Zero Latency Devices / Latency List
Technically yes, BUT (and this is HUGE) when that processing latency is less than the sample rate, there will be NO latency IN THE AUDIO PATH - this is what is important for audio processing.kitekrazy wrote:Zero latency is a marketing phrase. There will always be latency with a computer.
Remember that computers process data at a rate MANY times faster than the audio sample rate, measured in GIGAHERTZ rather than kilohertz, which means they can make MANY calculations long before the audio path needs to see the next sample.
Hopefully I've explained this correctly - if not someone please correct me!
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Selig Audio, LLC
Actually its more of a data throughput issue than a computing issue, a USB bus can only transfer data so fast, memory can only copy data so fast.. There is DSP systems that actually work sample by sample or even bit by bit, without any latency, so its not really inherent with "digital" (if the OPs use of "computer" implied this). That said, audio in itself is tied to timeselig wrote:Technically yes, BUT (and this is HUGE) when that processing latency is less than the sample rate, there will be NO latency IN THE AUDIO PATH - this is what is important for audio processing.
Remember that computers process data at a rate MANY times faster than the audio sample rate, measured in GIGAHERTZ rather than kilohertz, which means they can make MANY calculations long before the audio path needs to see the next sample.
Hopefully I've explained this correctly - if not someone please correct me!

But obviously you're right with the core statement, the "problem" is ADDITIONAL latency, above the buffer size of the system and yes, CPUs are INCREDIBLY fast in terms of computing alone.
Possibly wrongly, I was assuming audio already in the digital domain - obviously getting analog audio in and out of a digital system DOES induce some amount of latency. But my point was related to how an internal process such as a plugin can be "zero latency" within the system.normen wrote:Actually its more of a data throughput issue than a computing issue, a USB bus can only transfer data so fast, memory can only copy data so fast.. There is DSP systems that actually work sample by sample or even bit by bit, without any latency, so its not really inherent with "digital" (if the OPs use of "computer" implied this). That said, audio in itself is tied to timeselig wrote:Technically yes, BUT (and this is HUGE) when that processing latency is less than the sample rate, there will be NO latency IN THE AUDIO PATH - this is what is important for audio processing.
Remember that computers process data at a rate MANY times faster than the audio sample rate, measured in GIGAHERTZ rather than kilohertz, which means they can make MANY calculations long before the audio path needs to see the next sample.
Hopefully I've explained this correctly - if not someone please correct me!
But obviously you're right with the core statement, the "problem" is ADDITIONAL latency, above the buffer size of the system and yes, CPUs are INCREDIBLY fast in terms of computing alone.

Selig Audio, LLC
OK - Hi everyone
Can someone please explain how and when to use this,, to a complete newbie! - In practical terms: say i used drums with kong put a synth line down with the stock plugs (should still all be zero lat) THEN I decide to use bass guitar or lead guitar using kussa rack extensions:
Once you know the latency time in milliseconds what do you do? do you go to preferences and change something there? so confused about this. Im luckily got a UAD Apollo 8p so vox coming in should be zero latency! and my studio monitors are zero latency as I'm mixing through an external analog desk. Just confused because I've never adjusted anything in terms of latency settings (wherever that is)
Can someone please explain how and when to use this,, to a complete newbie! - In practical terms: say i used drums with kong put a synth line down with the stock plugs (should still all be zero lat) THEN I decide to use bass guitar or lead guitar using kussa rack extensions:
Once you know the latency time in milliseconds what do you do? do you go to preferences and change something there? so confused about this. Im luckily got a UAD Apollo 8p so vox coming in should be zero latency! and my studio monitors are zero latency as I'm mixing through an external analog desk. Just confused because I've never adjusted anything in terms of latency settings (wherever that is)
@Chrisdude: Unless you do use parallel processing (e.g. via parallel mix channels) the minimal latency differences don't really do anything bad, so don't have to care about this. But when you do use an RE with latency on a parallel mixer channel and you notice a comb filter like sound, then you've got a problem, where you should compensate for the latency introduced by the RE (unless you actually like the sound of the comb filter fx).
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I agree with jam-s. It depends on what kind of Music style we are in and what we want to achieve exactly. The first time when a latency problem bothered me was with Polar while recording a melody. Knowing this i figured i could use a) Normen's device or b) just play the sequence dry and then add the particular effect after the recording session. The whole process with Music creation requires flexibility and workaround anyway.
Even when delay compensation is on, sometimes the plugin still creates a delay due to the nature of the effect.
Thats why TS-1 is great for realigning
Thats why TS-1 is great for realigning
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Is this still a big issue as Reason has PDC now, I don't think I have issues with it.
