Mac Audient ID14 latency issues

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Hi, 

I know this is probably the oldest question in the book, but I still can't seem to find a conclusive answer to it.

So, in the last session where I was recording vocals, there was a significant delay when the vocalist sang her lines resulting in a horrible sort of echo effect. We managed to get round it by her basically not hearing herself in the headphones and just relying on performance, but it was far from ideal.

Just to give some details of my set-up:

Audient ID14
Mac mini 2012- 2.3 GHz Intel Core i7, 16 GB ram
Recording onto an external drive.

The tracks are admittedly quite developed and running quite a few plug-ins, which presumably is slowing things down. Generally about 25-30 tracks, with compressor vst's etc. It's a pretty powerful Mac though, so I would have thought it could handle such things.

I tried turning the buffer size down and this did help somewhat, but didn't entirely resolve the issue.

Any suggestions?

I have another session on Monday and would like to have this sorted before we start if possible.

Many thanks,

Dom
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Comments

  • goldtopgoldtop Frets: 848
    Mix the backing down into a single stereo track, mute everything else and get her to sing against that?

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  • domforrdomforr Frets: 163
    Yes, that is an option, although it doesn't allow for any flexibility - i.e. turning parts of the mix up etc. Certainly an effective way to get around the issue though.
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  • CirrusCirrus Frets: 3518
    edited June 29
    Audient's driver has two different latency settings.

    There's one, which is number of samples in the buffer (32,64,128,256 etc...) where obviously the higher number = longer delay.

    The other one is fast, normal, safe, ultra safe etc and this is to do with how often the ID14 interrupts the computer's processor to send and receive packets of data; the audio going in and out isn't a constant stream, it's sent in bursts and this setting determines how often those bursts are sent. With slower settings, you'll find that some of the lower buffer settings get greyed out because the gap between the packets of data being sent would be longer than the size of the audio buffer.

    So for the lowest latence settings, you want to basically set both those as low as possible. If you get crackling and dropouts, start increasing them. The lowest buffer size you can go to will purely be a factor of whether or not your computer can keep up with the realtime processing. The lowest latency setting you can get will be affected by how good your USB busses are - if there's another USB device trying to interrupt the same USB controller in your computer, they might clash and that'll affect how low that setting can reliably go.

    Then you've got latency introduced by effects. Some plugins are very low or almost no latency, and some have huge amounts of latency and hence just aren't suitable for realtime monitoring.

    The Audient does have zero latency monitoring, controlled by the same mixer you set the latency settings with. Obviously you can't add effects to that monitoring, but if you want to add reverb or echo or something you can set it up so that you monitor the direct signal with zero latency, then also monitor a reverb effect 100% wet where a few tens of millisecond latency just don't matter - it's basically free pre-delay.

    Worth pointing out that the Audient drivers are famous for not being great at low latency in contrast to, say, RME that's bloody amazing at it, though the Audient is a great piece of hardware and Audient have just released new drivers which may or may not improve things depending on your system.

    And I'd be remiss if I didn't close by suggesting not using headphones when recording vocals, unless there's an overwhelming practical reason for doing so. It's just more fun, and there are no monitoring issues to contend with at all!
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • domforrdomforr Frets: 163
    Great stuff - thank you.

    A few things that I'm not clear on though:

    1) Can you set the buffer size on the audient, or is it just through Reaper? I can't see any settings for buffer size on the ID software.

    2) Do you know if there's any way of muting plugins on a mix in Reaper to see if these might potentially cause problems?

    3) tHere's a setting in Reaper under 'Preferences - Recording' that says 'Use audio driver reported latency. You can change the output and input manual settings here. Is that of any use? 

    4) Not sure what you mean by recording vocals without headphones? Sounds great in theory, but how on earth would you stop bleed from the track being recorded through the mic? 

    Thanks again,
    Dom
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  • CirrusCirrus Frets: 3518
    1.) It's in the "settings" menu at the top of this window;



    2.) It depends how latency compensation works in reaper - probably hard-bypassing the plugin using reaper's own bypass option, rather than any bypass button build into the plugin's GUI, will also stop it introducing its latency. You might need to stop and play to get Reaper to re-calculate all its latency compensations.

    3.) That's only for if the driver is reporting the *incorrect* latency - then reaper will think the latency is different to what it actually is, and might place recordings at slightly ahead or behind the beat because it thinks you were hearing the playback a few milliseconds away from where you actually were.

