Returning my Presonus Quantum HD2

[TurtleDaze]

I got my Presonus Quantum HD2. I must say that I'm very disappointed. I could not even run it at 48khz and buffer set to 64. My Focusrite 2i2 3rd gen have no issues with that setting. I guess HD2 is a good interface but the drivers are budget. It should not be called a professional interface since it's so very unstable.

I'm looking for an interface that I could run at 48khz and buffer on 32 when I'm recording guitar bass and vocals. I use TH-U Premium for guitar and bass. With focusrite I have done tha but with buffer on 64. RTL is 9,5ms and it feels a bit to high for me.

I know one that would work for me. RME Babyface Pro FS. But in Sweden the price is 800 Euro and thats a bit too much. Is Motu M2, M4, M6 better alternatives compared to Focusrite 2i2 3rd gen?

The only reason I need a faster card is for guitar and bass so perhaps I should invest in a guitar box instead.

I should tell you what cpu I have. Intel Core i5-11400T with 16gb ram.

[Studio Support Gnome]

This might not sound very helpful , but it IS intended to be ….

Sound travels at roughly 1 foot ( 300mm) per millisecond, a Round trip latency of about 9.5ms Is roughly equivalent to standing about 8-9 feet ( 2.5/6ish metres ) from the cab while playing electric guitar.

If you can’t play in time with a drummer from 2.5 metres away you’ve got bigger problems than your interface.

People get too tied up with ultimate low latency specs, and not enough with reality.

In real terms , until your round trip is approaching 100ms I would not entertain the notion of blaming the latency for timing inaccuracy.

Lord help you on a festival stage if it’s that much of a problem.

Forget the spec , focus on the performance.

[Watchmaker]

hmm. I can defnitely hear a 10ms latency when recording, enough to disorient me from the meter. Latency is a form of delay, yes, but when tracking, a latency of 9.5ms will be noticeable by the vast majority of people.

Fender (nee' Presonus) should get on their driver issue pronto. When I bought their stuff ages ago I complained bitterly to their "support" team. Personally, I hope they go out of the interface business, but that's just me I suppose. Get the RME, you'll never have to worry about it again.

[James Perrett]

If you want RME drivers at a cheaper price then go for the Digiface USB with your choice of analogue convertor. You could use a new Behringer ADA8200 or go for something like a used Focusrite Octopre or MOTU 8Pre.

[ajay_m]

I agree. 10ms latency if I am playing piano through a VST definitely is perceptible to me.

If you only need a limited number of analogue inputs and not all the digital inputs on the Presonus, the Komplete Audio 6 (KA6) Mk2 from Native Instruments is a superb audio interface with very low latency. Unfortunately it's now discontinued I think but there are several on eBay, for very reasonable prices (like, £70 or even less). I highly recommend these, mine (which I still have) worked superbly for years and is only sidelined due to the acquisition of the Yamaha DM3S as my main studio interface, mixer and control surface. Otherwise I would still be using it. I could certainly get latency down well below 5ms in Reaper and it supports sample rates right up to 192KHz. Higher sampling rates allow you to get lower latency for a given block size and on modern computers don't really consume significant amounts of CPU.

[TurtleDaze]

Thanks for all the replies. I may be extra sensitive to latency but when playing fast funk beats my timing get affected by RTL over 10ms. 100 ms would be totally impossible.

Me and my bandmates are on the same level. None of us want more than 10 ms and that's not possible to get with HD2 with a stable connection without pops and clicks in my setup. No problem with Focusrite 2i2. I get about 9 ms stable connection and I can record vocals via direct monitoring, 2 guitars (TH-U), bass (TH-U) and keys. All with no extra effects. I raise the buffer from 64 to 1024 when I do the mix.

[g18llo]

I agree. 10ms latency if I am playing piano through a VST definitely is perceptible to me.

I never normally get involved in "how much latency can people hear", but as a piano player, 10ms latency on an AAX/VST is absolutely noticeable; same applies when playing drums with latency at that level too. It's irritating when people try and tell you, what you can and cannot detect, based on what they can or cannot detect.

[g18llo]

I got my Presonus Quantum HD2. I must say that I'm very disappointed. I could not even run it at 48khz and buffer set to 64.

I should tell you what cpu I have. Intel Core i5-11400T with 16gb ram.

I had the HD8, just before Christmas, and that went back after a day's use for exactly the same reason; I couldn't even get it down to 256 without audio issues, and that's on a PC with 128gb ram and an Intel Ultra 9-258k!!!

