Linux-audio-user November 2009

linux-audio-user@lists.linuxaudio.org

160 participants
143 discussions

by David Baron

Get this: ov 16 22:29:42 d_baron das_watchdog: started Nov 16 22:29:42 d_baron das_watchdog: Warning. The priority of the "ksoftirqd/0" process is only 0, and not 99. Unless you are using the High Res Timer, the watchdog will probably not work. If you are using the High Res Timer, please continue doing so and ignore this message. Nov 16 22:29:42 d_baron das_watchdog: Warning The "ksoftirqd/0" process is running SCHED_OTHER. Unless you are using the High Res Timer, the watchdog will probably not work, and the timing on your machine is probably horrible. I am using HIgh Res Timer, 1000, though pulseaudio's alsa interface does not think so. However, it might be best to set the ksoftirqd stuff. How do I do it?

16 years

Re: [LAU] open hw soundcard

by Mike Mazarick

Here are a couple of observations/suggestions: 1. There is a lot more that goes into a product than design (marketing, support, etc.). It may be a good idea to consider partnering with a company that already provides similar services (hardware, support, etc.) because they already have the distribution and marketing set up and operational. It is important not to discount the amount of effort and investment needed to make this happen and to understand that they will need to make some type of investment that has to pay off for them. This can still be done in the open source context, but there has to be some plausible answer to the question "If I put a bunch of money forward for doing the necessary 'missionary work', what is to stop anyone else from coming along and taking advantage of this investment and cutting off my legs with a lower price for the same thing after I've made the investment?". We all realize that an upfront investment in both R&D and marketing is needed for a successful soundcard, because in electronics, the unit costs go down dramatically after you pass each of the 1000, 10,000, and 100,000 unit thresholds. It almost makes the investment in R&D and marketing/sales look like a single cost with one cost for any number of units. Who knows? With an adequate explanation and a plan, one of the existing 'majors' may even provide some funding for this idea. 2. The proposed open hw soundcard is correctly focused on Ethernet. It may turn out that in order to use standard networking gear, that TCPIP be eliminated so you won't be limited by packet size. One way to get this done without having to think about it a lot is to adopt the ATA over Ethernet standard (it's open source) and make the audio read/writes look like a hard disk access. Another potential low cost solution that may dovetail this idea is to use the eSATA ports that a lot of more modern motherboards have built in. If the audio needs to be routed with a router over long distances, this can be attacked separately. It is doubtful if this can be done without some amount of latency. 3. I like the idea of using custom IC hardware. However, I also like using a FPGA (or PLD) to provide the 'glue logic'. One thing that has been learned from the effort of getting an old soundcard driver working (gadget labs - no hardware DMA built in) software DMA support creates a lot of extra effort but reduces the hardware cost. For a network device the equivalent is to use something like RDMA (remote DMA) in Infiniband. Some thought should be given to how to implement in hardware a direct memory to memory transfer without interrupting the CPU on the remote computers. 4. I obviously haven't read or researched enough about this, but some discussion would be warranted on what this will do for the musician that he can't do now. It's probably worth repeating (I'm too thick to get it in just one pass). 5. Obviously, it is important to stick with open source system as much as practical and make sure that the windows drivers are open source which should be part of the license for using the hardware (even if it has to inter-operate with ASIO, WDM, etc.). Best wishes and good luck!! Mike Mazarick

16 years

Re: [LAU] Damn Ubuntu 9.10 update

by Marco Castorina

Dear all, sorry for the late reply, I've been quite busy between studying and house maintenance. First of all thank you everybody for your answers. I was actually launching the older version of the kernel, and this was messing up modules loaded. Everything is working fine now, except that I get more xruns using Jack than before. I'm actually using the standard kernel (2.6.31-14) and Jack 1.9.3. Maybe I'll give Jack 0.117 a try or using a custom RT kernel, seeing if I can fix the problem. I'll make some experiments and let you know. I hope also to post my song soon. Thanks again! Marco Send instant messages to your online friends http://uk.messenger.yahoo.com

16 years

Re: [LAU] OT: online music promition/what do you think of bandcamp?

by Atte André Jensen

Lorenzo wrote: > Btw, I didn't say I love flash :) what I mean is that it look we're > stuck with it as it's being used by most hosting providers. I don't love flash either, I love my wife and kids :-) >> the possibility of easy donation, the spread the word on >> facebook/twitter feature, the selection of download formats, the >> scalability (bandwidth) and more. > But really these aren't flash-dependent... you could probably easily set > them all up on your own website. The only real advantage of this kind of > site is hosting. Interesting, hmmm. Should be possible. Could you post links to a few sites that does this in an elegant way? I haven't tried, but I'm pretty sure it's also possible to pay with other things than paypal (for instance visa) on bandcamp. For those not having a paypal account (I guess about 95% of the people I know bought my first CD) this makes a great difference. This surely isn't possible (without hastle + expences) to setup yourself. This is all guesswork for me, since I haven't run anything but linux for 10 years, but my feeling is that "all the other users" (those with evil OS'es) don't care or think about flash. They just surf around and occasionally hit a play button on a music page, flash or not. -- Atte http://atte.dk http://modlys.dk http://virb.com/atte

16 years

Autotuner for Linux?

by Giorgio - Audiophilo

Hi, i'm looking for an autotuner (to tune bad singers' voice), is there anything native for Linux? Thanks G

