
un-mute in Alsamixer. These settings were verified by having another
mixer read and display them (Alsamixergui).
As can be seen from the preceding output of aplay, sound was routed
directly from its source to ALSA and ALSA recognised the Audigy 2 card
on this computer.
Output from the Sound → Preferences test also suggests that ALSA is
interrogating the card and recognising what it can and cannot do.
However, there is a strange indication from ALSCTL;
alsactl init
which shows;
Unknown hardware: "Audigy2" "SigmaTel STAC9750,51"
"AC97a:83847650" "" ""
What is this saying?
The ALSA soundcard matrix says an Audigy2 card should be using emu10k1
modules..
http://www.alsa-project.org/main/index.php/Matrix:Vendor-Creative_Labs
though I can't find an exact match there for my Audigy 2 SB0400 card.
I tried forcing ALSA to reload its kernel modules using;
sudo alsa reload
The speaker emitted a soft pop and about half a second of very low level
white
noise, similar to what can be heard during system boot.
Commmand output was;
lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system
/home/dmorrell/.gvfs
Output information may be incomplete.
/sbin/alsa: Warning: Processes using sound devices: 3215(timidity).
Unloading ALSA sound driver modules: snd-emu10k1-synth snd-emux-synth
snd-seq-virmidi snd-seq-midi-emul snd-emu10k1 snd-ac97-codec snd-pcm-oss
snd-mixer-oss snd-pcm snd-page-alloc snd-util-mem snd-hwdep snd-seq-dummy
snd-seq-oss snd-seq-midi snd-rawmidi snd-seq-midi-event snd-seq snd-timer
snd-seq-device (failed: modules still loaded: snd-seq snd-timer
snd-seq-device).
Loading ALSA sound driver modules: snd-emu10k1-synth snd-emux-synth
snd-seq-virmidi snd-seq-midi-emul snd-emu10k1 snd-ac97-codec snd-pcm-oss
snd-mixer-oss snd-pcm snd-page-alloc snd-util-mem snd-hwdep
snd-seq-dummy
snd-seq-oss snd-seq-midi snd-rawmidi snd-seq-midi-event snd-seq
snd-timer
snd-seq-device.
Kernel module(s) the card is using were checked using
cat /proc/asound/modules
which shows;
0 snd_emu10k1
i.e. there is only card 0 and it is using module snd_emu10k1.
This was further confirmed using command lspci -v | grep “Audigy”, which
showed;
05:09.0 Multimedia audio controller: Creative Labs SB0400
Audigy2 Value Kernel driver in use: EMU10K1_Audigy
A test was done to confirm that the modules are actually loaded into
the kernel;
lsmod | grep "emu"
which showed a host of modules that seem as if they should be there,
though of course could not show any modules that aren't there and
should be;
snd_emu10k1_synth 14336 0
snd_emux_synth 40832 1 snd_emu10k1_synth
snd_seq_virmidi 13440 1 snd_emux_synth
snd_seq_midi_emul 14592 1 snd_emux_synth
snd_emu10k1 144288 1 snd_emu10k1_synth
snd_ac97_codec 112292 1 snd_emu10k1
snd_pcm 83076 3 snd_emu10k1,snd_ac97_codec,snd_pcm_oss
snd_page_alloc 16904 2 snd_emu10k1,snd_pcm
snd_util_mem 12288 2 snd_emux_synth,snd_emu10k1
snd_hwdep 15108 2 snd_emux_synth,snd_emu10k1
snd_rawmidi 29696 3 snd_seq_virmidi,snd_emu10k1,
snd_seq_midi
snd_seq 56880 10
snd_emux_synth,snd_seq_virmidi,snd_seq_midi_emul,snd_seq_dummy,
snd_seq_oss,snd_seq_midi,snd_seq_midi_event
snd_timer 29704 3 snd_emu10k1,snd_pcm,snd_seq
snd_seq_device 14988 8
snd_emu10k1_synth,snd_emux_synth,snd_emu10k1,snd_seq_dummy,snd_seq_oss,
snd_seq_midi,snd_rawmidi,snd_seq
snd 62756 14
snd_emux_synth,snd_seq_virmidi,snd_emu10k1,snd_ac97_codec,snd_pcm_oss,
snd_mixer_oss,snd_pcm,snd_hwdep,snd_seq_oss,snd_rawmidi,snd_seq,snd_
timer,snd_seq_device
Just for the hell of it, I had a look at information about the emu10k1
module using;
modinfo snd-emu10k1
which showed;
filename:
/lib/modules/2.6.28-15-generic/kernel/sound/pci/emu10k1/snd-emu10k1.ko
license: GPL
description: EMU10K1
author: Jaroslav Kysela <perex(a)perex.cz>
firmware: emu/emu1010_notebook.fw
firmware: emu/emu0404.fw
firmware: emu/micro_dock.fw
firmware: emu/emu1010b.fw
firmware: emu/audio_dock.fw
firmware: emu/hana.fw
srcversion: F52CF37385CBD708CAB4A2C
alias: pci:v00001102d00000008sv*sd*bc*sc*i*
alias: pci:v00001102d00000004sv*sd*bc*sc*i*
alias: pci:v00001102d00000002sv*sd*bc*sc*i*
depends:
snd-pcm,snd-util-mem,snd-page-alloc,snd,snd-rawmidi,snd-timer,
snd-hwdep,snd-ac97-codec,snd-seq-device
vermagic: 2.6.28-15-generic SMP mod_unload modversions 586
parm: index:Index value for the EMU10K1 soundcard.