I guess if you are doing live audio or weird routing it can still be an issue though.
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Wow! This was a lot of time and effort spent to help this community...THANK YOU SO MUCH!!!!thala wrote: ↑21 Jun 2015i got the VMG Sample Delay to Measure some things. after working a while with it, it seems that everybody who is interested in Latencys of a Device should get this, to test the devices everytime u need a latency compensation with that actual settings of that device.
but here is a small overview of the Devices i bought + that ones in "trial".
THIS LIST IS ONLY A FIRST TRY. WOULD BE NICE IF SOMEONE ELSE COULD CONFIRM THESE LIST.
maybe i did something wrong?
Sample Rate 44,1kHz
The Effect Section:
Pulverizer 1 Sample
The Echo 0 Sample
Alligator 0 Sample
Scream4 0-32 Samples (depends on settings)
Vocoder B512 (variable depending on Amount of Bands, first Sample-test up to 50+ Samples, Second Test down to 1 Sample... wierd)
RV7000mk1 0 Samples on Original Signal
Neptune 325 Samples (Live Mode, no "Low Freq") up to 1842 Samples (No Live Mode, Low Freq on)
Line6 Guitar Amp 71 Samples on init
Line6 Bass Amp 30 Samples on init
MClass EQ 0 Samples
MClass Comp 0 Samples
MClass Maximizer 0 Samples without lookahead
MClass Stereo Imager 0 Samples
RV7 0 Samples
DDL1 0 Samples
D11 Foldback 0 Samples
ECF 0-35 Samples (depends on Mode and Cutoff Setting)
CF-101 Chorus/Flanger 0 Samples (Send mode off)
PH90 Phaser 0 Samples
UN16 0 Samples
Comp01 0 Samples
PEQ-2 0 Samples
Audiomatic 0-xxx (depends on setting) (Preset "Cracked" 2-35 on wet)
Polar 1154-5129 Samples
Rotor 0 Samples on Dry
Softube Amp 26+ Samples
Softube Bass Amp 4-xx Samples (Amp bypass has higher latency than with amp...)
Synchronous 0 Samples
Blamsoft DC-9 3 Samples
Blamsoft Resampler 0 Samples on Dry
DCam Env Shaper 3-4 Samples on Dry
Etch Red 3 Samples on init
G-Clip 0 Samples without oversampling (with OS 2-3 Samples)
Ozone 262-3004 Samples
Chenille 0 Samples
Titus 0 Samples
Black Knight 0 on Parametric Mode, 1 Sample on Grphic Mode
Neutron 0 Samples on Dry
Futzbox Filters, Distortion and Noize 0 Samples. Gate 11 Samples, Sim depends on Sim
Carve 0 Samples
Glitch 64 Samples
Repeat Looper 0 Samples
Buffre 0 Samples
3Plex 0 Samples
RE 200 Bass Enhancer 0 Samples
RE 202 Exciter 0 Samples
Selig Gain 0 Samples
Selig Leveler 0 Samples
Saturation Knob 3-4 Samples
Spring Reverb 0 Samples
Tube Delay 7 Samples on Dry
Dynamite 3 Samples
Bitspeak not possible to measure
Echobode 0 Samples on Dry
GSX 66 Samples with Limiter, 0 Samples without Limiter
SB Filter Pattern 0 Samples on Dry
SB Slice Arranger 0 Samples on Dry
Synapse Fat Space 0 Samples
Synapse GQ-7 0 Samples , even with automatic Gain on!
Synapse 90 Vintage Phaser 0 Samples
Mr. Overdrive 0 Samples
T2 Phaser 0 Samples
UHBIK-A 0 Samples on Dry
G8 Gate 0 Samples with 0ms Lookahead and "flip" on, flip off =25 Samples
Yoko Band Splitter 1-8 Samples (depends on Slope)
Hilfsgeräte:
Combinator 0 Samples
Audio Splitter 0 Samples
Mixer 14:2 0 Samples
Line Mixer 6:2 0 Samples
Polymodular Audio Splitter 0 Samples
Polymodular Audio Merger 0 Samples
Morfin Crossfader 0 Samples
Executioner DJ Mixer 0 Samples
Skope 0 Samples
Skope M4 0 Samples
VMG Sample Delay 0 Samples![]()
A/B Switch 0 Samples
RE181 M/S Audio Converter not possible to measure
CMD:Education 0 Samples
Scope Jr. 0 Samples
Instruments: (using audio in)
Kong:
Kompressor 0 Samples
Filter LP=1+ Samples / BP and Highpass=0+ Samples (depends on cutoff)
Overdrive/Resonator 0-30 Samples
Parametric EQ 0 Samples
Rattler 0 Samples
Ringmodulator 0 Samples
DrumRoom Reverb 0 Samples
TapeEcho 0 Samples
Transient Shaper 0 Samples
Thor Audio in > Audio out (via modmatrix) 0 Samples
Malström 0 Samples
Parsec not possible (used audio in and gave a gate on via matrix...)
