[Alsa-user] Help configuring HDSP9632
Hi guys, I've been trying off and on for weeks now, but I can't get my RME HDSP9632 configured under ALSA properly. (Is it me or does ALSA really not make this an easy process? I can't find anywhere to get any feedback from the system on configuration errors. Hell, if you make a typo in the config file, you only find out because of some scary-looking output from aplay. The docs on the website seem to be quite old and often conflict with each other and I can't find any relevant man pages.) Anyhow, the card works out of the box, insofar as it lets one application play sound through it at a time. So I'm trying to do the common thing of configuring dmix to let multiple applications output sound at once. Nothing fancy, really! At least, at this point. My /etc/asound.conf file is as follows (pieced together from tuts and the like): pcm.!default { type plug slave.pcm hdsp9632_dmix hint { show on description Default device: Plugs into hdsp9632_dmix. } } ctl.hdsp9632_dmix { type hw card 0 } pcm.hdsp9632_dmix { type dmix ipc_perm 0660 ipc_key 1025 ipc_key_add_uid false slave { pcm hw:0,0 rate 44100 channels 2 period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } hint { show on description hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. } } Using this file, I get # aplay -L null Discard all samples (playback) or generate zero samples (capture) default Default device: Plugs into hdsp9632_dmix. hdsp9632_dmix hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. which is what I hope for. But, if I try and play something with vlc, I get an error message and No AutoSync source for requested rate comes up in dmesg. The card is currently set to clock master at the same sample rate as the audio (44.1kHz). If anyone can shed any light on what I'm doing wrong (and, ideally, some methodology on configuring ALSA which doesn't require scrabbling around in the dark!), I will be grateful until the end of time. Thanks a lot, Matt -- Free Software Download: Index, Search Analyze Logs and other IT data in Real-Time with Splunk. Collect, index and harness all the fast moving IT data generated by your applications, servers and devices whether physical, virtual or in the cloud. Deliver compliance at lower cost and gain new business insights. http://p.sf.net/sfu/splunk-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Thinkpad X41 Tablet audio (AD1981B) unusably quiet after upgrade to 2.6.37 kernel
Hi all, (I sent this last week but I guess it died in the moderation queue?) Audio worked well on my Thinkpad X41 Tablet (using Arch Linux) until I did a system upgrade a few days ago that included a new kernel. At first I thought I had no sound at all, but after some troubleshooting I realized there is sound, it's just really really quiet. If I play an MP3 with all mixer controls unmuted and volumes at max I can barely make out music with quality headphones. Built in speaker output works but is entirely drowned out by system hums and clicks. The mute and volume mixers appear to work. There don't seem to be any errors or changes in the output of the various alsa utils, though I'm not very familiar with the linux/ALSA system. The problem is independent of audio program or file source. I reran alsaconf and reset my mixers and tried again, no change. Output of alsa-info.sh here: http://pastebin.com/ep8R2mLB Is this a regression? Is there anything I can do to help get it fixed? Thanks and cheers. -- John Galt tagalog8...@fastmail.fm -- http://www.fastmail.fm - Or how I learned to stop worrying and love email again -- Free Software Download: Index, Search Analyze Logs and other IT data in Real-Time with Splunk. Collect, index and harness all the fast moving IT data generated by your applications, servers and devices whether physical, virtual or in the cloud. Deliver compliance at lower cost and gain new business insights. http://p.sf.net/sfu/splunk-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help configuring HDSP9632
Hi Matt, I didn't use a RME HDSP9632 for quite a long time (also I never used it with the dmix plugin). However, the dmesg message sounds like a clock source problem. All I can suggest is to check for the correct rate settings, e.g. compare what hdspconf is showing to the output of cat /proc/asound/card0/hdsp --fe Matthew Robbetts schrieb am 26.02.2011 15:18: Hi guys, I've been trying off and on for weeks now, but I can't get my RME HDSP9632 configured under ALSA properly. (Is it me or does ALSA really not make this an easy process? I can't find anywhere to get any feedback from the system on configuration errors. Hell, if you make a typo in the config file, you only find out because of some scary-looking output from aplay. The docs on the website seem to be quite old and often conflict with each other and I can't find any relevant man pages.) Anyhow, the card works out of the box, insofar as it lets one application play sound through it at a time. So I'm trying to do the common thing of configuring dmix to let multiple applications output sound at once. Nothing fancy, really! At least, at this point. My /etc/asound.conf file is as follows (pieced together from tuts and the like): pcm.!default { type plug slave.pcm hdsp9632_dmix hint { show on description Default device: Plugs into hdsp9632_dmix. } } ctl.hdsp9632_dmix { type hw card 0 } pcm.hdsp9632_dmix { type dmix ipc_perm 0660 ipc_key 1025 ipc_key_add_uid false slave { pcm hw:0,0 rate 44100 channels 2 period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } hint { show on description hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. } } Using this file, I get # aplay -L null Discard all samples (playback) or generate zero samples (capture) default Default device: Plugs into hdsp9632_dmix. hdsp9632_dmix hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. which is what I hope for. But, if I try and play something with vlc, I get an error message and No AutoSync source for requested rate comes up in dmesg. The card is currently set to clock master at the same sample rate as the audio (44.1kHz). If anyone can shed any light on what I'm doing wrong (and, ideally, some methodology on configuring ALSA which doesn't require scrabbling around in the dark!), I will be grateful until the end of time. Thanks a lot, Matt -- Free Software Download: Index, Search Analyze Logs and other IT data in Real-Time with Splunk. Collect, index and harness all the fast moving IT data generated by your applications, servers and devices whether physical, virtual or in the cloud. Deliver compliance at lower cost and gain new business insights. http://p.sf.net/sfu/splunk-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help configuring HDSP9632
On Sat, 26 Feb 2011, Friedrich Ewaldt wrote: Hi Matt, I didn't use a RME HDSP9632 for quite a long time (also I never used it with the dmix plugin). However, the dmesg message sounds like a clock source problem. All I can suggest is to check for the correct rate settings, e.g. compare what hdspconf is showing to the output of cat /proc/asound/card0/hdsp --fe Matthew Robbetts schrieb am 26.02.2011 15:18: Hi guys, I've been trying off and on for weeks now, but I can't get my RME HDSP9632 configured under ALSA properly. (Is it me or does ALSA really not make this an easy process? I can't find anywhere to get any feedback from the system on configuration errors. Hell, if you make a typo in the config file, you only find out because of some scary-looking output from aplay. The docs on the website seem to be quite old and often conflict with each other and I can't find any relevant man pages.) Anyhow, the card works out of the box, insofar as it lets one application play sound through it at a time. So I'm trying to do the OK. It works. That finishes alsa. common thing of configuring dmix to let multiple applications output sound at once. Nothing fancy, really! At least, at this point. That in general has nothing to do with the card or the driver of the card. Most cards do not allow multiple inputs to all play at once. It is software. It is often pulseaudio or jack could be used as well. . My /etc/asound.conf file is as follows (pieced together from tuts and the like): pcm.!default { type plug slave.pcm hdsp9632_dmix hint { show on description Default device: Plugs into hdsp9632_dmix. } } ctl.hdsp9632_dmix { type hw card 0 } pcm.hdsp9632_dmix { type dmix ipc_perm 0660 ipc_key 1025 ipc_key_add_uid false slave { pcm hw:0,0 rate 44100 channels 2 period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } hint { show on description hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. } } Using this file, I get # aplay -L null Discard all samples (playback) or generate zero samples (capture) default Default device: Plugs into hdsp9632_dmix. hdsp9632_dmix hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. which is what I hope for. But, if I try and play something with vlc, I get an error message and No AutoSync source for requested rate comes up in dmesg. The card is currently set to clock master at the same sample rate as the audio (44.1kHz). If anyone can shed any light on what I'm doing wrong (and, ideally, some methodology on configuring ALSA which doesn't require scrabbling around in the dark!), I will be grateful until the end of time. The alsa users documentation has long long long been its greatest shortfall. And noone seems to be stepping up to the plate to write the docs. One of the problems with the open software movement. In a proprietary system, the company would assign someone to write the docs, whether they wanted to or not. In the free software world, people do things they want to do. Noone wants to write alsa docs--Noone does write alsa docs. (Despite that I like the free software movement, but it does have its problems.) Thanks a lot, Matt -- Free Software Download: Index, Search Analyze Logs and other IT data in Real-Time with Splunk. Collect, index and harness all the fast moving IT data generated by your applications, servers and devices whether physical, virtual or in the cloud. Deliver compliance at lower cost and gain new business insights. http://p.sf.net/sfu/splunk-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273 PhysicsAstronomy | Advanced Research | Fax: +1(604)822-5324 UBC, Vancouver,BC | Program in Cosmology | un...@physics.ubc.ca Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/ -- Free Software Download: Index, Search Analyze Logs and other IT data in Real-Time with Splunk. Collect, index and harness all the fast moving IT data generated by your applications, servers and devices whether physical, virtual or in the cloud. Deliver compliance at lower cost and gain new business insights. http://p.sf.net/sfu/splunk-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help configuring HDSP9632