(array of int)
parm: id:ID string for the EMU10K1 soundcard. (array of charp)
parm: enable:Enable the EMU10K1 soundcard. (array of bool)
parm: extin:Available external inputs for FX8010.
Zero=default. (array
of int)
parm: extout:Available external outputs for FX8010.
Zero=default.
(array of int)
parm: seq_ports:Allocated sequencer ports for internal
synthesizer.
(array of int)
parm: max_synth_voices:Maximum number of voices for
WaveTable. (array
of int)
parm: max_buffer_size:Maximum sample buffer size in MB.
(array of int)
parm: enable_ir:Enable IR. (array of bool)
parm: subsystem:Force card subsystem model. (array of uint)
Noticing the parameter 'enable' (ton enable a sound card) and feeling
desparate, I tried reloading the module with that parameter specified;
sudo modprobe -i snd_emu10k1 enable=true
Unsurprisingly, it made no difference as the card already appears to
be talking to its driver anyway.
The ALSA soundcard matrix mentions that some cards may need the
alsa-firmware package. That package is not on my system or in the
Ubuntu repositories. However, its headers are installed. I don't
know if this is significant.
Next, I tried reinstalling ALSA. The following packages were affected;
libsox-fmt-alsa
alsa-base
libasound2
libasound2-plugins
mcp-plugins
libesd-alsa0
This had no effect, so I considered completely uninstalling &
reinstalling. However, many other packages that depend on ALSA would
also have been removed. This looked like a short cut to a long
reconstruction.
Sound Card
--------------------------------
The card works under Windows XP and had worked under Ubuntu 8.04.
I have a Creative Labs Audigy 2 PCI card as card 0 on IRQ18 according
to cat /proc/asound/cards
0 [Audigy2 ]: Audigy2 - Audigy 2 Value [SB0400]
Audigy 2 Value [SB0400] (rev.0, serial:0x10011102) at 0x1040, irq 18
More information is given by
cat /proc/interrupts
which shows
18: 185146 IO-APIC-fasteoi uhci_hcd:usb4, EMU10K1,
radeon@pci:0000:01:00.0
There is a USB controller, the soundcard (EMU10K1) and the video card
(radeon) all using the same interrupt. The sound card does not appear
on any other interrupts.
BIOS reports all of these devices on IRQ10, not IRQ18. And yes, I have
checked carefully to be sure I am not confusing 8 with 0.
How could this discrepancy happen?
A hint may lie in the online book “The Linux Kernel” at
http://tldp.org/LDP/tlk/dd/pci.html.
“For Intel based systems the system BIOS, which ran at boot time, has
already fully configured the PCI system. This leaves Linux with little
to do other than map that configuration. For non-Intel based systems
further configuration needs to happen to:
- Allocate PCI I/O and PCI Memory space to each device,
- Configure the PCI I/O and PCI Memory address windows for each PCI-PCI
bridge in the system,
- Generate Interrupt Line values for the devices; these control
interrupt handling for the device. “
Possibly Linux has read the information set up by the BIOS, then the
PCI Fixup routine has remapped the Interrupt Line values even though.it
may not have needed to do so?
For a boots and all look at the PCI information about the sound card,
lspci -vvxxx
shows
05:09.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value
Subsystem: Creative Labs Device 1001
Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop-
ParErr- Stepping-
SERR+ FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort
- <TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 64 (500ns min, 5000ns max)
Interrupt: pin A routed to IRQ 18
Region 0: I/O ports at 1040 [size=64]
Capabilities: [dc] Power Management version 2
Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,
D2-,D3hot-,D3cold-)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
Kernel driver in use: EMU10K1_Audigy
Kernel modules: snd-emu10k1
The card has the following devices shown by;
aplay -L
This shows abridged);
front:CARD=Audigy2,DEV=0
Audigy 2 Value [SB0400], ADC Capture/Standard PCM Playback
Front speakers
rear:CARD=Audigy2,DEV=0
Audigy 2 Value [SB0400], ADC Capture/Standard PCM Playback
Rear speakers
…
Audigy 2 Value [SB0400], ADC Capture/Standard PCM Playback
IEC958 (S/PDIF) Digital Audio Output
null
Discard all samples (playback) or generate zero samples (capture)
To see all input devices;
arecord -l
which shows;
arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 0: emu10k1 [ADC
Capture/Standard PCM Playback]
Subdevices: 1/1
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 1: emu10k1
mic [Mic Capture]
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 2: emu10k1
efx [Multichannel
Capture/PT Playback]
Subdevices: 1/1
-- David Morrell Web site: www.davidmorrell.ozeweb.net (when I get it on
air again :) <http://www.davidmorrell.ozeweb.net> Email:
dsmorrell56(a)dodo.com.au <mailto:dsmorrell56@dodo.com.au> Ph: 0408 842
955 / 03 6343 5131 _______________________________________________
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