Linux-audio-user May 2004

linux-audio-user@lists.linuxaudio.org

154 participants
165 discussions

[linux-audio-user] ALSA and M-Audio Dio2448 SPDIF input (again)

by Anahata

Thanks to some recent help here, I've now found several applications that appear to control my DIO2448 with ALSA. I still can't get it to record fromt he SPDIF input though. ALSA has installed the generic CMI PCI 8738 driver, which I believe is the correct thing to do. Apps like alsamixergui have the right switches, but something seems to be disabled. The IEC958 In select doen't respond, and I can't find any docs which explain, e.g. - what combination of controls is required for this - what "IEC958 IN Valid" means (it's a control input, not a status display...) - what values I'm likely to need to change with iecset I can enable/disable monitoring of IEC958 input though analog output, and the IEC958 phase invert has the expected effect, so the driver is installed correctly. Should I be able to record with Audacity 1.2.0, with the alsa-oss package installed? Audacity plays OK. -- Anahata anahata(a)treewind.co.uk -+- http://www.treewind.co.uk Home: 01638 720444 Mob: 07976 263827

19 years, 11 months

Re: [linux-audio-user] Steinberg The Grand

by Kjetil Svalastog Matheussen

Carl Hetherington: > > Hi, > > Has anyone had any luck running Steinberg's The Grand under vstserver? > vstserver works beautifully for the Native Instruments B4 and the > Steinberg EP, but The Grand gets close and then wine starts complaining. > > I'll post more details if people are interested, but I'd just like to > know if anyone has managed to get it going. Luke Yelavich: > >Hi, > > > >Has anyone had any luck running Steinberg's The Grand under vstserver? > >vstserver works beautifully for the Native Instruments B4 and the > >Steinberg EP, but The Grand gets close and then wine starts > >complaining. > > OT, but have you tried B4 with the FST library? Compatibility regarding running vst-plugins is allways related to wine, not whether you use vstserver or the FST library (at least if we assume that both vstserver and the FST library are bugfree :). The big question is: Which version of wine are you using? If you are still using the 9.12.2003 version of wine, which I reccomend for use with (or at least when compiling) the vstserver, you can try to upgrade to the latest version. (But you _must not_ recompile the vstserver afterwards!) There are some chances plugins that didn't work earlier will work after a wine upgrade. --

19 years, 11 months

[linux-audio-user] parameters needed for managing multiple soundcards

by michele.spinolo＠tin.it

Hi guys, I installed Suse 9.1 with 3 Terratec EWS88D soundcars: Alsa properly recognize the 3 cards installed (3 Ice1712 divers are loaded), now I'm using BruteFir which require to set the Alsa's parameters for managing with multiple soundcards. The parameter is: hw:0,0 if you have a 1 card system, now does someone know how this parameters work with 3 soundcards? Thanks! Michele

19 years, 11 months

[linux-audio-user] alsa configuration for ews88d

by Vlada Karan

Recently I have bought new audio cards, RME HDSP9632 and Terratec EWS88D, but I am having problems fith both. Hdsp isn't working at all after flash update from RME site, saying that wrong irq or dma is causing that, but everything worked with older firmware revision. Ews is working, but kind odd, I only can use first ADAT channel, other channels aren't availible. I see three devices, but their usage is very unclear to me. How can I use other ADAT's and SPDIF. (rme card is DA converter in my configuration) Can anyone tell me how to configure system with .asoundrc, because I am newbie in Linux world.

19 years, 11 months

[linux-audio-user] Problem modprobing latency-test module for the latency benchmark: rtc -> no such device

by Ivica Ico Bukvic

Hi all, I've been trying to benchmark my notebook with the latency-test. However, after a successful compile, instructions tell me that I need to install the latency-test module before running tests. Yet, if I do modprobe latency-test, computer complains about rtc module saying "no such device" and ultimately fails to install the latency-test module which depends on rtc. I am using 2.6.5-mm5 kernel with some additional patches, none of which should necessarily affect the real-time clock (is that what rtc stands for?). The processor is mobile amd64 and the kernel when either compiled for i586 or K8 chips exhibits in both cases the same symptom. OS is Mdk 10.0 (community with all the patches that should bring it up to par with the official release, at least I think so). Any ideas? Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/

19 years, 11 months

[linux-audio-user] jamin: Splined length 1024 does not match BINS / 2 - 1 (1023)

by Tim Blechmann

hi all, i recently had a look at the lovely jamin mastering tool, but i get some error messages on the console: Splined length 1024 does not match BINS / 2 - 1 (1023) also, when saving a scene, the 30 band eq settings don't get saved. (if it's important: i installed jamin from the gentoo ebuild) what have i been doing wrong? cheers... Tim mailto:TimBlechmann@gmx.de ICQ: 96771783 -- The only people for me are the mad ones, the ones who are mad to live, mad to talk, mad to be saved, desirous of everything at the same time, the ones who never yawn or say a commonplace thing, but burn, burn, burn, like fabulous yellow roman candles exploding like spiders across the stars and in the middle you see the blue centerlight pop and everybody goes "Awww!" Jack Kerouac

19 years, 11 months

[linux-audio-user] A most bizarre problem.

