Lynx PCIe AES interfaces: AES16 and E22

Discussion in 'Digital: DACs, USB converters, decrapifiers' started by purr1n, Sep 8, 2016.

  1. Garns

    Garns Friend

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    Yesterday I revisited some of the claims I made earlier in this thread to check I'm not going crazy. I stand by what I said about disconnecting AES ground: in my system it yields a noticeable improvement.

    I also revisited this:
    And here I must report user error. The differences I heard were real, but have nothing to do with ASIO vs WASAPI but rather to do with bit depth. In Foobar, the ASIO drivers choose their own bit depth, and seem to do so successfully. The WASAPI drivers let you choose it yourself. I hadn't really considered this and left it set to 32 bit. Turns out that this is not the correct choice into Yggdrasil. Once I set it to 24bit the WASAPI immediately improved, and I cannot now hear any difference at all between it and ASIO.

    Long story short: check your output bit depth in particular if you use Foobar. If it's 32bit, try 24 instead. Sounds better to me.
     
  2. Torq

    Torq MOT: Headphone.com

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    If there's transformer isolation for the coax-output on that board it's either well disguised, in the wrong place, or in the connector assembly itself (never seen that with what amounts to an RCA jack, though it's not uncommon for RJ45/Ethernet connections).

    (Or my cold is affecting more than just my ability to hear shit.)
     
    Last edited: Feb 20, 2017
  3. mrflibble

    mrflibble Friend

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    I used to have the Xonar Essence STX II . IMO, the Coax didn't sound quite as clear as using a glass TOSLink cable. The Xonar TOSLink out was equal to the Elfidelity AXF-8 TOSLink out.
     
  4. mrflibble

    mrflibble Friend

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    Not sure whether to post this here or start a new thread in the config forum.

    I'm using an RME 9632 on Linux. I have it working with my custom ALSA config, but there are a few problems. I have HDSPConf and HDSPMixer installed to control the sound card, the standard ALSA mixer is not fully supported. I can't get the card to support sample rate switching from an application. The problem is that it has three modes, Single-Speed, Double-Speed, and Quad-Speed - these support sample rates of <=48000Hz, 64000Hz to 96000Hz, and >= 128000Hz, respectively. With each mode the number of channels is reduced and the channel numbers change, this then makes my ALSA configuration invalid. There no way to automatically switch modes from within an application, only by using the HDSPConf utility.

    In summary:

    1) Auto-switching to a sample rate in a different mode does not work automatically from an application, only when selecting the sample rate from the HDSPConf utility.
    2) If, for example, I switch to Double-Speed mode and my config is set up for Single-Speed, my configuration will break.
    3) If, for example, a 96Khz file is played in Single-Speed mode it is resampled to 48Khz, even if using my "bit_perfect" plugin (see ALSA config below). I do not want the file to be resampled.

    I don't think there is a solution to the problem of automatically setting the Speed mode from an application, There is some information here: https://www.linuxjournal.com/node/7024/print , regarding setting the Speed mode and having to restart applications:

    I may be able to hack something together with my ALSA config, I will give it some more thought. If I can overcome having to state the number of channels, I may have a chance. This is what I have so far:

    Code:
    # Remember to disable PulseAudio
    
    # Identify the card by name to avoid maintaining hardcoded numbers ("hw:2,0") that may
    # change depending on what else is plugged into the PC
    #
    # Name: cat /proc/asound/cards or aplay -l
    # http://alsa.opensrc.org/Proc_asound_documentation
    #
    
    # HDSP Configuration
    # https://www.forum.rme-audio.de/viewtopic.php?id=9166
    
    pcm.HDSP {
      type hw
      card "DSP"
    }
    
    ctl.HDSP {
      type hw
      card "DSP"
    }
    
    
    # DMixer playback, use instead of PulseAudio
    # http://mondogrigio.blogspot.co.uk/2014/10/how-to-disable-pulseaudio-and-enable.html
    # http://www.alsa-project.org/main/index.php/Asoundrc
    
    pcm.playback {
      type dmix
      ipc_key 1024
      slave {
        pcm "HDSP"
        channels 12
        format "S32_LE"
        rate 48000
        buffer_size 2048
        period_size 1024
      }
      bindings {
        # ANALOG
    #    0 10
    #    1 11
        # SPDIF
        0 8
        1 9
      }         
     
      hint {
        show {
          @func refer
          name defaults.namehint.extended
        }
        description "HDSP Playback"
      }
    }
    
