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?
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
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
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.dkhttp://modlys.dkhttp://virb.com/atte
... 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
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
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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
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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