    4.) You minimise it using sensible microphone placement, using the mic polar pattern to your advantage, and playing back the track just loud enough for the vocalist to feel comfortable relative to how loudly they sing. What bleed there is is *never* an issue, unless you've done something daft like blast out bass notes that are subsequently changed or loads of tambourine or something. You might be surprised how many successful records were made this way.
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • domforrdomforr Frets: 163
    edited June 29
    Weird, I don't seem to have a settings option in my toolbar? Are you on a Mac?


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  • domforrdomforr Frets: 163
    Interesting re: the recording without headphones. I've genuinely never heard of that technique and always thought isolation and separation was the way to go. Just goes to show that vibe and feel conquers technical considerations.
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  • CirrusCirrus Frets: 3518
    edited June 29
    Ah, looks like on macs the latency settings are controlled within the DAW, so ignore that bit - I assume you've found the setting in reaper that controls that.
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • domforrdomforr Frets: 163
    Okay, that makes sense. Yes, I've got the Reaper settings to hand. Thanks again for the advice.
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  • Winny_PoohWinny_Pooh Frets: 2865
    I've been through this with an id22 and it's just not able to monitor the source through the DAW without latency. So I have the track in record mode but output muted in the DAW and monitor dry in the Audient software mixer.


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  • StuckfastStuckfast Frets: 333
    Cirrus said:
    Audient's driver has two different latency settings.

    There's one, which is number of samples in the buffer (32,64,128,256 etc...) where obviously the higher number = longer delay.

    The other one is fast, normal, safe, ultra safe etc
    This is only on Windows -- instead of writing their own drivers they license a third-party one that is a bit shit.

    On Mac OS they use the built-in USB class compliant driver which is actually quite good.

    If you want to reduce the latency to negligible levels you probably need to be running at a buffer size of 32 or 64 samples, *and* not using any plug-ins that impose a noticeable delay -- convolution reverbs, lookahead compressors and limiters, that sort of thing.


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  • domforrdomforr Frets: 163
    So you think running at 256 would be problematic? Don't think I've ever tried below that, but worth a shot I suppose.
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  • domforrdomforr Frets: 163
    Just tried 128 and it's unplayable, so 256 is the minimum for these sessions.
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  • StuckfastStuckfast Frets: 333
    edited July 2
    Assuming you're working at 44.1kHz then the absolute theoretical lowest round-trip latency you could possibly get with a 256-sample buffer would be (256x2)/44100, which works out at 11.6ms. However, that's not achievable in the real world because of additional latency introduced by the A-D and D-A conversion, any internal DSP in the interface, and the DAW/plug-ins. You're probably looking at 15ms or more in total which is likely to be something you can feel and hear.

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  • HeartfeltdawnHeartfeltdawn Frets: 11260
    Stuckfast said:
    Cirrus said:
    Audient's driver has two different latency settings.

    There's one, which is number of samples in the buffer (32,64,128,256 etc...) where obviously the higher number = longer delay.

    The other one is fast, normal, safe, ultra safe etc
    This is only on Windows -- instead of writing their own drivers they license a third-party one that is a bit shit.

    On Mac OS they use the built-in USB class compliant driver which is actually quite good.

    If you want to reduce the latency to negligible levels you probably need to be running at a buffer size of 32 or 64 samples, *and* not using any plug-ins that impose a noticeable delay -- convolution reverbs, lookahead compressors and limiters, that sort of thing.


    Audient's drivers are written by Thesycon. The tests on the new drivers over at Gearslutz suggest that theonly improvement comes when you run it at 96kHz. 
    Clarity over quantity.  
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  • CirrusCirrus Frets: 3518
    It's a shame as I'm really impressed with everything else about the id22 I've been using loads over the last three months - great preamps, conversion, form factor... Low latency performance is the only failing.
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • domforrdomforr Frets: 163
    Interesting. So it's a flaw within the Audient? I managed to work around it yesterday by bouncing down an mp3 of the tracks as they are and working off that within the session, with all of the individual tracks/plugins muted. Buffer was at 256 and it seemed to work okay.  Not an ideal scenario but an acceptable workaround to reduce stress on the day.  So ideally, should I record at 96kHz in future? Are there any downsides to this over 44.1?
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  • wave100wave100 Frets: 124
    It's not really a flaw in the Audient - all audio interfaces introduce some latency, but the trick is to use the mixer app (as illustrated above) to create a zero latency monitor mix which means that the voice monitor signal goes straight from input to output without going through the soundcard's buffers. The DAW then sorts it out using Automatic Delay Compensation and places the recorded audio at the correct location on the timeline. For this to work correctly, you do not use the track "record monitor" button as this will result in hearing both the zero latency signal and the latent signal at the same time. It also means you can't have the DAW fx on the track while recording, probably not too much of a problem for vocals but it means you can't track with a guitar sim plug in for example. The other occasion when latency is an issue is when playing soft synths.