In theory you don't need the dmix thing anymore. If your applications use ALSA natively, it will automagically mix sound from several applications (in software). If the applications use OSS, you can force it to use alsa with aoss. BITD you'd run esddsp or artsdsp -m app to do this sort of thing. Depending on the sound daemon of choice you happened to be using. These days those daemons just get in the way, chew up resources and cause XRUNs or other woes. If your applications are configured to use ALSA, this should be a non-issue. Assuming that you're running something current and not RH 5.1 from some book or something. In the few times that I tried to use dmix BITD, it was generally the cause of problems, not the solution. If your applications use OSS and you don't launch them with aoss, then they will lock the device (per days of old). I'm not sure if that's addressed with oss emulation or not. And aoss isn't perfect as something like a browser will launch pop ups that are NOT launched with aoss and break the very thing you were trying to avoid. Mostly problematic with internet gaming where the games are pop ups. But for most other application you can select the audio system of choice. alsa, oss, jackd, artsd, esd, pulse-audio, and probably others. Alsa, having the least overhead IMO, if you're coming up short on system resources. check your .asoundrc and whatever system defaults were created for you or by you in /etc/. I'm not sure of that locations default naming convention as it probably varies between distros. alsa.conf? asound.conf? +/- an /etc/ or /etc/alsa/ or /etc/sound/ or ??? And various tricks of old to delete the asound.state file to force new defaults. Located at /var/lib/alsa/asound.state on my system. YMMV HTH, - James On 2/26/11, Bill Unruh un...@physics.ubc.ca wrote: On Sat, 26 Feb 2011, Friedrich Ewaldt wrote: Hi Matt, I didn't use a RME HDSP9632 for quite a long time (also I never used it with the dmix plugin). However, the dmesg message sounds like a clock source problem. All I can suggest is to check for the correct rate settings, e.g. compare what hdspconf is showing to the output of cat /proc/asound/card0/hdsp --fe Matthew Robbetts schrieb am 26.02.2011 15:18: Hi guys, I've been trying off and on for weeks now, but I can't get my RME HDSP9632 configured under ALSA properly. (Is it me or does ALSA really not make this an easy process? I can't find anywhere to get any feedback from the system on configuration errors. Hell, if you make a typo in the config file, you only find out because of some scary-looking output from aplay. The docs on the website seem to be quite old and often conflict with each other and I can't find any relevant man pages.) Anyhow, the card works out of the box, insofar as it lets one application play sound through it at a time. So I'm trying to do the OK. It works. That finishes alsa. common thing of configuring dmix to let multiple applications output sound at once. Nothing fancy, really! At least, at this point. That in general has nothing to do with the card or the driver of the card. Most cards do not allow multiple inputs to all play at once. It is software. It is often pulseaudio or jack could be used as well. . My /etc/asound.conf file is as follows (pieced together from tuts and the like): pcm.!default { type plug slave.pcm hdsp9632_dmix hint { show on description Default device: Plugs into hdsp9632_dmix. } } ctl.hdsp9632_dmix { type hw card 0 } pcm.hdsp9632_dmix { type dmix ipc_perm 0660 ipc_key 1025 ipc_key_add_uid false slave { pcm hw:0,0 rate 44100 channels 2 period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } hint { show on description hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. } } Using this file, I get # aplay -L null Discard all samples (playback) or generate zero samples (capture) default Default device: Plugs into hdsp9632_dmix. hdsp9632_dmix hdsp9632_dmix: The dmix plugin - plugs into hdsp9632. which is what I hope for. But, if I try and play something with vlc, I get an error message and No AutoSync source for requested rate comes up in dmesg. The card is currently set to clock master at the same sample rate as the audio (44.1kHz). If anyone can shed any light on what I'm doing wrong (and, ideally, some methodology on configuring ALSA which doesn't require scrabbling around in the dark!), I will be grateful until the end of time. The alsa users documentation has long long long been its greatest shortfall. And noone seems to be stepping up to the plate to write the docs. One of the problems with the open software movement. In a
[Alsa-user] K42Jc Notebook(ALC269) port switch between headset, headphone, mic
Hi, I want to switch between headphone, headset, mic input/output modes when I plug in devices. Alsamixer shows two capture channels, one is for the internal mic. I don't know how to use the other. I've tried switching some things on and off in hda-analyzer but I failed to get the functionality that I wanted. This notebook only has one audio port(TRS 3.5mm), so the selection is vital if I'm going to use an external mic. Currently the port is acting as an output only port. And something fishy goes on when I select Analog Stereo Input in profiles in Sound Preferences. It lets me use the external mic through the audio port but only if I push the jack half way, the input is detected and I can record it(but there's no sound output after I select this profile, its input only). I some how think that this is the mic mode and if I can get it to work on headset mode then I'd be able to use it for input and output. This is the information that I got from the script, http://www.alsa-project.org/db/?f=e70dae201e5d830874536be93090aef0454c8c1e Thank you, - Madura A. -- Free Software Download: Index, Search Analyze Logs and other IT data in Real-Time with Splunk. Collect, index and harness all the fast moving IT data generated by your applications, servers and devices whether physical, virtual or in the cloud. Deliver compliance at lower cost and gain new business insights. http://p.sf.net/sfu/splunk-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user