    pcm.capture {
      type dsnoop
      ipc_key 1025
      slave {
        pcm "HDSP"
        channels 12
        format "S32_LE"
        rate 48000
        buffer_size 2048
        period_size 1024
      }
      bindings {
        # ANALOG
        0 10
        1 11
      }
    
      hint {
        show {
          @func refer
          name defaults.namehint.extended
        }
        description "HDSP Capture"
      }
    }
    
    pcm.dmixer {
      type plug
      slave {
        pcm {
          type asym
          playback.pcm "playback"
          capture.pcm "capture"
        }
      }
    
      hint {
        show {
          @func refer
          name defaults.namehint.extended
        }
        description "DMixer"
      }
    }
    
    
    # Bit Perfect Playback
    # http://audiokarma.org/forums/index.php?threads/stop-resampling-under-linux.575308/
    # http://www.audiomisc.co.uk/Linux/ALSA/NoMoreSilence.html
    
    pcm.bit_perfect {
      type plug
      slave {
        pcm "HDSP"
        channels 12
        format "S32_LE"
        rate "unchanged"
      }
      # ANALOG
    #  ttable.0.10 1
    #  ttable.1.11 1
      # SPDIF
      ttable.0.8 1
      ttable.1.9 1
    
      hint {
        show {
          @func refer
          name defaults.namehint.extended
        }
        description "Bit Perfect"   
      }
    }
    
    
    # Default sound chain
    pcm.!default {
      type plug
      slave {
        pcm "dmixer"
    #    pcm "bit_perfect"
      }
    
      hint {
        show {
          @func refer
          name defaults.namehint.extended
        }
        description "Default Playback"
      }
    }
    
    If anybody has any ideas or has already overcome this problem, I would be very grateful.
     
  5. mrflibble

    mrflibble Friend

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    For the analog output I could route it to channels 0 and 1 because they are present in every Speed mode. The SPDIF output changes channel with Speed mode however...
     
  6. mrflibble

    mrflibble Friend

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    Okay, I've given up on that. I'm pretty sure it is not possible. The best I could manage is to set the card to Double-Speed mode in a bash script that is run at startup and then define two pcm's in the ALSA config, one for 88.2Khz and an other for 96Khz. Then I select which pcm I want to use to upsample, for 44.1Khz material I use the 88.2Khz pcm and for 48Khz material I use the 96Khz pcm. This works well from the command-line with aplay and with Audacious. Unfortunately, it doesn't work with DeaDBeeF, the music has dropouts.

    Here is my script:

    Code:
    #!/bin/bash
    
    # Connect channels
    # https://www.mail-archive.com/[email protected]/msg12802.html
    # https://www.mail-archive.com/[email protected]/msg12803.html
    # https://www.mail-archive.com/[email protected]/msg12813.html
    
    for out_left in {0..14..2}
    do
      let out_right=$out_left+1
      let in_left=$out_left+16
      let in_right=$out_right+16
      amixer -q -cDSP cset numid=5 $in_left, $out_left, 32768
      amixer -q -cDSP cset numid=5 $in_right, $out_right, 32768
    done
    
    sleep 1
    
    # Set to double speed
    # https://www.linuxjournal.com/node/7024/print
    amixer -q -cDSP cset numid=11 6
    
     
  7. Changeling

    Changeling Tube Slut

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    Happy to see this. Confirms my findings. I wanted to hear a difference and at first I thought WASAPI was better and then I sort of changed my mind finding WASAPI to be a bit hot, but in the end, as of right now, I can't hear any difference between the two with same settings.
    I wouldn't be able to ABX one from the other and don't think I need to either. I agree with @Garns that if you hear a distinct difference you might want to look for settings that might be the cause rather than a general difference in sound. With Lynx AES16 PCI there's no difference to be heard.
    Feels good to live and learn.
     
  8. e.schell

    e.schell Friend

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    Planning to order an E22 tomorrow...been digging around and reading through the thread haven't made it all the way through...for the sake of making sure i do this the right way and order everything i need to get it up and going at once... what is the best way to connect bnc spidf from E22 to Gungnir Multibit.... switch the digital I/O via jumpers from AES to spidf on the card then use one the Sescom SES-AES-EBU-1 on the xlr output from the LYNX CBL-L22SYNC to a bnc spidf cable to Gungnir Multibit?
    My apologies if this exact question has already been answered... i plan to read through the rest of the thread but want to get this sucker ordered ASAP. Thank you.
     
