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TAFKAT
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Loc: Australia
Audio Interface - Low Latency Performance Data Base :
      #918351 - 06/06/11 04:10 AM
Hey All,

I am sure all of us have seen posts over the years where end users have purchased a new audio interface and have experienced varying degrees of performance issues. A lot of the qualms have been due to the low latency performance, or to be exact, lack of , so I thought it would be a good idea to place the spotlight on that specific aspect of the current crop of audio interfaces.

By testing the interfaces head to head and reporting warts and all, I hope to place some of these developers on notice that jamming as many shiny baubles onto an interface , but neglecting the crucial aspect of the actual driver performance , isn't going to swing for all of us.

I have been working on this for about 15 months odd so the preliminary results were actually from earlier in 2010 , but are still valid and are a good starting point as I add further to the list over the coming month/years.

Here are some initial results for both Desktop and Notebook .

Desktop System Detail:
Intel i7 920 Quadcore/ 2.66 GHZ/
Intel X58 / 6 GB DDR3-PC12800.

Notebook System Detail:
Intel i7 620 Dualcore / 2.66 GHZ/
Intel HM55 / 4 GB DDR3-PC10600.

O.S Detail:
Windows 7 x64 Pro





All of the interfaces listed except for the baseline RME HDSPe units are FW and its quite obvious from the results that there are large variables in LLP- Low Latency Performance depending on the efficiency of the respective ASIO drivers / FW controllers being employed on the respective interfaces.

I have posted a summary of the initial testing at the DAWbench site Here : http://www.dawbench.com/audio-int-lowlatency.htm

I have a second round of testing on the way which will include some of the newer AVID interfaces at both ends of the spectrum , Mbox Pro 3 and also one that I have had quite a few requests to put the thru its paces, the PTHD Native PCIe card.

Stay Tuned

Peace

Vin Curigliano
AAVIM Technology

--------------------
AAVIM Technology
DAWbench.com


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Pete Kaine
Scan Computers


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918375 - 06/06/11 08:31 AM
Whilst I knew Maudio were doing a good job in keeping up with RME these days, I'm quite shocked to see Focusrite trailing by those figures. I'm not however suprised to see the Focusrite and the Presonus coming out the same through as they are both built around the same controller chipset... also nice to see that my contempt for its abilitys is well founded!

--------------------
ScanProAudio & 3XS Audio Systems
ScanProAudio Blog


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redleicester
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918406 - 06/06/11 09:33 AM
Yet another good reason to stick with RME. Low latency and rock solid driver sets for a decade now. Great stuff.

--------------------
Ars longa, vita brevis, occasio praeceps, experimentum periculosum, iudicium difficile.


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918446 - 06/06/11 11:37 AM
Hi Vin,

I bet you’ll end up taking a lot of flak for these measurements (although your armour-plated shell must be largely impervious by now), but I have to say that my own recent measurements of low latency performance by audio interface manufacturer do largely tie in with these

RME have slipped a little on occasions with driver performance (as sadly have Lynx Audio, who used to be up there at the top with them), but it’s good to M Audio come back with such comparatively good results


Martin

--------------------
YewTreeMagic


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Parco



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918465 - 06/06/11 12:41 PM
So how about the echo audio?

--------------------
Cubase 5.1 + Foobar2000 -> ASIO + MMCSS + Windows7 -> SIIG TI chips 1394 PCIe -> Echo AudioFire 4 -> HH MX250 + Wharf Delta70
A native USA style contemporary R&B artist in HongKong


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918492 - 06/06/11 02:00 PM
Hey Guys,

I think its high time we collectively hold these audio developers accountable for the delivered driver performance , and its just becoming more and more of a bloody minefield lately as more and more developers simply tack on these generic OEM FW controllers without any regard of the overall performance. The recent spate of mixers/come audio interfaces are prime examples , and there is one currently extremely popular "Digital Live " mixer that has the worst drivers I have encountered in all my years, even worse than Dice II , and they market it as perfect for studio use, I almost fell flat on my arse laughing qualifying it a while back. I'll hold my cards on that one for now...:-)

Re The Dice II powered interfaces , as bad as they perform on DAWbench DSP, on DAWbench VI its even more pronounced where I recently tested a Focusrite unit which collapsed under that test to the point of it being practically unusable for anyone working with sample based VI's at the preferred working latencies. Those ongoing results can be read at the DAWbench forum thread until I finalise the second report.

Thread is Here

Out of interest , in the recent review in SOS of an "unlocked" Sandybridge system that was shipped with a - quote "well‑known Firewire audio interface" to Martin for review, that then had to subsequently be replaced to get a respectable test result, it would be interesting to get some clarification on what that interface was.

Now I know that is walking a very sensitive line for Martin seeing the name of the interface was withheld , but I am not asking Martin, I'm asking the DAW builder who supplied the unit to disclose what he supplied for the review, and I feel he should be extremely grateful that Martin went the extra yard instead of just reviewing as supplied :-)

@ Parco,

Interestingly with the ECHO, I just completed a config for a client with his existing Layla 24 under Win7 x64, and it performed reasonably well against the current crop of PCIe cards, but it does have the trick of using double buffers on playback which becomes a juggle with Round Trip Latency using Guitar Amp sims for example. You can wind it down to the 032/064 setting when tracking the guitars thru the sim, and it felt quite comfortable. Drivers are very solid overall.

Peace

V:


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Parco



Joined: 25/02/10
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918656 - 07/06/11 04:00 AM
And so how about the Apogee and Motu?

--------------------
Cubase 5.1 + Foobar2000 -> ASIO + MMCSS + Windows7 -> SIIG TI chips 1394 PCIe -> Echo AudioFire 4 -> HH MX250 + Wharf Delta70
A native USA style contemporary R&B artist in HongKong


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CaptainChoptastic



Joined: 09/12/09
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918740 - 07/06/11 11:29 AM
Hmmm, somewhat disturbing news, given that I have a Focusrite Saffire Pro 24 DSP (kind of rolls off the tongue, don't it?). It doesn't mention the FW chipset on the spec sheet, but given the above and the reference to JetPLL, it's probably all but a given.

Vin: as you were running this under Win 7 x64, can you tell us if you were using the "regular" Win 7 FW drivers, or the legacy ones that Focusrite mention on their site as potentially having better performance under x64?


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: CaptainChoptastic]
      #918906 - 07/06/11 08:53 PM
Quote CaptainChoptastic:


Vin: as you were running this under Win 7 x64, can you tell us if you were using the "regular" Win 7 FW drivers, or the legacy ones that Focusrite mention on their site as potentially having better performance under x64?




Hey ,

The Focusrite and most of the Presonus units are definitely using the Dice II chipset , no question whats so ever.

Test system is actually running under the legacy FW at the moment due to the large number of interfaces that are constantly on and off the rig. In testing I found that even units like the RME's that run fine under Native FW drivers( TI in this case ), do have a small performance advantage under legacy as well at the lowest latencies 48 , but by 064 it levels out. In short there is no disadvantage running the legacy drivers, so I leave them on as default , but also test and compare with the native as new drivers come out.

To add insult to injury, I can't get the Dice II interfaces to be even recognised on my new audio notebooks at all ,due to the inability of getting a TI controller option , and this will become even more wide spread . I suspect Apple have also dropped TI on the new line and lack of Express slots becoming the norm, this is sounding loud bells for these interfaces.

Lets hope the devs don't play Ostrich.

I digress..