    The amount of latency depends on the AI drivers and how fast your computer is - faster computer = lower latency. Also the more plug-ins your computer has to deal with, the higher the latency, so it's a good idea to do your tracking first then mix later. If your DAW has a track freeze facility, you can use this to reduce the processing power used, and thus the latency.

    I would not recommend recording at 96K, this doubles your file size thus resulting in a halving of the number of tracks your system can handle, plus any plugins will use twice the processing power. Judging from the buffer sizes you describe I would guess you are not using the most powerful computer around. Another thing to check (assuming you are on Windows) is that you are using the ASIO driver, selected from within Reaper.  With a Windows machine, it can often be useful to optimise your computer for audio recording - excellently described by Robin Vincent in his Molten Music channel on the tube. 

    In my experience, USB interfaces all have fairly high latency. RME interfaces have a reputation for lower latency, but do not come cheap.
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  • domforrdomforr Frets: 163
    Good info - thank you. I'm on a 2012 Mac mini but it's a decent spec:  Quad core, 2.3 GHz Intel Core i7, 16 GB ram, 250 gb SSD.

    I would have thought that would be more than enough for 30 odd tracks, but it has begun to struggle when loading up and the number of plug-ins seems to be taking its toll. This could be a problem when doing final mixes with everything running. Is the idea to run at a higher buffer when mixing as latency isn't really a big issue then?
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  • CirrusCirrus Frets: 3518
    domforr said:
    Good info - thank you. I'm on a 2012 Mac mini but it's a decent spec:  Quad core, 2.3 GHz Intel Core i7, 16 GB ram, 250 gb SSD.

    I would have thought that would be more than enough for 30 odd tracks, but it has begun to struggle when loading up and the number of plug-ins seems to be taking its toll. This could be a problem when doing final mixes with everything running. Is the idea to run at a higher buffer when mixing as latency isn't really a big issue then?
    Exactly right.

    Honestly, I've pretty much always just stuck with 1024 buffer size unless I've had a reason to lower it, whether I'm recording or mixing. But if low latency is important to you while recording, yes increase the buffer size to mix.
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • domforrdomforr Frets: 163
    ..and do you think an external SSD would improve performance? I'm currently using an oldish (non-ssd) Hard Drive to record onto but perhaps a small upgrade would improve things somewhat?

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  • CirrusCirrus Frets: 3518
    I don't think it'd improve anything - hard disk read/ write speeds are very rarely a bottleneck in the system. This is really an issue with brute CPU power - can it perform the real-time audio processing fast enough as dozens of separate applications (since that's what plugins are) demand time to do their calculations, while still doing all the other stuff in the background that keep the system moving?
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • wave100wave100 Frets: 124
    My machine is a 2012 i5 MBP with 8 gig ram and an ssd with an SPL Crimson interface, which I would regard as similar to the Audient performance wise, I usualy have the buffer at 256 or less. Is there a bit accurate driver, and if so are you using it in Reaper?
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  • wave100wave100 Frets: 124
    I also tend to render vst instruments quite early in the process so I can share the files with others who are less well endowed in the plug-in department.
    ;-)
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  • domforrdomforr Frets: 163
    wave100 said:
    My machine is a 2012 i5 MBP with 8 gig ram and an ssd with an SPL Crimson interface, which I would regard as similar to the Audient performance wise, I usualy have the buffer at 256 or less. Is there a bit accurate driver, and if so are you using it in Reaper?
    Hi, it uses Audient ID14 drivers, but I couldn't tell you much more than that? Probably right about the rendering - bit of a pain but would certainly free up some much needed CPU!
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  • funkyfrazfunkyfraz Frets: 72
    I bet its a particular vst you're using. One of the eqs in reaper causes big latency for me! I only use it when everythings been recorded. 
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