I bought an Antelope Zen Go Synergy Core (USB, not TB), as a stop-gap whilst I decided what else to use for mixing (I rarely record in my studio, I go elsewhere that's better suited for those jobs), and that's been entirely stable at 64 and 32. As others have mentioned, anything RME will be the most reliable and stable interface you're ever likely to buy.

[BigRedX]

Also because the brain uses both audio and visual cues to determine where sounds are coming from and how far away they are, a sound source 11 feet away does not have the same "feel" as one with 10ms delay piped directly into your ears via a set of headphones.

[ajay_m]

So looking at Reaper with the DM3 at 96KHz I get 3.1ms inbound and 5.1ms outbound with a buffer size of 256. That's a fairly conservative setting that handles demanding VSTs. For just recording/playback, If I set the buffer size to 64 I get 1.1ms inbound and 2.1ms outbound. That is for an 18:18 USB interface. So you should very definitely expect an audio interface to achieve those figures.

[Johnsy]

Re the OP:

9.5ms - or 456 samples @ 48k - is rather a high RTL given a buffer size of 64 samples.

Either they're using very substantial additional buffering (which at least they're declaring), or they've got hideously slow converters (I'm joking here - no modern AD/DAs are that slow!).

The RME UFX (USB), just for comparison, manages a RTL of a fraction over 5ms at 44.1kHz with 64-sample buffers. This is composed of 91 samples on the input side, and 132 on the output side (where additional 'safety' buffering is typically required).

Just out of interest: To those who can hear/feel 10ms - How low does the latency have to be that you don't perceive it? And what latency are you comfortable with?

[TurtleDaze]

To the question how low the latency has to be for me to feel comfortable I can only refer to standing close to my amp. Perhaps maximum 6 feet from my amp or recording my vocals by direct monitoring. But my electric guitar need overdrive and other effects so due to not having good enough gear in my studio I never had a stable connection under 9ms and that's to much for me. I think I would be more than happy with 5 ms RTL

[TurtleDaze]

I got my Presonus Quantum HD2. I must say that I'm very disappointed. I could not even run it at 48khz and buffer set to 64.

I should tell you what cpu I have. Intel Core i5-11400T with 16gb ram.

I had the HD8, just before Christmas, and that went back after a day's use for exactly the same reason; I couldn't even get it down to 256 without audio issues, and that's on a PC with 128gb ram and an Intel Ultra 9-258k!!!

I bought an Antelope Zen Go Synergy Core (USB, not TB), as a stop-gap whilst I decided what else to use for mixing (I rarely record in my studio, I go elsewhere that's better suited for those jobs), and that's been entirely stable at 64 and 32. As others have mentioned, anything RME will be the most reliable and stable interface you're ever likely to buy.

How is it possible that they call it a pro audio interface? I'm glad I'm not alone but it's also too bad that we had to experience this. I'm glad you found a better interface. I think I will go with RME.

[TurtleDaze]

I agree. 10ms latency if I am playing piano through a VST definitely is perceptible to me.

I never normally get involved in "how much latency can people hear", but as a piano player, 10ms latency on an AAX/VST is absolutely noticeable; same applies when playing drums with latency at that level too. It's irritating when people try and tell you, what you can and cannot detect, based on what they can or cannot detect.

I agree totally. I think some musicians have a better feel for timing not saying that you are a better musician because of that. You are just more sensitive to latency as I also seem to be. Not all musicians have the feel for micro timing as I have noticed playing with other musicians. We are all different we just have to be more aware of that. This went way off topic. But one thing is clear to me. I love Studio Pro but I will never again by any audio interface from neither Presonus nor Fender. Trying to decrease latency without considering driver stability is not the way to go.

Dos anyone have any good expeience with Motu?

[Wonks]

Have you considered recording with a small amp in the room but taking a DI feed to record? So you can then pass the recorded DI track through the amp sim and effects.

Of course if you can only record through headphones because of noise issues this won’t work easily without a lot more equipment.

But an amp sim pedal could well work for you as they normally have just 1-3ms delay depending on the unit.

[TurtleDaze]

Have you considered recording with a small amp in the room but taking a DI feed to record? So you can then pass the recorded DI track through the amp sim and effects.

Of course if you can only record through headphones because of noise issues this won’t work easily without a lot more equipment.

But an amp sim pedal could well work for you as they normally have just 1-3ms delay depending on the unit.

I'm considering buying Hotone Ampero II.

[Tomás Mulcahy]

This might not sound very helpful , but it IS intended to be ….