Peace

V:


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Dishpan



Joined: 01/09/04
Posts: 813
Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918910 - 07/06/11 09:17 PM
Can I ask: are you including information on the safety (and other) buffers used, as well as the buffer size set in the application. There is quite a difference here between manufacturers and by way of a (possible exaggerated example), a system with 40 sample safety buffer and 32 sample latency (set in DAW) would have the same overall latency as a system with an 8 sample safety buffer and 64 sample latency set in DAW, yet on the charts you'd show them at different latencies making the first system look far more efficient than the second.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Dishpan]
      #918933 - 08/06/11 12:31 AM
Quote Dishpan:

Can I ask: are you including information on the safety (and other) buffers used, as well as the buffer size set in the application. There is quite a difference here between manufacturers and by way of a (possible exaggerated example), a system with 40 sample safety buffer and 32 sample latency (set in DAW) would have the same overall latency as a system with an 8 sample safety buffer and 64 sample latency set in DAW, yet on the charts you'd show them at different latencies making the first system look far more efficient than the second.




I hear ya.. :-)

Yes I am including all that info ,I make a note of that in the initial article and all of the actual I/O latency figures for the respective interfaces are listed with the results on the dedicated thread at DAWbench.

I'll create some charts to show the total I/O and Round Trip figures for each respective interface as I progress.

It needs to be noted that the output buffer value is the one that takes precedent on these tests , so the devs who use the double buffer trick will be definitely be highlighted, none of the listed used that tweak and although output value differed, there was not huge discrepancies.

Thats not to say the Input value and or RT - Round Trip - is any less important, because its those values that takes precedent when playing the Virtual Instruments live and Guitar Amp Sims for example.

Peace

V:


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robinv



Joined: 31/08/04
Posts: 779
Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918980 - 08/06/11 10:38 AM
Quote TAFKAT:

Hey Guys,

I think its high time we collectively hold these audio developers accountable for the delivered driver performance , and its just becoming more and more of a bloody minefield lately as more and more developers simply tack on these generic OEM FW controllers without any regard of the overall performance.




Hi Vin,
I'm not convinced that developers disregard overall performance quite like that. I imagine they work very hard trying to get their interfaces to do the job. It also depends on who you talk to as to where they believe the challenges lie.

You should bear in mind that there are many many happy users out there - the best feedback a manufacturer gets is in sales and returns. Obviously clever marketing and features can get the sales but if the performance was "unworkable" then they would become quickly aware of that you would think. You look at performance from a bit of a narrow perspective - from your position of scrutinising latencies of 64 samples and such (and you've been doing it for years) i can totally understand your concerns. My experience in the industry is a bit different and i find that most people i encounter are using buffer sizes of 256 or 512 samples very happily - this is where your tests show a bit more of an even level of performance. I know that there are people for whom this is completely unacceptable and unusable but i would suggest that there are many many more for whom it's completely fine and "real-time" etc etc.

I did my own tests on some interfaces last year when i was looking for a new laptop and i compared an Edirol FA66, Saffire Pro 24 and Presonus Firestudio and there were marked differences between the Firestudio and Saffire so i'm not convinced that they are the same, at least these models weren't. The Edirol beat the other two by a huge margin and it has very unsophisticated driver set with bugger all control panel ( http://www.pc-music.com/content/lifting-lid-audio-laptops-part-2 ). I was shocked at how poorly the Firestudio performed in these tests. However I have run them on desktop machines and other laptops without problems at the usual 256/512 level. The Saffire did all right i thought

I guess it comes down to what you believe is "fit for purpose". It might be more interesting to find out what sort of buffer sizes people actually use - i don't imagine many venture below 256 samples (but i could be totally wrong but i'm going on the complete lack of having to do tech support for latency related issues). You'll note that the only trouble Martin had in his review is when running your low latency tests - any other test that both he and i did before sending it out using what i'd suggest are more common buffer sizes, it performed fine.

I think comparing audio interfaces like this is a really great idea and will generate a lot of useful data that will help people chose the right interface for them - good on ya mate. Similarly the Cubase/Pro Tools comparison is a real eye-opener. But i also don't believe this level of low latency is the whole story for everyone. Not sure you're going to start a revolution but any improvement to driver performance is certainly a good thing

So, yeah, you know, different perspectives and stuff

--------------------
Molten Music Technology - Computers for doing music on


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onesecondglance



Joined: 02/01/08
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #918984 - 08/06/11 10:49 AM
this stuff is fascinating as always, Vin. i would be very interested to see how higher end FW stuff like Prism and Apogee stack up in this dept.

--------------------
hourglass | random thoughts | doubledotdash!? collective


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Spangler



Joined: 21/01/05
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #918990 - 08/06/11 11:19 AM
Fine for mixing, but for playing virtual instruments or realtime FX, I'm one of those who need it a lot tighter. Playing eDrums through Superior, anything over 64 samples is no good on my interface (of course midi performance also comes into play in that scenario).

Quote robinv:



My experience in the industry is a bit different and i find that most people i encounter are using buffer sizes of 256 or 512 samples very happily - this is where your tests show a bit more of an even level of performance. I know that there are people for whom this is completely unacceptable and unusable but i would suggest that there are many many more for whom it's completely fine and "real-time" etc etc.





--------------------
clicky

Edited by Spangler (08/06/11 11:23 AM)


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919113 - 08/06/11 08:28 PM
It looks as if I ought to set up a thread/poll so SOS Forum users can tell us what buffer size they currently use, and for what purpose (live monitoring, playing softsynths, or mixing/mastering), so we can all get a better handle on who wants what


Martin

--------------------
YewTreeMagic


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #919120 - 08/06/11 09:39 PM
Quote robinv:



You should bear in mind that there are many many happy users out there - the best feedback a manufacturer gets is in sales and returns. Obviously clever marketing and features can get the sales but if the performance was "unworkable" then they would become quickly aware of that you would think.




Horses for courses, we may have a different client demographic.


Quote:

You look at performance from a bit of a narrow perspective - from your position of scrutinising latencies of 64 samples and such (and you've been doing it for years) i can totally understand your concerns. My experience in the industry is a bit different and i find that most people i encounter are using buffer sizes of 256 or 512 samples very happily - this is where your tests show a bit more of an even level of performance. I know that there are people for whom this is completely unacceptable and unusable but i would suggest that there are many many more for whom it's completely fine and "real-time" etc etc.




My work over the years in regards to focusing on native low latencies performance was due to direct requests and demands from my client base , and the volumes of traffic across multiple forums that I was active on. You may feel its narrow, but again we have a different demographic.

For those not requiring the lower latencies then the interfaces that are being highlighted as performing badly at those latencies are a non event , but I disagree that the majority of clients working latencies are 256/512, anyone who can actually play would not be comfortable using a VI at those latencies, I know I wouldn't. Also current systems with good audio interfaces can easily be set and forgot at those lower latencies and not have to do the buffer juggle if they decided to go back and do an overdub when mixing , for example. Also many composers process while they compose so again the lower latencies are extremely important.

Quote:

I did my own tests on some interfaces last year when i was looking for a new laptop and i compared an Edirol FA66, Saffire Pro 24 and Presonus Firestudio and there were marked differences between the Firestudio and Saffire so i'm not convinced that they are the same, at least these models weren't.




Its a constantly moving target, Presonus use multiple OEM FW controllers across the range, its why they have different drivers and not one universal one. Also I did have some variance between the 2 on my notebook testing , so there is some wiggle room in how they tweak the drivers perhaps.

Quote:

You'll note that the only trouble Martin had in his review is when running your low latency tests - any other test that both he and i did before sending it out using what i'd suggest are more common buffer sizes, it performed fine.