by Malcolm Baldridge

A few weeks into my admittedly bizarre Jack-based telephone logging + music-on-hold system, I ran into an interesting "bug". But first, the hardware in play: Motherboard: ECS P4IBMS (Pentium 4, i845 chipset) 00:1f.5 Multimedia audio controller: Intel Corp. 82801BA/BAM AC'97 Audio Card: Intel 82801BA-ICH2 Chip: Realtek ALC200/200P rev 0 ALSA-1.0.3b w/1.0.4 drivers + libraries, jackd is from late April 2004 CVS. My jackd invocation line is as follows: jackd -R -d alsa -S -C -p 4096 -M -n 2 -r 11025 I have an AT&T MERLIN phone system in my closet along with my main home server (see above hardware description). I have my PCM (Line out) plugged into the MERLIN system's music-on-hold port, that works well. I have two phone lines hooked up to JD Audio Phone Patch boxes to provide local + remote side level correction (trans-hybrid technology), which feeds into my Line In jack [each patch box provides a mixed in/out audio source on a mono output, so I have Line 1 feeding Left, and Line 2 feeding Right]. This too works beautifully with my vox-activated audio logging program which I'd adapted from one of the jackd sample clients. [It's ended up quite something else, but not too bad if I say so myself.] PROBLEM: When I have someone on hold on Line 2, say, and someone else is talking on Line 1, the person on Line 2 hears the music-on-hold *AND* the conversation of Line 1!! It gets better. If I dial out on Line 3 [lines 3+4 are outbound only], put that call on hold, and there are people talking on Lines 1+2, the caller at the other side of Line 3 hears not only the music-on-hold, but BOTH Line 1 + 2's conversations!!! YIKES!!! This was a big shocker. Naturally, I never did this sort of testing, being more concerned with DC bias, filtering curves and bandpass settings, and which compressors to use for the online-browsable phonelogs as well as the losslessly-compressed (FLAC of course) data archival logs. Obviously the PCM output is being mixed into whatever feeds into the Line input of the AC97 codec. This is annoying, and I can see no way of switching it off. I've explored every Mute/on/off option in the alsamixer, alas the dark fire did not avail me. I see nothing about "hardware monitoring" as an option for this chipset, however Hammerfall and others are mentioned as supporting this feature; nothing in the intel-8x0 driver suggest this is a supported feature. I'm relatively new to PC audio hardware, but has the LINE-IN been fed into the PCM output mixer from the beginning? Is this a funky chipset bug or weird issue or interaction? I know that the output mixer is fed audio from the CD and the WAVE device, but it seems a bit strange that the LINE-IN port is also routed through in this way. I can't try this under Windows, as this machine has been kept clean from the Dark Software of Udûn since its deployment. I obviously can't mute the PCM output, since that's the music-on-hold. After much experimentation, I figured I'd try the Microphone input instead. It seems noisier, even with the +20dB turned off. The impedance is quite a bit different, though fortunately I can get by with the gain set to the bare minimum [3 in the ALSA mixer, which is the smallest non-zero value I'm allowed]. But the cross-surveillance bonus to the music-on-hold problem is gone. Maybe I can do something tricksy like emit the music-on-hold on the right channel only, and hook up Line 2's audio feed into the left channel of the LINE-IN. This is a cruel hack which depends on the mic+mix functions in the codec doing their job properly, which is a huge ask. Ah well. I'm back in action, only now I have only one channel of recording capability. :( :( Is there a mixer option/setting to disable this "pass-through" routing of LINE-IN to the LINE-OUT sound port? In Mordor where the PC platform lies, =MB= -- A focus on Quality.

19 years, 11 months

Re: [linux-audio-user] jack and sblive inputs

by michelangelo79＠libero.it

---------- Initial Header ----------- >From : linux-audio-user-bounces(a)music.columbia.edu To : "A list for linux audio users" linux-audio-user(a)music.columbia.edu Cc : Date : Sat, 22 May 2004 07:00:02 -0500 Subject : Re: [linux-audio-user] jack and sblive inputs > To use a mic just use a stereo to dual mono splitter in the line input > and use either the left or right channel. Thanks, this is a practical solution, but wishing to use both mic and line-in? Is it possible that there is no way to enable the mic plug? Michelangelo

19 years, 11 months

[linux-audio-user] jack and sblive inputs

by michelangelo79＠libero.it

Hi People! I've been searchin around to solve my little problem but couldnt find any solution yet. I have a Creative Sound Blaster Live! sound card and the problem is that when i star jack deamon, i have only 2 input channel and 2 output channel. Not bad for the output (even if my sound card has more, i need only that two). Too bad for the input! I noticed that the two output channel are not really two separate channel, but left channel and right channel of the same output plug in sblive. In the same way the two input channel are left channel and right channel of the line-in plug. I wonder how can I add a third input channel for the microphone plug? (just one 'cause mic is mono). To start jack i use qjackctl, i tried to put 3 input channels in the setup section, but deamon do not start with this configuration! : configuring for 44100Hz, period = 1024 frames, buffer = 2 periods Couldn't open hw:0 for 32bit samples trying 24bit instead Couldn't open hw:0 for 24bit samples trying 16bit instead ALSA: cannot set channel count to 3 for capture ALSA: cannot configure capture channel cannot load driver module alsa 13:33:10.468 JACK was stopped successfully. While when starting jack deamon with only 2 channels (one plug) the command line used by qjackctl is: jackd -v -R -t500 -dalsa -dhw:0 -r44100 -p1024 -n2 -i2 -o2 Any help is apriciated! :) Thank you all! Michelangelo

19 years, 11 months

[linux-audio-user] M-Audio Delta 66 breakout box cable

by Cal

howdy folks, WRT an M-Audio Delta 66, can anyone tell me if the 15 way cable connecting the card to the breakout box is a straight through cable, or does it have funky things like special shielding or oddball connection patterns? cheers, Cal

19 years, 11 months

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Linux-audio-user May 2004