Punch BD 0 Samples
Antidote not possible to measure ... tested on left chanel, even with all effects on bypass or everything dry (setup was correctly, retested with audio)
the instrument section is a hell of a job... canceled
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Here’s a question: since Reason has delay compensation now, why do we “care” about latency? I’m not trolling, this is a serious question. The most annoying part of Reason for me isn’t plug-in latency at all, but the inconsistent and incorrect timing of the EMI.
I’m going to start a thread about this issue, but wanted to thank you for the effort and ask why latency is important with delay compensation enabled in Reason.
DAW: Reason 12
SAMPLERS: Akai MPC 2000, E-mu SP1200, E-Mu e5000Ultra, Ensoniq EPS 16+, Akai S950, Maschine
SYNTHS: Mostly classic Polysynths and more modern Monosynths. All are mostly food for my samplers!
www.soundcloud.com/jimmyklane
SAMPLERS: Akai MPC 2000, E-mu SP1200, E-Mu e5000Ultra, Ensoniq EPS 16+, Akai S950, Maschine
SYNTHS: Mostly classic Polysynths and more modern Monosynths. All are mostly food for my samplers!
www.soundcloud.com/jimmyklane
I guess because a) It's only compensated between mixer channels, not in combinators or anywhere else and b) many people use Reason to play instruments livejimmyklane wrote: ↑17 Jun 2018Here’s a question: since Reason has delay compensation now, why do we “care” about latency?
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Ok, that’s reasonable. Thanks for answering me!normen wrote: ↑17 Jun 2018I guess because a) It's only compensated between mixer channels, not in combinators or anywhere else and b) many people use Reason to play instruments livejimmyklane wrote: ↑17 Jun 2018Here’s a question: since Reason has delay compensation now, why do we “care” about latency?
DAW: Reason 12
SAMPLERS: Akai MPC 2000, E-mu SP1200, E-Mu e5000Ultra, Ensoniq EPS 16+, Akai S950, Maschine
SYNTHS: Mostly classic Polysynths and more modern Monosynths. All are mostly food for my samplers!
www.soundcloud.com/jimmyklane
SAMPLERS: Akai MPC 2000, E-mu SP1200, E-Mu e5000Ultra, Ensoniq EPS 16+, Akai S950, Maschine
SYNTHS: Mostly classic Polysynths and more modern Monosynths. All are mostly food for my samplers!
www.soundcloud.com/jimmyklane
Can you please add this :
Selig Leveler (look ahead activated) : 88 ms
Selig DeEsser (look ahead activated) 617 ms
I hope reason studios update a fix for this because it is causing me real headaches as is not even registered in Automatic delay compensation.
Selig Leveler (look ahead activated) : 88 ms
Selig DeEsser (look ahead activated) 617 ms
I hope reason studios update a fix for this because it is causing me real headaches as is not even registered in Automatic delay compensation.
Last edited by Josdams on 29 Nov 2019, edited 1 time in total.
my last project ! https://www.youtube.com/watch?v=Lk-w43nTgws
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On another note, Glitch RE produces phasing, even when bypassed, when applied to a parallel channel. Doesn't report latency. 
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Another observations
Reason Needs to fix:
Reason Maximizer when 4ms look ahead is activated : 177 ms /current auto compensation : 176 is Wrong
Izope Ozone Maximizer when IRCII mode is activated : 2992 ms / current auto compensation 3014 is Wrong
Reason Needs to fix:
Reason Maximizer when 4ms look ahead is activated : 177 ms /current auto compensation : 176 is Wrong
Izope Ozone Maximizer when IRCII mode is activated : 2992 ms / current auto compensation 3014 is Wrong
my last project ! https://www.youtube.com/watch?v=Lk-w43nTgws
It depends on what you are doing and your interface/soundcard.
Generally 256 is ok for low latency softsynth playing and for most cards somewhat on the (high side but if it aint bothering you, its not a problem).
If you are doing audio recording and want to hear plugins in realtime, this is what stresses the system the most and you would probably want as low settings you can get.
generally If you can get roundtrip under 10 ms, you are great.
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