Be very grateful he wasn't using DAWbench VI as a reference test, as the "unnamed" interface would have fallen in a heap at the working latencies anyone would use for live playing of VI's , right up to 256 samples. 512 was O.K in the test without convolution verbs, it stalled with, but seriously Mate , 512 samples is purely for mixing only. You can't play a VI comfortably at that level, you can't use a Guitar Amp Sim, you can't play live MIDI drums into a Drum Sampler , etc, etc.

In short it simply negates a huge potential of the advantages we enjoy using these current uber fast systems.

Quote:

I think comparing audio interfaces like this is a really great idea and will generate a lot of useful data that will help people chose the right interface for them - good on ya mate.




That's the aim.

Quote:

Not sure you're going to start a revolution but any improvement to driver performance is certainly a good thing




I'll call and raise ya... :-)

The past work has already resulted in better drivers and multiprocessor performance on some DAW hosts , revolution may be too lofty, but this will grow legs , how hairy will be up to the audio hardware developers.

Lets see where the dust settles.

Yes I have my helmet and armour on.. :-)

P.S: you still haven't disclosed the interface used in the review, if its such a non event, then it shouldn't be a problem.. :-)

Peace

V:


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mhaigh



Joined: 14/11/07
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #919131 - 08/06/11 10:32 PM
Quote robinv:


It might be more interesting to find out what sort of buffer sizes people actually use - i don't imagine many venture below 256 samples (but i could be totally wrong but i'm going on the complete lack of having to do tech support for latency related issues).




hi Robin
i run an echo audiofire 2 and find their drivers totally stable. As Vin says, Echo use double safety buffers, but I ALWAYS run at 128 samples, whether tracking or mixing. I can go to 64 samples but i find this makes the system a little unstable, but no issues at 128. This is on a totally unoptimised i5 750/win7 x64/4gb system with onboard FW (ASRock mobo!) bought from eBay for about £400, with wireless ON and also Avast AV...seriously good drivers, mate. A typical project runs 8- 10 instances of Guitar Rig 4, Steven Slate (Kontakt 4) drums and a couple of VSTIs...and obviously various FX when when mixing, including convo reverbs and look ahead FX like T-Racks Algo EQ...

--------------------
My metal music! My other music!


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robinv



Joined: 31/08/04
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919163 - 09/06/11 07:26 AM
Quote TAFKAT:



Horses for courses, we may have a different client demographic.



Maybe - and thanks for taking my comments in the spirit they were meant. I didn't mean "narrow" as a criticism either, rather just indicating where your focus lies - which is fine! I wasn't making any criticisms just talking from another viewpoint

Quote TAFKAT:


but I disagree that the majority of clients working latencies are 256/512, anyone who can actually play would not be comfortable using a VI at those latencies, I know I wouldn't.




But that's a judgement call - i play, can "actually" play guitar, keyboards and percussion. I use an Alesis Performance pad to enter drum tracks using Battery or Abbey Road on 256 samples - i havent moved the buffer in years. I do everything in the box, effects, instruments the lot. Either i'm an idiot or perhaps it's not as unplayable as you suggest, or at least to everyone - i concede that i might be an idiot.


Quote TAFKAT:


In short it simply negates a huge potential of the advantages we enjoy using these current uber fast systems.




That suggests that one can only enjoy their system if they are running it at "acceptable" latencies as defined by you. I don't think any one chooses to make their system unplayable - i dont opt for 256 because i like a bit of lag, it's rather that it doesnt occur to me to change it because for me there is no lag, it is real-time and i dont need to sacrifice more CPU cycles to get it better than real-time.

My point about the tech support is that honestly i havent had a call in years about latency and i regularly supply Focusrite, Presonus, Steinberg and RME interfaces set to 256 as default. I don't doubt that other people have different needs but i remain unconvinced that it's all about the buffer size, and what people believe is playable can vary.

Quote:

The past work has already resulted in better drivers and multiprocessor performance on some DAW hosts , revolution may be too lofty, but this will grow legs , how hairy will be up to the audio hardware developers.




That's excellent mate. If you talk to Digi then you'll find it's all Microsoft's fault and their appalling implementation of the firewire protocol, so you'd best get Balmer in your sights too

Which reminds me - do you have any reference data of buffer sizes on the Mac?

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Pete Kaine
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #919189 - 09/06/11 08:54 AM
I agree with you to a point Robin but I'm very much in Vin's camp in wanting to see certain firms get a kick up the arse.

I've been heard to mutter about the DICE controller any number of times now and an engineer I've worked with was sent the DICE development pack for use in his use in developing interfaces when it was first brought to market, and after some time with it he simply decided it wasn't good enoough for the project they were doing and for me that is the right attitude.

My own testing be it Alesis, Focusrite, Prosonus shows up performance loss (I know the's a few more names out there with problems that I haven't tested) compared to other cards not using this generic solution that perform far better. I have to ask myself why would I wish to recommend one of those interfaces when i know other brands can outperfom them?

Focusrite I stick with for it's well regarded signal chain, and I'll recommend those to players who want a good cheap "recording" device working with a limited number of channels of audio, but I wouldn't be so keen to offer them to someone working with electronica or mixing and processing heavily in the box who needs the performance.

Given those reasons why would I want to recommend a device from the other two above when I know I already stock better hardware than those in regards to performance when we should be guiding end users to the most suitable packages for their requirements?

Why put in the effort to develop well tuned systems as DAW builders if we're going to hamstring ourselves with substandard interfaces?

I'd say the vast majority of the time I get a customer with an interface problem it's a DICE box... Why would i want to go with a solution that I know can be problematic when the end user just want's a box that works?

As much as I'd love to see it I don't see the firms out there making the effort to invest in new solutions, otherwise they would have done already instead of going off the shelf with their current models. But any tuning of the drivers that a thorn in the side campaign brings can only be a good thing.

*Edit* Robin - A lot of those devices work great on mac's and don't seem to have the same issue they do on the P.C. The's a couple I tested as being unusable on the P.C. which run fine on a MAC so yeah the cat calling over Windows firewire implimentation has some merit...

But we're talking about half a dozen firms using the SAME controller solution which they apparently know doesn't work great in windows.

In that case stop wingeing about the windows implimentaion and do something about your code! The's a load of you surely you can pool your resources and do something about it rather than just shrug your shoulders and point the blame! If Maudio can get it right in their budget end of the market then what excuse do the rest of them have?

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ScanProAudio & 3XS Audio Systems
ScanProAudio Blog


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #919220 - 09/06/11 10:28 AM
Quote robinv:



That suggests that one can only enjoy their system if they are running it at "acceptable" latencies as defined by you. I don't think any one chooses to make their system unplayable - i dont opt for 256 because i like a bit of lag, it's rather that it doesnt occur to me to change it because for me there is no lag, it is real-time and i dont need to sacrifice more CPU cycles to get it better than real-time.




Hang on, I am not making any suggestions of the sort , also what is an acceptable latency is defined by the individual end user and their working environments and requirements. 256 samples is not real time when using RTL- Round Trip Latency,anything above 064 will actually feel laggy for guitarists , I personally know drummers who are even further hampered , so just because your working environment works for you, it doesn't mean it will for others. That doesn't make you any more or less an idiot or otherwise, than those that will not find 256 acceptable.

Re CPU cycles, a good audio interface with efficient drivers will have minimal variance between the moderate latencies of 064/128 , I have seen variables as low as 20% between 032 samples and 256 when scaled to break, so if you are getting substantial CPU cycle hits running anything below 256, then I would be taking a serious look at the interface you are using.

Quote:

I don't doubt that other people have different needs but i remain unconvinced that it's all about the buffer size, and what people believe is playable can vary.




