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This topic has 2 voices, contains 2 replies, and was last updated by  rayj 117 days ago.

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December 15, 2011 at 17:48 #1401

rayj

It doesn’t seem terribly active here, so excuse the noise post. I just wanted to say that I’ve recently migrated my systems to Arch, and it is working very well indeed.

If anyone is interested, let me know, and I’ll start rambling about configs and usage. I’m a self-taught wannabe IT fellow, and still working on my knowledge base to be sure, so I’m trying to keep my natural tendencies towards casual posting down…

January 23, 2012 at 11:34 #1429

Jon Kristian

Welcome:) We encourage everyone to use our forums for such things, feel free to share your configs and other experience. Also if you haven’t already, join us on freenode #archaudio for a chat:)

January 23, 2012 at 20:05 #1436

rayj

Well, hello then!

I’m currently reviewing options for a new configuration.

I have a couple of older systems (1x1smp AMD64+2GIG ram, 1x2smp AMD64+4GIG ram), and like the concept of using netjack to lock multiple clients into sync. Rather than focusing on realtime performance, I would rather focus on keeping playback streams in sync with networked clients via smpte or mtc (smpte for ‘soundtrack’ production with video, and mtc for sequencer+playback mixdowns).

I’m enthusiastic about the possible flexibility here…for example, if I decide to configure an older pc to run linuxsampler exclusively, I can chain it to the preexisting lan config via netjack, and run the entire enterprise without having to reconfigure other clients dramatically. The more cpu’s you can get working in tandem, the better…right?

I’m also looking into using less resource-intensive applications (cli-based, or server modes) on the clients. I’d rather have system resources devoted towards high-quality sync (hpet, etc.) and keeping clean A/D conversions on the hardware (48kHz/24bit being the target conversion rates). I’m from the school of ‘the fewer operations performed on recording sources, the better’, and plan on calibrating the recording side for ideal amplitude while avoiding transient peaks. I’ve found that this can be tricky, especially without some expensive test equipment.

My A/D converters are all pci M-Audio interfaces (a delta 1010 and a 66 chipset card), and I eventually want to sync them to an external clock. This will require some tests, as if it doesn’t clean up recording signals in addition to playback…well, it ends up not being cost effective.

I’m currently thinking of using ecasound as the recorder, and using ardour as an editing suite. Ardour is great, but I currently end up using only a fraction of it’s capabilities. However, ecasound has it’s own audio format…and I’m committed to clean source audio files.

I’m definitely open to suggestions. Eventually, I’d consider porting in the curious or helpful, and make it a sort of community test suite/reference.

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