  9. k4rstar

    k4rstar Britney fan club president

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    Jowls just answered you in the other thread... you can do it either way. Some claim that the BNC input of the Gungnir is slightly superior to the coaxial SPDIF. I wouldn't disagree but the difference is so slight it is not worth introducing a transformer in the path. Use the SPDIF output of your card.
     
  10. e.schell

    e.schell Friend

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    So far i'm under the impression about the E22 with LYNX CBL-L22SYNC from the manual and a few other posts that the bncs on the breakout cable are used only for world clock, leaving only the XLR as the digital output which can be either AES or SPIDF depending on how you configure the jumpers on the card... leaving the need to convert from XLR to BNC to feed Gungnir Multibit.
     
  11. Comzee

    Comzee Facebook Friend

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    I recently picked up a Lynx aes16e, unfortunately it creates a group loop with my system.
    It also generally picks up noise for my PC too, example is that I'll move my mouse, and I can hear that :D

    I've read of people off boarding the power to the pcie card, but I can't find a good solution.
    Any suggestions?
     
  12. Tachikoma

    Tachikoma Almost "Made"

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    The Lynx would have the same ground as the rest of the PC, so a ground loop with the card itself should be just about impossible. Maybe try putting the DAC and amp on the same power board as the PC?
     
  13. Comzee

    Comzee Facebook Friend

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    I read it could be a ground potential mismatch, and that's what it was.
    I had my PC power routing thru a surge protector, eliminated that from the scenario and it fixed the static.
     
  14. Comzee

    Comzee Facebook Friend

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    The static came back, I troubleshot the issue incorrectly. I have it figured out now in the sense of how, not really why tho.

    My normal setup is PC-Lynx -> Yggdrasil -> SE -> Fire-Bottle(Amp) -> speakers
    This creates static, which I'm hypothesizing is a ground potential mismatch issue (related to ground loop).

    I just started a process of plugging in and unplugging everything, different wall outlets etc etc....
    I found out that when I had an ampY connected via Yggdrasil XLR out, it stopped the noise.
    Although, only if a power cable is connected to ampY, regardless if ampY is on/off.

    It makes me think the Yggdrasil is looking for another ground path that matches the potential of the digital out my PC-Lynx is feeding the Yggdrasil.

    Keep in mind, none of this happens with my SU-1, a device with dedicated PSU and ground.

    I guess on one point I'm glad I have a method to eliminate the noise, but on the other curious on how I would fix it without the xlr ampY solution.


    @Marvey you were also correct, I was forced to really a/b the SU-1 and Lynx throughout my troubleshooting.
    The Lynx is hands down better, very noticeable.
     
  15. Garns

    Garns Friend

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    I also had grounding issues feeding Yggdrasil from an AES soundcard. If you have determined that grounding really is the issue, a simple solution is to use a cable with pin 1 disconnected on the XLR at the Yggdrasil end.

    You can buy a ground lifter XLR adapter to go on the end of your existing cable, but in my experience they seem to be badly made. I would suggest getting a custom cable made up (or DIY one if you are so inclined). This will work with the AES16e:

    https://www.redco.com/Custom-Cable.html?cableid=PivCqd

    Adjust length to taste. This goes in Port A of the AES16 and receives the first AES output from the card.

    The main potential issue with doing this is RF interference. With pin 1 disconnected the XLR shield acts as an antenna. In my case, I had to be careful not to lay the AES cable near any power cables (due to RF spikes when switching equipment on/off) but then everything worked fine (and sounded better than with pin 1 connected).
     
    Last edited: Mar 29, 2017
  16. Vent

    Vent Friend

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    So.... $800 for a soundcard that then requires a DIY hack to eliminate static. Things just got serious.
     
  17. Garns

    Garns Friend

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    To be fair, pretty much any two audio devices of your choice could have a grounding issue when you connect them up. My monitors have transformer balancing AND a ground lift switch AND a section in the manual about ground loops, so I think it's not an esoteric problem. At least with XLR there is an easy fix.
     
  18. GoodEnoughGear

    GoodEnoughGear Evil Dr. Shultz‎

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    There are XLR pass-through ground lift gizmos available as well if you're not wanting to cut into cables.
     
  19. Garns

    Garns Friend

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    They are excellent as a diagnostic tool. Although I somehow have a pair which are completely useless as the pins are the wrong size and don't easily fit in anything!
     
  20. Clemmaster

    Clemmaster Friend

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    Do you get static when all your components (including the PC) are on the same power outlet?
     

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