While you place far less importance on efficient LLP , I have yet to see an audio interface manufacturer market their products in that manner, quite the opposite. Most if not all will try and sell the merits of low latency , whether they can actually deliver the efficiency is another matter. If it was of no importance to the end users, why would they bother emphasizing the low latency aspect.

Quote:

If you talk to Digi then you'll find it's all Microsoft's fault and their appalling implementation of the firewire protocol, so you'd best get Balmer in your sights too




Interestingly the new AVID MboxPro 3 is outperforming both the M-Audio and the RME's on some recent testing I have done, even on a non TI generic OHCI FW controller on my latest audio notebook , while a Focusrite unit would not even be detected at all, so I think Digi and Microsoft are in the clear, Focusrite/DiceII on the other hand.....

Quote:

Which reminds me - do you have any reference data of buffer sizes on the Mac?




Well thats a whole other thread as Core Audio is inherently not as efficient as ASIO , plus the cross platform Hosts are not on par , so its kind of harder to get a head to head. I did some testing with RME/M-Audio in the last batch of cross platform DAW testing , and in short, the windows hosts / ASIO driver at any of the respective latencies are more efficient.

I do plan to include more OSX comparative data if there is enough interest from that side of the pond.

Peace

V:


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Sam Inglis
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919241 - 09/06/11 11:24 AM
Interesting tests there Vin.

Can anyone confirm for sure that the Focusrite Saffire Pro 40 definitely uses the Dice II chipset? I've never managed to get very low latency out of it on my Dell laptop, but at least it works -- by contrast, the same laptop won't recognise my M-Audio ProFire 2626 at all, and did not work properly with a Mackie interface that I know uses Dice II.


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Pete Kaine
Scan Computers


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Sam Inglis]
      #919259 - 09/06/11 12:11 PM
I might be wrong... and I should know better than using Google to back up a claim, but:

http://bit.ly/iBKi08

(used because the resultant link broke it's own formatting in the message board code)

--------------------
ScanProAudio & 3XS Audio Systems
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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Sam Inglis]
      #919290 - 09/06/11 02:24 PM
Quote Sam Inglis:

Interesting tests there Vin.

Can anyone confirm for sure that the Focusrite Saffire Pro 40 definitely uses the Dice II chipset? I've never managed to get very low latency out of it on my Dell laptop, but at least it works -- by contrast, the same laptop won't recognise my M-Audio ProFire 2626 at all, and did not work properly with a Mackie interface that I know uses Dice II.




Hey Sam,

As someone noted earlier, the listing of the JetPLL™ in the spec is a give away that its Dice II.

Laptops are a moving target, especially if they are non TI, and even if they use a TI card in an Express slot. It can depend on what chipset/ controller the actual Express slot is , and even how the low level resources are assigned at the board/BIOS level.

Its interesting that you have mentioned having trouble with the Profire 2626 , you are the second person to mention that to me in recent weeks, interestingly the Profire 610 works perfectly on the same notebook , and they share the identical driver , so it has to be some variance at the controller level.

FWIW: The performance of the 2626 is identical to the Profire 610 on the desktop and miles ahead of any Dice II interface I have ever tested.

Re Mackie using Dice II, more than likely on their rack units, but I recently tested an Onyx Mixer with the onboard FW controller and it definitely wasn't Dice II. Available buffer settings all the way down to 032 samples , and it performed quite well across all buffer settings with DAWbench DSP , DAWbench VI was another story tho.. :-(

What was really odd with that unit was that it did not report latency values to Cubase past the theoretical values for the respective buffer settings, sp no AD/DA , no safety buffers, etc, so it could have been padded to the hilt with double buffers on playback and we would be none the wiser.

Peace

V:


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #919295 - 09/06/11 02:52 PM
Quote Martin Walker:

It looks as if I ought to set up a thread/poll so SOS Forum users can tell us what buffer size they currently use, and for what purpose (live monitoring, playing softsynths, or mixing/mastering), so we can all get a better handle on who wants what




OK - I've just created this and posted it on the Music & Recording Technology forum so we get both Mac and PC votes. Here you go - vote away and lets' see what we can deduce from the results!

www.soundonsound.com/forum/showflat.php?Cat=&Number=919293


Martin

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James PerrettModerator



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919316 - 09/06/11 03:36 PM
As a user I'd offer a different perspective - I want stability more than absolute low latency. I remember certain Focusrite testers around here saying that early versions of the DiceII based Saffire Pro40 were much more stable than the Bridgeco based Saffire Pro26. I suspect that M-Audio have stuck with Bridgeco chips as have Prism. Apart from RME, are there any other manufacturers making firewire audio chipsets?

James.

--------------------
JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Pete Kaine]
      #919321 - 09/06/11 03:43 PM
Quote Pete Kaine:



In that case stop wingeing about the windows implimentaion and do something about your code! The's a load of you surely you can pool your resources and do something about it rather than just shrug your shoulders and point the blame! If Maudio can get it right in their budget end of the market then what excuse do the rest of them have?



I don't disagree with that. My Edirol FA66 beats everything i've tried and it's on version 1.0 of the 64bit driver - it was right first time. Shame it's not a very inspiring box
I confess that since last year and my laptop testing i've tended to avoid Presonus and favour other interfaces when asked for an opinion - but at the same time these are companies you want to work with and have success with so, you know, depends, as you say......

--------------------
Molten Music Technology - Computers for doing music on


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919326 - 09/06/11 03:52 PM
Quote TAFKAT:

so just because your working environment works for you, it doesn't mean it will for others. That doesn't make you any more or less an idiot or otherwise, than those that will not find 256 acceptable.



That, my friend, is exactly what i'm trying to say

Quote:



While you place far less importance on efficient LLP , I have yet to see an audio interface manufacturer market their products in that manner, quite the opposite. Most if not all will try and sell the merits of low latency , whether they can actually deliver the efficiency is another matter. If it was of no importance to the end users, why would they bother emphasizing the low latency aspect.



Oddly i think they tend to emphasise the misleading "zero latency monitoring" angle rather than low latency as such.


Quote:


Interestingly the new AVID MboxPro 3 is outperforming both the M-Audio and the RME's on some recent testing I have done, even on a non TI generic OHCI FW controller on my latest audio notebook , while a Focusrite unit would not even be detected at all, so I think Digi and Microsoft are in the clear, Focusrite/DiceII on the other hand.....



All very interesting.

Quote:



I do plan to include more OSX comparative data if there is enough interest from that side of the pond.



Cool.

I think this is all fascinating. I'm still looking for an interface to replace my Delta 44/Omni which is, god, 10 years old or something so i'm always on the lookout. Disappointed that there were no interesting things at the shows this year and lots that seem to appear and disappear like Alesis and Lexicon boxes. Hey can we get together to design and produce our own box?

--------------------
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Pete Kaine
Scan Computers


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: James Perrett]
      #919334 - 09/06/11 03:58 PM
Quote James Perrett:

I remember certain Focusrite testers around here saying that early versions of the DiceII based Saffire Pro40 were much more stable than the Bridgeco based Saffire Pro26. I suspect that M-Audio have stuck with Bridgeco chips as have Prism. Apart from RME, are there any other manufacturers making firewire audio chipsets?





Maybe that comes down to drivers again and kinda comes back to the fact they have got it right "now". I've got a Maudio 1814 I've been using as a test device since for about 5 years now. I probably wouldn't have trusted it on xp but on win 7 it's been flawless.

