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  • ASIO overload means basically that audio data cannot be delivered *in time* to the device, which can have several reasons.
    whereas formerly in most cases the PCI-bus was the bottleneck it can be any bus between your drive having stored the sample data and the audio output - a few things to consider:
    - you should not record to the same drive you are reading sample data from
    - make sure your drive is able to deliver the requested amount of data
    - make sure you don't have any other firewire device on the same FW-bus
    - latest drivers, audio settings properly configured (eg. not 2 applications share ASIO)

    though you might not like another technical hint: one stereo voice needs 176 kB/s, in certain cases (eg. crossfades) even more and you should not expect signifcantly more than 25 MB/s throughput from a single harddisk at average access for sample streaming. do not rely on marketing-speech like *up to 100 MB/s* or similar, such figures are meant as boosts from the cache.
    christian

    and remember: only a CRAY can run an endless loop in just three seconds.
  • the numbers I've quoted were in "stop" mode (not playing),
    so there's nothing to read yet...

    when I hit play the ASIO meter jumps up only a little.

    anyway I have 4 SATA2 drives - system, projects/audio, vsl1, vsl2
    + video on ext. FW800

    the Fireface runs on FW400

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    @Abel said:

    48k is the film industry standard


    Not too sure about that...

    J.

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    @Abel said:

    48k is the film industry standard


    Not too sure about that...

    J.

    please elaborate.

  • jerome, i remember at least earlier media compser and symphony have been always kind of troublesome (in detail the meridien board) with 44,1, especially if somewhen (eg. for rushes) DV or digiBETA was in the game - but i'm not at all up to date here ...
    christian

    and remember: only a CRAY can run an endless loop in just three seconds.
  • abel, unfortunately we continue to be technical ...
    it is said that the firewire controller has only different ports and does internally not seperate the FW 800 and the FW400 bus (relying on the built-in downword capability instead of having different controllers) - it would be at least worth a try to reverse connections, i.e. fireface 800 on the FW800 and the harddisk on FW400. for video streaming the FW400 is still faster than data could be delivered from a single harddisk.
    in no case i would chain the devices.
    usually the RME fireface is not very sensitive having also a harddisk on the same controller, but who knows ....
    christian

    and remember: only a CRAY can run an endless loop in just three seconds.
  • http://techreport.com/reviews/2006q1/gigabyte-iram/index.x?pg=1">http://techreport.com/reviews/2006q1/gigabyte-iram/index.x?pg=1

    To add to the discussion....

  • yep, great concept - we already have some here (though it has not been easy to find a vendor) ... but my approach was more to use this device for the pagefile and i don't get it now how it might be related to this thread?
    christian

    and remember: only a CRAY can run an endless loop in just three seconds.
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    @Abel said:

    please elaborate.


    You didn't elaborate yourself... [:)]

    But I work in the film industry as a film composer / orchestrator / technician and out of all the people I know or worked for, I can say 48k is far from being the "industry standard" you're refering to.

    Some people work solely in 44.1k, some in 48k, and some switch between various sample rates.

    It's just not like *everybody* is working at 48Khz... it's as much a standard as 44.1Khz.

  • Hardware-wise, 48 kHz is defacto the accepted standard for audio in the video-world. This is not due to the higher audio-standards in this part of the business, as one might assume (quite the contrary!), but due to the fact that 48 kHz encodes much easier in a video-stream.

    Of course, people working _for_ video in the audio-world often use 44.1, which has to get converted to 48 at some point, though.

    (Edited for typos [:P])

    /Dietz - Vienna Symphonic Library
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    @Dietz said:

    ..., which has to get coverted to 48 at some point, though.


    Yes.

    If you don't have to run against a 48k digital clock, refrain from realtime upsampling.

    You will get far better results converting the finished mix with a high order interpolation tool.

    In realtime operation there are quality vs speed tradeoffs.

    /c

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    @Dietz said:

    Of course, people working _for_ video in the audio-world


    who cares about anyone else? [:P]

    J.