EQing / filtering the AC97 input (mic) for Skype etc

Discussion in 'Effects and the DSP' started by christopherw, Nov 16, 2010.

  1. christopherw

    christopherw New Member

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    A while back I fancied EQing my mic input for Skype or Teamspeak etc. How hard could it be? As it turns out, apparently impossible...

    I have the Sound Blaster Audigy 2 PCI card, running kX 5.10.00.3550. I have a fair understanding of the DSP and I've hacked together custom signal paths and used a variety of modules before.


    However, even with EVERYTHING in the DSP disconnected and even removed - even with the prolog removed! - Skype and other Windows apps can still get the incoming AC97 Mic signal via 0/1. Whenever I disconnect or add things to the signal chain for the ASIO prolog -> epilog or prolog -> xrouting -> epilog connections, that can be heard in things like Cubase (which use ASIO) but Windows apps seem completely unaffected by an empty DSP. It's intensely frustrating.

    What am I doing wrong? :mad:
     
  2. Russ

    Russ Well-Known Member

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    By default, Windows recording is done inside of epilog (which makes it impossible to filter with other plugins), so as long as epilog is loaded, WIndows recording is possible.

    Also, note that a blank DSP does not work as you would expect it would, so do not test with it completely blank (really, epilog (or another output plugin) should always be loaded or you can get unexpected results).

    To disable the default Windows recording path (for AC97), make sure that the "AC97" slider on the "Recording" page of kxmixer is muted.

    Instead of the above, use prolog and the "AC97" slider on the"Ins and Out's" page of kxmixer (same as you would do with ASIO recording), connected to the RecL/RecR pins of epilog (with EQ somewhere between).
     
    Last edited: Nov 17, 2010
  3. christopherw

    christopherw New Member

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    Hi Russ, never thanked you at the time for your advice - was most useful! Tonight I decided to have a play (to get a decent compressed, de-essed and EQed feed for Skype) - inputs routed direct from prolog via three FX and to RecL/RecR - with a loopback to In0 and In1 also for monitoring. AC97 fader is muted in Recording, AC97 in Ins 'n Outs is controlling the master levels.

    Very happy with the setup, cheers for the pointers - I would've never figured it out I think otherwise (and I did RTFM beforehand!) This community is bloody brilliant.
     
  4. Russ

    Russ Well-Known Member

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    Glad I could help :)
     
  5. christopherw

    christopherw New Member

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    I'm going to pick your brain if I may:

    [​IMG]

    That's my current routing. Got some gentle EQ and de-essing after the compressor (FM-style vocal compression! :D); it works beautifully with Skype - I can mix in stereo mix audio should I wish (that's my bottom-most Stereo Mix + Gain instance) and I can adjust Front and Rear output monitor levels which is useful for me as I use headphones. It's a bit of a dirty kludge but overall it works very well given its slightly fiddly routing.


    However Teamspeak seems to disregard the bypassing of the output routing - and always outputs sound on 0/1, even using DirectSound. I'm using 4/5 for outputs on Skype - effectively mix minusing the Skype caller from the return. A bit perplexed by Teamspeak's behaviour, have you seen other programs ignore this routing before and just directly route to outputs?
     
    Last edited: Apr 19, 2011
  6. Russ

    Russ Well-Known Member

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    What OS are you using?

    I have not heard of any routing issues related to any of the wave devices, other than Master Mixer (Vista/7). Have you tried connecting peak plugins to FxBus 0/1 and 4/5 to verify that TS is not using 4/5?

    Aside from the above:

    It is a little hard to follow the routing in your picture, but, Surrounder is set to 2.0 (surround off), so there is no signal to the second inputs of Stereo Mix (i.e. only front (top 2) outputs of Surrounder are active (and top 4 inputs)). With that setting, you would (basically) only hear FxBus audio from FxBus 0/1 and 2/3.

    Also, on your recording path, it is hard to see which signal is going to the 2nd inputs of Stereo Mix + Gain (is it sL/sR?), but, there is a potential feedback loop there (depending on which signal it is, and Surrounder settings).
     

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