I've had it down trade shows at earls court and the NEC running 8 or 9 hours a day for weeks at a time and it's rock solid and we're pushing it in some pretty daft situations. Perhaps they got a new coder or two care of Avid after the take over, or perhaps Win 7 is just better to work with driver wise (I'm not a code monkey so I couldn't say). The DICE situation whilst stable and I'd never say otherwise (you've just got to get it working first...) hasn't really improved performance wise from one driver set to the next, so I have to question the hardware in that instance.

All very nice saying it's stable but then so is a ford focus. I don't want an engine from one of those in my Ferrari through

--------------------
ScanProAudio & 3XS Audio Systems
ScanProAudio Blog


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Sam Inglis
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919345 - 09/06/11 04:23 PM
Quote TAFKAT:



Re Mackie using Dice II, more than likely on their rack units, but I recently tested an Onyx Mixer with the onboard FW controller and it definitely wasn't Dice II. Available buffer settings all the way down to 032 samples , and it performed quite well across all buffer settings with DAWbench DSP , DAWbench VI was another story tho.. :-(




That's interesting. This was the Onyx Blackbird rackmount unit and I know it's a DICE chipset because Mackie told me so.

I wonder if this might make an interesting SOS feature -- PM me if you want to talk about it further!

Sam


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Sam Inglis
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: James Perrett]
      #919346 - 09/06/11 04:24 PM
Quote James Perrett:

Apart from RME, are there any other manufacturers making firewire audio chipsets?





Yamaha perhaps?


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Remeniz



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919353 - 09/06/11 05:09 PM
Mmm...

Any results for the Steinberg MR816?

I was working at 32 sample buffer setting until I realized that I was running @ 96Khz sample rate and went into the MR editor to go back down to 44.1Khz.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Remeniz]
      #919412 - 09/06/11 09:35 PM
Quote Remeniz:

Any results for the Steinberg MR816?





Hey R,

I have been wanting to put one thru its paces, but the pricing here in Oz is so ridiculously out of proportion to the US market for example , that no one wants to touch them locally. I have heard mixed reports, over at the Cubendo forum I had a few end users running some numbers on a near identical system to mine, but they couldn't get anywhere near my better FW numbers , on the other hand a trusted DAW colleague of mine in the US tells me they are neck and neck with the RME's for example.

I'll try and get in front of one for the new round of testing, I'd be really interested in getting a clear angle on them.

Peace

V


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #919416 - 09/06/11 09:52 PM
Quote robinv:

Oddly i think they tend to emphasise the misleading "zero latency monitoring" angle rather than low latency as such.




Hey Robin,

Sadly end users who do not understand the mechanics simply see the term low latency and think its all encompassing.. :-(

Re designing a box , or helping to design one, I would love to if given the opportunity , but seriously, I can't see it being different to some of the better ones already on offer, I just wish there was a wider spread at the lower budget ranges, but M-Audio and AVID are proving that can be done. Also I have a little interface that I tested a while back from a lesser know developer that sits firmly at the budget end that bested the best, more on that later... :-)

Peace

V:


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redleicester
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919431 - 09/06/11 10:57 PM
Quote TAFKAT:

Sadly end users who do not understand the mechanics simply see the term low latency and think its all encompassing.. :-(





It's one of the inherent dangers of Martin's latency survey thread - one man may be able to get away with 6ms buffers as they have highly efficient and well clocked MIDI, another may struggle at 6 simply because their MIDI is all over the place... one man's tiny buffer size is another man's.... oh dear, I think I'll stop there!

--------------------
Ars longa, vita brevis, occasio praeceps, experimentum periculosum, iudicium difficile.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919456 - 10/06/11 03:38 AM
The plot thickens..,

I have been doing some research on the OEM chips being used by the manufacturers , and the more I discover the murkier it gets. First off there are different versions of the BridgeCo chip being used, also the chip being used by M-Audio on the Profire 610/2626 range is the Dice chip - not Dice II, buts its precursor , the rest of the FW range uses one of the BridgeCo chips. Focusrite uses the Dice II across the current range , while Presonus use both Dice II & BridgeCo , Apogee, Prism also use BridgeCo.

There is wide and varied performance on the range of interfaces using the BridgeCo chips , so obviously not all chips are equal and the base driver can be improved on , also , the original Dice Chip is probably the best of the bunch but from the information I could gather, it was not an available option to a developer/manufacturer that I spoke to this morning, so how M-Audio has managed to corner that chip is interesting in itself.

What was that about taking the red pill.. :-)

V:


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James PerrettModerator



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919521 - 10/06/11 09:09 AM
Quote TAFKAT:

so how M-Audio has managed to corner that chip is interesting in itself.





I'd guess that they bought all of Wavefront's remaining stock when they decided to discontinue it. I've been in a similar situation this week with a chip that I wanted to use - the manufacturer won't sell them to me unless I order 10,000 whereas I can't see us building more than 20 units.

James.

--------------------
JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: James Perrett]
      #919603 - 10/06/11 01:02 PM
Quote James Perrett:

I'd guess that they bought all of Wavefront's remaining stock when they decided to discontinue it. I've been in a similar situation this week with a chip that I wanted to use - the manufacturer won't sell them to me unless I order 10,000 whereas I can't see us building more than 20 units.




Hey James,

You are more than likely correct, much in the same way UAD bought up all of the remaining Mpact 2 chips for the UAD1's back in the late 90's when the company went south. I hope M-Audio/AVID have plenty in reserve... :-)

I'm wondering if its the same dev team, as the original Dice was credited to TC Applied Technologies in Canada. Wavefront are listed as being part of the TC Group ( info appears in Google Listing but not on website ), but they could simply have acquired the name.

Its also interesting that Dice II is no longer listed on their products page , the Google listing links to a product .pdf only.

Either way, the 2 chips are chalk and cheese in my experience.

V:


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alex s
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919701 - 10/06/11 08:42 PM
Gosh this is the first thread Ive read every post.

From an audio pc specialist Im sure it helps to narrow down choices to the best.
Why waste money stocking faff. Give clients the best. Keep the reputaion for being a solid audio pc specialist.
You know the ole qual vs quant.
Not to mention helps the bottom line & saves on after sales tearing hair out.

From an audio manufacturer perspective they prolly have cheap (lite) version of the same controller chip, just with different names.
As in they make just one chip but the ones that pass their rigorous torture testing get given to say Prism.
The ones which pass say 75% given to say RME.
Anything below that given to the £100-£300 FW audio boxes.


Anyway they prolly will move away from FW boxes just aint worth the hassle for them any longer.
Not to mention trying to remove ground loops from FW boxes. Good luck on that one.
What's the news on audio boxes having USB3, Thunderbolt.


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Kayvon



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #919914 - 12/06/11 11:59 AM
I never see Marian mentioned much. I'm looking into their PCIe solutions for my next desktop PC. Anyone care to venture any opinions on them?


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Kayvon]
      #919985 - 12/06/11 08:27 PM
Quote Kayvon:

I never see Marian mentioned much. I'm looking into their PCIe solutions for my next desktop PC. Anyone care to venture any opinions on them?




Marian unfortunately do not have any x64 drivers at present, which kind of leaves them off the radar for many if not most.

I have access to them, fingers crossed they get the x64 driver sorted.

V:


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Jez (mahoobley)
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Remeniz]
      #919992 - 12/06/11 08:43 PM
Quote Remeniz:

Mmm...

Any results for the Steinberg MR816?

I was working at 32 sample buffer setting until I realized that I was running @ 96Khz sample rate and went into the MR editor to go back down to 44.1Khz.




Being the user of an MR816 I doubt the results for it will be terribly good. With anything other than the most simple of projects I find it starts crapping out at anything below 192 samples - and I believe it's actually using double that.