Sound travels at roughly 1 foot ( 300mm) per millisecond, a Round trip latency of about 9.5ms Is roughly equivalent to standing about 8-9 feet ( 2.5/6ish metres ) from the cab while playing electric guitar.

If you can’t play in time with a drummer from 2.5 metres away you’ve got bigger problems than your interface.

People get too tied up with ultimate low latency specs, and not enough with reality. Le notion of blaming the latency for timing inaccuracy.

Lord help you on a festival stage if it’s that much of a problem.

Forget the spec , focus on the performance.

I think the acoustic distance argument is flawed- performance is the key here. The acoustic distance argument falls apart because live scenarios (like jamming with a drummer 2.6m away) add symmetric delay to both sounds, keeping them in sync. DAW monitoring latency hits only your VST instrument’s feedback, creating an action-to-sound mismatch that musicians detect way below 9.5ms. As evidenced by other people's experience above.

Trained ears pick up keystroke-to-tone delays around 100ms on average, in studies on auditory-motor coupling such as this one. Guitarists feel VST disruption at 5-10ms in practice.

[Mick C]

This went way off topic. But one thing is clear to me. I love Studio Pro but I will never again by any audio interface from neither Presonus nor Fender

Apologies in advance if you’ve already tried this, and I’m not trying to teach you how to suck eggs, but I’m surprised at the problem at low buffer sizes when you’re using Presonus hardware and software together.

Studio Pro has a low latency mode where you set different buffers for recording and playback, and there’s also the dropout protection setting too. This mode shows up as a small green letter ‘Z’ in the buffer settings page, when engaged.

I’m not in front of my computer to describe it more accurately but if you’ve not been using it then check for Gregor on YouTube.

https://youtube.com/shorts/Wvov9Nn4He4? ... jHsrCAsRPi
https://youtu.be/jfSIKdL2DYA?si=61r2KVl5h-zCFjTZ

[TurtleDaze]

This went way off topic. But one thing is clear to me. I love Studio Pro but I will never again by any audio interface from neither Presonus nor Fender

Apologies in advance if you’ve already tried this, and I’m not trying to teach you how to suck eggs, but I’m surprised at the problem at low buffer sizes when you’re using Presonus hardware and software together.

Studio Pro has a low latency mode where you set different buffers for recording and playback, and there’s also the dropout protection setting too. This mode shows up as a small green letter ‘Z’ in the buffer settings page, when engaged.

I’m not in front of my computer to describe it more accurately but if you’ve not been using it then check for Gregor on YouTube.

https://youtube.com/shorts/Wvov9Nn4He4? ... jHsrCAsRPi
https://youtu.be/jfSIKdL2DYA?si=61r2KVl5h-zCFjTZ

Every tip is appreciated. So no worry. When you use HD2 there is an even better mode with a blue D and I tried that. It was working ok but the first thing I want is for the interface to be able to handle at least EZdrummer at 48khz an buffer 64 without blue Z or blue D. When all I hear is crackling and popping things are very bad. I compare it to my Focusrite 2i2 3rd gen.

HD2 is on it's way back to the seller and I have ordered Hotone Ampero II.

[TurtleDaze]

This went way off topic. But one thing is clear to me. I love Studio Pro but I will never again by any audio interface from neither Presonus nor Fender

Apologies in advance if you’ve already tried this, and I’m not trying to teach you how to suck eggs, but I’m surprised at the problem at low buffer sizes when you’re using Presonus hardware and software together.

Studio Pro has a low latency mode where you set different buffers for recording and playback, and there’s also the dropout protection setting too. This mode shows up as a small green letter ‘Z’ in the buffer settings page, when engaged.

I’m not in front of my computer to describe it more accurately but if you’ve not been using it then check for Gregor on YouTube.

https://youtube.com/shorts/Wvov9Nn4He4? ... jHsrCAsRPi
https://youtu.be/jfSIKdL2DYA?si=61r2KVl5h-zCFjTZ

Every tip is appreciated. So no worry. When you use HD2 there is an even better mode with a blue D and I tried that. It was working ok but the first thing I want is for the interface to be able to handle at least EZdrummer at 48khz and buffer 64 without green Z or blue D. When all I hear is crackling and popping things are very bad. I compare it to my Focusrite 2i2 3rd gen.

HD2 is on it's way back to the seller and I have ordered Hotone Ampero II.

[Studio Support Gnome]

I think there’s a distinct lack of grasp of my comment by many .

Let me put this another way .