16 years

yoshimi records

by cal

... maybe. I've gotten Record to the point where it hasn't done anything violent to me for a while, but it still has plenty of potential for unexpected shortcomings. The story so far ... - new Audio/Midi settings - default recordings directory and a checkbox for 32 bit float wavs. The default is 16 bit shorts. The wavs are courtesy of libsndfile, so that's a new dependency. - Select your wav file via the 'Wav file' button. Opening an existing wav starts in append mode. - Orange rewind button does a rewind/overwrite. - Green starts recording. - Blue pauses recording. Yes, that's a weird choice of colours. Send me some hex colour numbers and I'll gladly change them. To be honest, I'm struggling to hold focus well enough and long enough to give it the sort of testing it needs. There's more than a few different usage paths and despite best intentions, serious disciplined testing just isn't happening. So I'm asking for some help with testing. Background ... if record is running, at the point where audio is generated it gets fed off to a fifo. The buffer on that is currently set to 256k. For no particular reason I can think of I felt more comfortable with it at 512k. I've no idea how low that can go before the mechanism chokes and splutters. Yet another thread reads from the fifo and writes it to wav file. Does it properly manage the integrity of wav files in all possible use cases? I think so, but I'm not at all sure. How busy can the system get before audio and/or recording quality fails? Don't know. When you find it broken, the best I can promise is that I will try to fix it :-). <http://www.graggrag.com/yoshimi/yoshimi-0.037.tar.bz2> cheers folks. Cal

16 years

M-Audio Profire Lightbridge

by rosea grammostola

Hi, Does someone know whether this device works on Linux or not? M-Audio Profire Lightbridge http://www.ffado.org/?q=devicesupport%2Flist&filter0=M-audio&filter1=Profir… Regards, \r

16 years

Damn Ubuntu 9.10 update

by Marco Castorina

Dear all, today I made the biggest mistake of my life I think, I upgraded from ubuntu 9.04 to 9.10. Everything is working almost fine, but the audio is gone. If I try to boot from the CD, there's no problem, but when I get back to the actual system, there's no sound. The card is recognized: marco@marco-x41:~$ lspci | grep audio 00:1e.2 Multimedia audio controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) AC'97 Audio Controller (rev 03) !!Soundcards recognised by ALSA !!----------------------------- 0 [ICH6 ]: ICH4 - Intel ICH6 Intel ICH6 with AD1981B at irq 22 but not for playing audio: APLAY aplay: device_list:223: no soundcards found... ARECORD arecord: device_list:223: no soundcards found... Also when I try starting Jack: Could not open ALSA sequencer as a client. ALSA MIDI patchbay will be not available. ALSA lib seq_hw.c:457:(snd_seq_hw_open) open /dev/snd/seq failed: No such file or directory and there are no sound cards to select. Any ideas? Thanks, Marco Send instant messages to your online friends http://uk.messenger.yahoo.com

16 years

status usb2 for sound

by Hartmut Noack

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, USB1 works OK with ALSA but what is the status for USB*2*-Soundcards? I cannot find any useful info on http://www.alsa-project.org/ . So what are you experiences? Anybody out there working on drivers for stuff like this: http://de.m-audio.com/products/de_de/FastTrackUltra.html ..that is not class-compliant?? Or is it working already for anybody out there? thanks for any hints ;-) best regs HZN -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkrylYkACgkQ1Aecwva1SWNrYQCgjSLvflzTCgV1j14fYbM9u+56 r9AAn1CPMimhnsrWyIUsy0gG7jr8fLu+ =fHXW -----END PGP SIGNATURE-----

16 years

Fluidsynth, soundfonts, jack, and latency

by Guru Prasad B. R.

Hello everyone, So I've been trying out AVLinux. By the looks of it, quite impressive - but I soon ran into a rather serious issue, but I doubt whether it has anything to do with AVLinux or not. I need your help in this knowing for sure. My usual setup involves a piano sample on Linuxsampler and a strings soundfont on qsynth. I had an unusual gig last week that required no piano sounds, but instead use of multiple layered samples. I therefore used only qsynth, with 3 engines running concurrently. So in my brand new AVLinux setup, I loaded up my samples, and had sooperlooper running 2-3 loops concurrently, all on a core2-duo macbook. Initially, all was ok. No noticeable latency, no xruns whatsoever.But during the rehearsals, this happened: after sometime, specially at some point when sooperlooper was running full steam, suddenly there was this HUGE latency, coupled with lots of dropped notes. As in, I could hold down a dozen keys simultaneously on the midi controller, and not all notes could be heard; those that did, would be heard at various instants upto about half a second later. Completely unuseable. This would persist even after restarting the jack server, but dissappear on rebooting the OS. To me, what is scary and irksome, is not (only) that this glitch happened consistently, but that there were no error messages in the qjackctl window. No xruns. Nothing that suggested that something had gone terribly wrong somewhere (which it had). I actually compromised on my gig sound, by offloading a lot of samples what I wanted layered. I don't know if it was because of this, but I didn't notice this particular issue on the gig itself (but it gave me sleepless nights prior). But then, 5 mins before the start of the program, I was left with no audio out of my system! (again, no error messages). Restarted jack server, and everything back to normal. Again, no error messages/warnings. Not so much as a by-your-leave. ;) I'm now intent on finding the source of the problem. I'm looking for pointers... where do I begin? Where's the 'smoking gun'? Thanks in advance! Cheers, Guru

16 years

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Linux-audio-user November 2009