Fortunately I find that acceptable for my needs. Unfortunately there are other issues with the device that make it somewhat of a pain in the arse - the big one being the way it zeros all it's monitor inputs when you quit Cubase/Nuendo and I have to manually load a preset to bring them back up every time or I get silence from the gaming card I have routed through it.

Steinberg have been aware of this issue for over two years but you know what Steinberg are like for fixing bugs with their hardware!

--------------------
http://www.jeremycorbett.co.uk


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920020 - 13/06/11 05:16 AM
Some results from the latest testing of the current batch of audio interfaces across both DAWbench DSP and DAWbench VI :

DAWbench DSP - RXC

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

AVID MBox Pro 3 : Driver : 1.0.11

064 - 134 RXC : I/O - 2.517/2.517: RTL - 5.034

128 - 148 RXC : I/O - 3.968/3.968 : RTL - 7.936

256 - 152 RXC : I/O - 6.871/6.81 : RTL - 13.742

----------------------------------------------------

DAWbench VI - CV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

AVID MBox Pro 3 : Driver : 1.0.11

064 - 140 Notes Of Polyphony : I/O - 2.517/2.517: RTL - 5.034

128 - 280 Notes Of Polyphony : I/O - 3.968/3.968 : RTL - 7.936

256 - 500 Notes Of Polyphony : I/O - 6.871/6.81 : RTL - 13.742

512 - 620 Notes of Polyphony : I/O - 12.676/12.676 : RTL - 25.534

-----------------------------------------------------------------

DAWbench VI - NCV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

AVID MBox Pro 3 : Driver : 1.0.11

064 - 200 Notes Of Polyphony : I/O - 2.517/2.517: RTL - 5.034

128 - 300 Notes Of Polyphony : I/O - 3.968/3.968 : RTL - 7.936

256 - 520 Notes Of Polyphony : I/O - 6.871/6.81 : RTL - 13.742

512 - 620 Notes of Polyphony : I/O - 12.676/12.676 : RTL - 25.534

-------------------------------------------------------------------

DAWbench DSP - RXC

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

M-Audio Profire 610 : Driver 5.10.0.5082 - 1.0.8.1

064 - 126 RXC : I/O - 3.152/3.152: RTL - 6.304

128 - 148 RXC : I/O - 4.603/4.603 : RTL - 9.206

256 - 152 RXC : I/O - 7.506/7.506 : RTL - 15.012

----------------------------------------------------

DAWbench VI - CV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

M-Audio Profire 610 : Driver 5.10.0.5082 - 1.0.8.1

064 - 160 Notes Of Polyphony : I/O - 3.152/3.152: RTL - 6.304

128 - 280 Notes Of Polyphony : I/O - 4.603/4.603 : RTL - 9.206

256 - 500 Notes Of Polyphony : I/O - 7.506/7.506 : RTL - 15.02

512 - 620 Notes of Polyphony : I/O - 13.311/13.311 : RTL - 26.622

--------------------------------------------------------------------

DAWbench VI - NCV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

M-Audio Profire 610 : Driver 5.10.0.5082 - 1.0.8.1

064 - 200 Notes Of Polyphony : I/O - 3.152/3.152: RTL - 6.304

128 - 300 Notes Of Polyphony : I/O - 4.603/4.603 : RTL - 9.206

256 - 520 Notes Of Polyphony : I/O - 7.506/7.506 : RTL - 15.02

512 - 620 Notes of Polyphony : I/O - 13.311/13.311 : RTL - 26.622

-----------------------------------------------------------------

Good results for both the MBox Pro 3 and Profire 610 , near identical in performance and very similar pattern of reported latency values per respective buffers with both In/Out having equal value. From the data I would suggest they are using the same FW controller ( Dice - Not Dice II ) and the variance in the added buffers on the Profire 610 is down to the inherent latencies of the AD/DA being slightly higher.

O.K Onto the Focusrite Sapphire Liquid 56

DAWbench DSP - RXC

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

Focusrite Sapphire Liquid 56: Driver : MixControl 2.2

064 - 058 RXC : I/O - 1.905/3.900 : RTL - 5.805

128 - 105 RXC : I/O - 3.356/5.351 : RTL - 8.707

256 - 127 RXC : I/O - 6.259/8.254 : RTL - 14.513

-----------------------------------------------------------------

Results were down from the first round of testing with version 2.0 of the MixControl / Driver , so its actually going backwards.

There is no point sugar coating this this , the Dice II FW controller and accompanying drivers are comparatively not even on the same ball park as the AVID/MAido offerings on the DSP test.

Just to drive home the point even further, check out the DAWbench VI results.

--------------------------------------------------------------
DAWbench VI - CV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

Focusrite Sapphire Liquid 56: Driver : MixControl 2.2

064 - 000 Notes Of Polyphony : I/O - 1.905/3.900 : RTL - 5.805

128 - 180 Notes Of Polyphony : I/O - 3.356/5.351 : RTL - 8.707

256 - 260 Notes Of Polyphony : I/O - 6.259/8.254 : RTL - 14.513

512 - 560 Notes of Polyphony : I/O - 12.063/14.059 : RTL - 26.122

---------------------------------------------------------

DAWbench VI - NCV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

Focusrite Sapphire Liquid 56: Driver : MixControl 2.2

064 - 000 Notes Of Polyphony : I/O - 1.905/3.900 : RTL - 5.805

128 - 180 Notes Of Polyphony : I/O - 3.356/5.351 : RTL - 8.707

256 - 260 Notes Of Polyphony : I/O - 6.259/8.254 : RTL - 14.513

512 - 580 Notes of Polyphony : I/O - 12.063/14.059 : RTL - 26.122

---------------------------------------------------------

The drivers collapsed even further in the VI testing, I really have to question anyone using these Dice II interfaces for working environments utilising sample based virtual instruments , you could easily be loosing 50% of your potential system overhead at the preferred tracking/working latencies of 256 and below.

More results as I finalise them.

Peace

V:


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Spangler



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920047 - 13/06/11 09:37 AM
Interesting results Vin - this is exactly what I want to know about any interface.

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clicky


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920454 - 15/06/11 04:35 AM
RME Fireface 800 numbers..

DAWbench DSP - RXC

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

RME Fireface 800 : Driver 3.034 : FW Legacy Mode

064 - 140 RXC : I/O - 2.472/3.628: RTL - 6.100

128 - 145 RXC : I/O - 3.923/5.079 : RTL - 9.002

256 - 148 RXC : I/O - 6.825/7.892 : RTL - 14.717

----------------------------------------------------

DAWbench VI - CV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

RME Fireface 800 : Driver 3.034 : FW Legacy Mode

064 - 160 Notes Of Polyphony : I/O - 2.472/3.628: RTL - 6.100

128 - 280 Notes Of Polyphony : I/O - 3.923/5.079 : RTL - 9.002

256 - 500 Notes Of Polyphony : I/O - 6.825/7.892 : RTL - 14.717

512 - 620 Notes of Polyphony : I/O - 12.630/13.787 : RTL - 26.417

--------------------------------------------------------------------

DAWbench VI - NCV

Cubase 6.0.0

i7 920 @ 2.66 GHZ:

6GB PC12800 DDR 3

Windows 7 x64 Pro

RME Fireface 800 : Driver 3.034 : FW Legacy Mode

064 - 220 Notes Of Polyphony : I/O - 2.472/3.628: RTL - 6.100

128 - 300 Notes Of Polyphony : I/O - 3.923/5.079 : RTL - 9.002

256 - 520 Notes Of Polyphony : I/O - 6.825/7.892 : RTL - 14.717

512 - 620 Notes of Polyphony : I/O - 12.630/13.787 : RTL - 26.417

-----------------------------------------------------------------

The results for the earlier 2.992 driver were near identical except on DAWbench DSP @ 064 samples where the results were down a touch at 130 RXC over 140 on the newer 3.034 driver. However I should point out that the newer driver under the Native TI FW driver, the results for DAWbench DSP were down about 8-10% across the board , the results for DAWbench VI under Native were down around 20 notes of poly at each setting, so I definitely would recommend using the legacy driver as it shows no disadvantage to do so, and a touch more headroom.