In a large ensemble, be it orchestral or choral , you do NOT listen to each other for timing reference , you watch the conductor . ( or metronome , beat counter , etc etc )

In a stage arrangement that can be 80 -100 feet across and 50 feet deep , or more if combining large choir with full orchestra, you can end up with much more significant audio delays across the ensemble , and when you add reaction time to performance, you end up with an utter mess .

Think my least favourite example was recording the 200 strong choir of the parliament with an orchestra for a Christmas carol service . They’re not bad singers but you could tell segments of them were listening rather than watching , the sync between orchestra and choir was poor , and even within the choir was chaotic .

The point here is that there is an acquired skill to playing in time with different auditory offset delays and it’s not about latency performance .

I have routinely played live with a usb ( Focusrite forte) interface on a 14 year old MacBook Pro ( last generation 17” MBP) at 48 kHz and 128 sample buffer . Along with other live musicians , and to backing tracks . It’s not about the latency . It’s about the ability to play in time and compensate for perception delays , I’ve also played on large stages standing 50 feet away from my amp and using local monitoring , or similarly using wireless in ears .,, at no point did the variation in arrival time mean I could not play the part . Given that I occasionally bang out multi octave runs and sweep arpeggios in not a lot of time , I can be delivering multiple notes in tbe time it takes audio to cross the stage . In an orchestra or choir , you watch the conductor , in a band you watch the drummer , in a studio you watch the bar/beat counter . ( and yes screens also have latency .) . The key here is that you watch , not listen . This removes both system latency and distance from the timing equation , .,

That’s why I said focus on the performance not latency .

[James Perrett]

As a drummer I find it massively easier to play tightly if the bass amp is next to me or if the bass is clear in the drum monitor sitting next to my left foot. I also find that the difference between an OK player and a great player is with timing of just a few mS. These small differences matter to me.

While coping with latency may be a skill useful to some, why make life more difficult by adding latency when there are alternative products with lower latency?

[Tomás Mulcahy]

I also said focus on performance not latency. My issue is with dismissing latency with the old speed of sound v distance argument, when a musician clearly states it is a performance issue for them. If the computer rig responds in the same way as your instrument does in real life scenarios, then it's fine. But often it does not, because of latency. A singer cannot tolerate any latency in their monitoring, because it does not occur in real life, hence direct monitoring being the norm in most interfaces. For example a piano does not have latency, niether does a mechanical pipe organ. But a console organ does, and players learn to compensate, because the other option can often be to sit further away from the conductor with a mirror, which brings its own problems. Obviously there are various tried a tested solutions to that problem, player preferences and a number of variables depending on the venue, which we don't really need to go into- I think the point should be clear.

All I am saying is, the speed of sound/ distance argument is flawed, since it is often given out of context of the actual scenario causing the problem for the audio interface user.

[TurtleDaze]

I also said focus on performance not latency. My issue is with dismissing latency with the old speed of sound v distance argument, when a musician clearly states it is a performance issue for them. If the computer rig responds in the same way as your instrument does in real life scenarios, then it's fine. But often it does not, because of latency. A singer cannot tolerate any latency in their monitoring, because it does not occur in real life, hence direct monitoring being the norm in most interfaces. For example a piano does not have latency, niether does a mechanical pipe organ. But a console organ does, and players learn to compensate, because the other option can often be to sit further away from the conductor with a mirror, which brings its own problems. Obviously there are various tried a tested solutions to that problem, player preferences and a number of variables depending on the venue, which we don't really need to go into- I think the point should be clear.

All I am saying is, the speed of sound/ distance argument is flawed, since it is often given out of context of the actual scenario causing the problem for the audio interface user.

I have played in many different types of constellations and you can never compare latency in a live situation to studio. You adapt to other musicians in a live situation where all intruments have different latency but you hear you own instrument with almost zero latency. Lets try this. If the only way you could hear your own instrument was in headphones with 100ms latency. Every other intrument was in absolute sync. Are you serious when you say that you don't care about latency and you only focus on your performance? If that is so I can only congratulate you.

[TurtleDaze]

As a drummer I find it massively easier to play tightly if the bass amp is next to me or if the bass is clear in the drum monitor sitting next to my left foot. I also find that the difference between an OK player and a great player is with timing of just a few mS. These small differences matter to me.

While coping with latency may be a skill useful to some, why make life more difficult by adding latency when there are alternative products with lower latency?

I totally agree. We all have different abilities to feel timing. I have learned to adapt to those that have less feel for timing but in som instances timing is crucial. Not saying that it's the only attribute for a good musician. We all have more or less than others. Music should not be a competition as long as we love what we do.