DAW Latency Part 2 – Cue Mix using DAW Plugin Processing

We’re still going through the old archives as we tidy things up for anyone who finds this stuff useful. Though not directly related to external hardware tempo-synchronisation it is still a critical aspect of music production with the DAW now the central hub of 99% of all studios – large commercial facilities and bedroom studios alike. Feel free to email if any of this doesn’t ring true as it goes a while back.

The following test case assumes use of an all-in-one prosumer audio interface as a pseudo input mixer/monitoring system rather than an outboard analogue mixing console or an alternative all-analogue signal path for zero latency monitoring.

Steinberg UR44

 

Here is a real-world example of a latency issue when recording anything in real-time against a DAW generated click track or against previously recorded material as a fresh take or as an overdub. Our DAW audio click track guide was routed directly to a Focusrite Saffire 40 headphone output panned hard left. For the purpose of this test we used a direct clone of that same audio click track but routed to a direct analogue output of the Focusrite Saffire 40 then, on to an external analogue mixer (zero additional latency) and back to an input on the Focusrite Saffire 40 for recording back into the DAW. If we take a very common production scenario where a vocalist wants to monitor their performance through various DAW plugin processing (compression/EQ etc) we must route the armed destination DAW audio track (set to input monitor) to the Focusrite Saffire 40 headphone output. We then pan the destination audio track output hard right in the headphones so we can check the offset between the DAW reference playback guide track (what the performer is singing against) and the live processed take they are monitoring in real-time.

What the vocalist (or any other performer) experiences is a fixed delay between their own natural physical feel and phrasing as they sing/play and their ‘foldback’ processed audio coming back to the headphones. Minimal latency values will result in odd phase/tonal shifts. Larger latency values will produce flamming and in the more extreme cases a slap echo. In either case the less enjoyable the recording experience will be and the end result will most likely be an average take.

As a direct comparison this is a quote from John Klett’s paper -¬†Delay in Large Format Digital Music Consoles:-

“Put yourself in the position of vocalist. You are very good and want a cue mix with your voice in it so you can hear yourself as you sound. You need to hear how you sound fairly accurately so can moderate your tonality.

Typically, in an all-analog path, it takes between 2 and 8 microseconds for a signal to make the trip from a microphone, through a preamp, analog dynamics and EQ to a hole in the patch bay that I would then patch to an input on a console.”

http://www.technicalaudio.com/reading/digitalconsoledelay.html

All analogue maximum monitoring latency = 8 microseconds

All digital monitoring latency (DAW@ 48kHz/512 Buffer) = 26 milliseconds

Monitoring delay increased by a factor of 3250 times.

Saffire 40 Cue OutLive UAD ProcessingSF Saffire Cue Out Offset 1

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