On a side note Cubase 6 is proving a little hungrier than its predecessor.

P.S: I'll get some tables / graphs up shortly that will make reading and comparing the data easier.

Peace

V:


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johnny h



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920473 - 15/06/11 08:17 AM
Any chance of you doing win7 / cubase vs OSX / logic. Or renoise win7 / OSX?

thanks


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920567 - 15/06/11 12:42 PM
Hi johnny!

Vin has already posted considerable amounts of data re the relative performance of Cubase on OSX and Windows 7.

However, I can’t really see the point of comparing Win7/Cubase with OSX/Logic - you’re not going to change anyone’s mind about their choice of sequencer, and you also run the risk of drawing incorrect conclusions about whether any performance differences are due to the operating system or the sequencer


Martin

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johnny h



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #920612 - 15/06/11 02:10 PM
Quote Martin Walker:

Hi johnny!

Vin has already posted considerable amounts of data re the relative performance of Cubase on OSX and Windows 7.

However, I can’t really see the point of comparing Win7/Cubase with OSX/Logic - you’re not going to change anyone’s mind about their choice of sequencer, and you also run the risk of drawing incorrect conclusions about whether any performance differences are due to the operating system or the sequencer


Martin



Certainly useful for those people who are thinking of switching sequencers, or starting from scratch. It really doesn't matter to the end user what specific part of the system is slowing down the whole experience! I don't care if its Logic, windows, OSX or the neighbour's cat that is the culprit - its how the whole thing works together when its in front of me that matters.

If the choice is between PC or Mac, a lot of people will choose logic if they go for a Mac, which obviously isn't possible under Windows. Therefore I think comparing Logic's Mac performance to Cubase/Cakewalk/etc's PC performance can help people make the right decision.


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mjfe2



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: James Perrett]
      #920670 - 15/06/11 06:42 PM
Quote James Perrett:

As a user I'd offer a different perspective - I want stability more than absolute low latency.




Ditto


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920691 - 15/06/11 08:39 PM
Quote johnny h:

I think comparing Logic's Mac performance to Cubase/Cakewalk/etc's PC performance can help people make the right decision.




Hey J,

I have covered this numerous times in my other dedicated threads covering DAWbench DSP/VI , firstly my main focus is on testing the cross platform Hosts first and foremost , Logic and Sonar being platform specific are less of a priority for that very reason.

The Logic subject keeps coming up because it is the only Host on OSX that has any chance of giving comparable performance to say Cubase/Reaper on Windows , but Logic in itself is very difficult to do a head to head comparative with using my methodologies because it uses a Hybrid playback engine where despite the input latency setting, playback is always a lot higher - 1024 at default. What that means is what ever buffer setting, the results are the same on my tests - well at least the DAWbench DSP , I haven't ported DAWbench VI as yet.

I may get around to porting the tests across eventually, but to be honest the Logic community is not one that I have ever felt warmed by , so the incentive is pretty low.

Re Sonar, my interest waned a few years back after the numerous encounters with development over the reported issues with an inbalance in thread/core loadings that was repeatedly dismissed by development as a non event. As its still an issue to this day which can compromise the available resources when running at lower latencies, my apathy level is in line with numerous power users that I had worked with in the past, in short , couldn't care less about Sonar.. :-)

I Digress,

For those wondering why I haven't also done an OSX Audio interface performance comparative , well that was the original plan until I discovered we we comparing Apples to Peaches..

Windows 7 x64 Pro / OSX 10.62

Presonus Firestudio Mobile : Driver 3.5.2 :

032 : I/O - Win7 - N/A : OSX 10.6.2 - 4.807/4.807

064 : I/O - Win7 - 1.905/3.900 : OSX 10.6.2 - 5.533/5.533

128 : I/O - Win7 - 3.356/5.351 : OSX 10.6.2 - 6.984/6.984

256 : I/O - Win7 - 6.259/8.254 : OSX 10.6.2 - 9.887/9.887

Say No More......

Quote mjfe2:

Quote James Perrett:

As a user I'd offer a different perspective - I want stability more than absolute low latency.




Ditto




I don't see the argument unless I missed something in the mix.

If there is no instability for those requiring and using the lower latencies with the better interfaces , the question of stability is not even on the table , nor has it been in any of my testing.

Peace

V:


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johnny h



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920695 - 15/06/11 09:00 PM
Quote TAFKAT:

Quote johnny h:

I think comparing Logic's Mac performance to Cubase/Cakewalk/etc's PC performance can help people make the right decision.




Hey J,

I have covered this numerous times in my other dedicated threads covering DAWbench DSP/VI , firstly my main focus is on testing the cross platform Hosts first and foremost , Logic and Sonar being platform specific are less of a priority for that very reason.

The Logic subject keeps coming up because it is the only Host on OSX that has any chance of giving comparable performance to say Cubase/Reaper on Windows , but Logic in itself is very difficult to do a head to head comparative with using my methodologies because it uses a Hybrid playback engine where despite the input latency setting, playback is always a lot higher - 1024 at default. What that means is what ever buffer setting, the results are the same on my tests - well at least the DAWbench DSP , I haven't ported DAWbench VI as yet.

I may get around to porting the tests across eventually, but to be honest the Logic community is not one that I have ever felt warmed by , so the incentive is pretty low.




So logic has a much more efficient way of handling low latency audio than anything on Windows.. yet you have little incentive to test it! For someone deciding between logic on OSX and Cubase on PC, this is very useful information! It makes no difference whether the optimisations are in the audio driver or the host itself; its the end result to the user that should count.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920711 - 15/06/11 09:52 PM
Quote johnny h:


So logic has a much more efficient way of handling low latency audio than anything on Windows.. yet you have little incentive to test it! For someone deciding between logic on OSX and Cubase on PC, this is very useful information!




Hmmm, its not as simple as saying its more efficient, there are caveats attached to the whole core audio / audio units paradigm, lets say it was a clever way to circumvent some of the them. Samplitude has the same principle employed on Windows, and I don't see anyone falling over themselves to copy it.. :-)

Firstly, this is not the thread to take this up , there are other dedicated threads where this topic has already been covered at other forums

Secondly, this is not a Logic v the Rest of the world thread, which I am sure could be of interest to some and no doubt will get a lively debate , but just to be clear, I said I have no incentive in focusing on Logic - that isn't stopping you or anyone else, knock yourself out.. :-)

Now where were we...


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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920742 - 16/06/11 05:17 AM
Has anyone around here subjected the Roland Octacapture USB2 interface to these latency performance tests? A review on Harmony Central suggest that the low latency performance is very good but I have not seen other test based reviews.


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johnny h



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920744 - 16/06/11 06:41 AM
Quote TAFKAT:

Quote johnny h:


So logic has a much more efficient way of handling low latency audio than anything on Windows.. yet you have little incentive to test it! For someone deciding between logic on OSX and Cubase on PC, this is very useful information!




