ardour / jackctl / delta 1010lt

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ariev3
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Joined: 2010-09-15
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Hi all,

I have bought myself a M-audio delta 1010lt.

This works:
- playing previously recorded audio with ardour
- making other apps send their audio to the card (like hydrogen)

What I can't get to work:
- hearing myself play through 1 of the 1010lt's inputs

It must be somewhere in jackd config, but I cannot get it right (have been trying off and on for a few days now).

Any help is greatly appreciated!

Best regards,
Arie Van Drie.

tomas vtipil
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Joined: 2007-11-13
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it's not the jackd thing.

you have to set up the hw monitoring of the soundcard itself. you can do it using alsamixer (type alsamixer in terminal) or via dedicated mixer application. for delta1010lt, envy24 control utility is what you need.

cheers,
tomas

paul
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@ariev3: you can also use Ardour for monitoring too (Options -> Monitoring -> Ardour does monitoring)

The latency is larger than when using the delta1010 itself, but its also a bit less hassle to set up in some cases.

ariev3
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Joined: 2010-09-15
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Yeah, great!

I did not succeed in doing it with the envy24 mixer, but alsamixer worked! Thanks for that.

Now, there is a bit of latency when monitoring directy through the hardware. I am using 2.6.32-22-generic #36-Ubuntu SMP and the relevant bit in /etc/security/limits.conf is

arie - rtprio 99
arie - memlock 3072000
arie - nice -19

Do you have any suggestion on how to minimize latency?

Thanks for the help so far.

Arie.

paul
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@ariev3: the delta1010 is supposed to have close to zero latency for direct monitoring. I'm not sure you have set it up in the best way. Generally people using envy24ctl for that.

Please remove the "nice" line from limits.conf - it is irrelevant and unfortunately a lot of web pages suggest adding it even though it has nothing to do with low latency audio.

ariev3
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I have been changing some settings in jackd config. I started out with a latency of 42 ms. This was with frames/period of 1024. I have set that to 64, and now latency is 2.9 ms. The jackd manual page suggests to set that as low as possible for having low-latency, but xruns are happening for everything below 1024.

I am going to try now with an rt kernel, even though it is said that my current kernel would be able to run stuff in realtime.

Thanks so far.

Arie

ariev3
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Ok,

rt kernel (2.6.31-11-rt #154-Ubuntu SMP PREEMPT RT) clearly performs better: frames/period is now at 256 without xruns. Latency is at 11 ms. I can work with that; a bit less would be nice though.

btw Paul: the nice line in limits.conf may have nothing to do with low latency, it does let me run applications with higher cpu priority, which is, for especially for audio applications, not a bad thing. Seen from that perspective: are you sure about removing it?

Grtz
Arie.

adaw
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Joined: 2009-02-24
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You can do it with the hardware monitoring (Digital Mixer) of the 1010lt.
One thing you should know: The digital mix can only be sent to analog out 1/2 or to spdif out.
Use the alsa Envy24 Control Utility to set it up.

It should work like this:

Setup:
Play a stereo track in Ardour and record while doing 1010lt-mixer monitoring.

Ardour out: jack playback 1/2
Record in (mono) : 1010lt input 1 = jack capture_1
Monitor out: 1010lt output 1/2

Envy24 Control Utility, Tabs:
Patchbay Router: H/W Out 1 set to Digital Mix L; H/W Out 2 set to Digital Mix R
Monitor Inputs: Unmute H/W In 1 and rise the left and right fader to send the input to the digital mix.
Monitor Outputs: Unmute PCM Out 1 and PCM Out 2 and rise the left fader of Out1 and the right fader of Out 2 to send to signal from Ardour to the digital mix.
Analog Volume: Check DAC 0, DAC 1, ADC 0 if they are set properly otherwise the signal won't be routed.
Profiles: You can save the settings here.

It's some time ago I did this, but it should work.
I Just had a look at the diagram i made the last time i was completely lost with the 1010lt. mixer ;)
Now it here: http://www.alsa-project.org/main/index.php/Delta_1010lt_Mixer

paul
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@ariev: nice only affects SCHED_OTHER scheduling priority. It is irrelevant for SCHED_FIFO or SCHED_RR, which are the only priorites that realtime audio applications should use in any real-life production context.