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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DAW Latency Numbers

I found these numbers which expand on the previous post – Latency and Feel. I haven’t revisited these for some time but from memory they are accurate. Let me know if anyone thinks they are off the mark. First image shows the waveform offset between the DAW source track audio and the bounced audio after taking a round trip through the audio interface DAC > ADC running at 48kHz with a buffer of 1024 samples and onto a new DAW audio track. A simple application of this would be sending a vocal from a DAW track out to an external effect unit (analogue filter/compressor etc) and then back to an input on the audio interface for recording to a new DAW track. If you are cue/headphone monitoring through the DAW with plugin processing via your audio interface you will have similar issues to contend with. The second image shows the various latency values at a variety of DAW sample rate/buffer settings.
DAW Audio Latency ADAW Latency Numbers A

 

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Roger Linn and the MPC groove myth.

Another one from the archives. Since the release of the MPC-60 in 1988 there have always been rumours about some internal magic feel programmed into the heart of all MPCs thus making them ‘feel’ more musical than playing back real-time MIDI in a DAW environment. The man himself sets the record straight.

Roger Linn MPC and Timinghttp://www.attackmagazine.com/features/interview/roger-linn-swing-groove-magic-mpc-timing/

https://en.wikipedia.org/wiki/Akai_MPC60

MPC-60

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Latency and Feel

Just going back through the archives to extract some useful information for those interested. This relates specifically to digital recording consoles with regards to cue monitoring issues in a digital environment. Track forward to 2016 and the exact same issues still apply to the modern DAW platform with real-time external digital signal processing, active plugin delay compensation, audio interface latency and buffer size, external sequencer tempo-synchronisation, real-time MIDI event delay and external MIDI hardware processing latency.
John Klett Research Paper Excerpt

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

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Sync-Gen II Software Authorization

Authorization Serial Number

http://innerclocksystems.com/New%20ICS%20Downloads.html

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Thank You

Thank U 634

http://innerclocksystems.com/New%20ICS%20Shop%20Online%20Page.html

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Metamorphose

Innerclock Metamorphose

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Roland TR-77 Litmus Test

Roland TR-77 Blog

https://en.wikipedia.org/wiki/Roland_Rhythm_77

> Audio Out – Max. Jitter between Steps: 54 samples (1.10ms)

Source: DC Clock (12PPQ) – Sync-Gen II

> Audio Out – Max. Jitter between Steps: Zero samples (0.00ms)

http://www.innerclocksystems.com/new%20ics%20litmus.html

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Bad Sync

Bad Sync 101

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Eggs and EGs this morning…….

MEGs 634

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