Hmmm, its not as simple as saying its more efficient, there are caveats attached to the whole core audio / audio units paradigm, lets say it was a clever way to circumvent some of the them. Samplitude has the same principle employed on Windows, and I don't see anyone falling over themselves to copy it.. :-)




Well why not? By not optimising their software in such a way, Steinberg is getting poor performance on OSX. Its easy to say 'oh but its OSX's fault', but that argument is null and void if Logic works absolutely fine on it! Each platform has its strengths and weaknesses and it appears Steinberg appears to be doing very little to optimise its products for OSX.
Quote:


Firstly, this is not the thread to take this up , there are other dedicated threads where this topic has already been covered at other forums




Really, where? Not everyone on this forum is also a regular reader of every other audio forum in the world. If you have no interest yourself in discussing this issue yourself, perhaps you could point us to someone who has?


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920758 - 16/06/11 08:40 AM
Quote johnny h:

If you have no interest yourself in discussing this issue yourself, perhaps you could point us to someone who has?




Heres an idea, try using a search function or even better yet , start another thread yourself.

While you are at it, add Protools, Studio One, Reaper and Ableton Live to your list of cross platform applications that perform measurably better on Windows. Your Steinberg isn't great at optimising for OSX mantra doesn't really hold much water , unless of course every other developer is equally as inept.

O.K, lets now try again to get back on topic.. !

Do you have any comparable performance data for respective audio interfaces , or even any interest on the actual subject being discussed on this thread , if not, then leave it those that do.



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The Red Bladder



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920778 - 16/06/11 09:28 AM
And whilst silly people are discussing the number of angels that are getting down wid da groove on the head of a pin -

iZ-Tech Radar, 1.2 to 2ms (depending on setting and model).


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Tombot



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920781 - 16/06/11 09:48 AM
The thing about dice ii is that it is a complete solution. The manufacturers dont write the drivers, tc applied technologies do. Its tc who need to sort thier drivers out, unless the drivers are pushing the chips to the limit and there's sonething in the design thats limiting them. I dont think anyone apart from tc knows.

--------------------
www.theautobots.com / www.scan.co.uk


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johnny h



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920782 - 16/06/11 09:55 AM
Quote TAFKAT:

Quote johnny h:

If you have no interest yourself in discussing this issue yourself, perhaps you could point us to someone who has?




Heres an idea, try using a search function or even better yet , start another thread yourself.




As you have set yourself up as such an expert I thought such a question would be trivial for you. I don't have any inclination to trawl through horrible geek fest sites like Gearslutz.
Quote:


While you are at it, add Protools, Studio One, Reaper and Ableton Live to your list of cross platform applications that perform measurably better on Windows. Your Steinberg isn't great at optimising for OSX mantra doesn't really hold much water , unless of course every other developer is equally as inept.




They are inept. Protools has measurably worse performance on both Windows and OSX than other DAWs. The fact that Logic performs so well proves that greater performance is possible. The only 'mantra' going on here is your desperation to 'prove' that OSX is always inferior the Windows by ignoring the huge elephant in the room (logic) which disproves your claims.
Quote:


Do you have any comparable performance data for respective audio interfaces , or even any interest on the actual subject being discussed on this thread , if not, then leave it those that do.




If you want to 'prove' to use the superiority of Windows, fine, I have no problem with that. But don't flake out of the logic question, as it seriously undermines all your claims and makes it seem like you have an anti-mac agenda.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920792 - 16/06/11 10:47 AM
Quote johnny h:


If you want to 'prove' to use the superiority of Windows, fine, I have no problem with that. But don't flake out of the logic question, as it seriously undermines all your claims and makes it seem like you have an anti-mac agenda.




The only one who seems to have an agenda here is you, by again derailing another thread into a B.S MAC v PC debate , where this thread had absolutely nothing to do with platform wars.

I have volumes of data that is all in the public domain , all of the data and "claims" are easily quantifiable by anyone who takes the time and energy to actually do the tests , I am not interested in spoon feeding the info to someone with your attitude, as I have learned over the years that no matter what data is brought to the table, you will find your own truth.

Re the elephant in the room , the onus isn't on me to prove anything , if you are so Hell bent on proving the superiority of Logic over any of the DAW hosts on Windows , roll your sleeves and bring something to the table instead of sitting back in the arm chair spitting out ad hominem attacks. Also be sure to bring the empirical data to prove that OSX is not a contributing factor to the performance variables of the other Hosts, be extra sure you include your explanation of why the AU plugin format is so superior as well with its 16 channel limit over VST's 512, and of course why Core Audio with anything but Logic is significantly less efficient due to needing to make 5 calls to the kernel per sample buffer over ASIO's 3, but of course you already know all of that I'm sure.

To the rest reading in,

I apologise but it looks like this thread is derailed for the time being , I really have no interest in continuing with a pointless and irrelevant cyclic debate with the resident under bridge dweller. I'll drop back when I have some final numbers on the current audio interface test pool I have at my disposal..

Till then

Peace

V:


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Will_m



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920799 - 16/06/11 11:07 AM
Quote johnny h:


So logic has a much more efficient way of handling low latency audio than anything on Windows.. yet you have little incentive to test it! For someone deciding between logic on OSX and Cubase on PC, this is very useful information!




As martin has said I'm not sure there will be many people using these numbers to decide on whether to go for cubase on pc or logic on a mac, that's quite a big difference to base on differences in low latency performance handling. I imagine more what you're looking for is a defence for OSX and it's poor results in cross platform benchmarks. It's just a little difficult to believe that someone as outspoken as yourself on the platform debate would be considering buying a windows machine.

The purpose of this thread is examining the differences in performance of various audio interfaces on Windows. If you would like to challenge the benchmarks then please by all means conduct your own research.

--------------------
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johnny h



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Will_m]
      #920805 - 16/06/11 11:24 AM
Quote Will_m:

Quote johnny h:


So logic has a much more efficient way of handling low latency audio than anything on Windows.. yet you have little incentive to test it! For someone deciding between logic on OSX and Cubase on PC, this is very useful information!




As martin has said I'm not sure there will be many people using these numbers to decide on whether to go for cubase on pc or logic on a mac, that's quite a big difference to base on differences in low latency performance handling. I imagine more what you're looking for is a defence for OSX and it's poor results in cross platform benchmarks. It's just a little difficult to believe that someone as outspoken as yourself on the platform debate would be considering buying a windows machine.

The purpose of this thread is examining the differences in performance of various audio interfaces on Windows. If you would like to challenge the benchmarks then please by all means conduct your own research.



I use windows right now. I am considering a switch to OSX but I am not happy with the Cubase / OSX numbers. I really didn't mean to cause such grief here. All I asked was for the logic numbers, and also to address the issue of why Steinberg is unable to perform as well on OSX as Logic.


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: johnny h]
      #920849 - 16/06/11 02:19 PM
Quote johnny h:

I really didn't mean to cause such grief here. All I asked was for the logic numbers, and also to address the issue of why Steinberg is unable to perform as well on OSX as Logic.




Hi johnny!

If that's truly the case I suspect you owe TAFKAT an apology, considering the HUGE amount of work he's carried out over many years measuring all sorts of PC-audio-related issues and being brave enough to go out on a limb with the results and face the wrath of the manufacturers

It's easy enough to come up with suggestions, but the work TAFKAT does can require days of careful measurements for each investigation he carries out, to be repeatable and eliminate possible errors.

Oh, and if anyone else wants to find out more about his work, point your browsers at:

www.dawbench.com

Be prepared for LOADS of detailed analysis, graphs and caveats - this isn’t bedtime reading


Martin

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James PerrettModerator



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #920872 - 16/06/11 04:41 PM
Quote TAFKAT:



Quote mjfe2:

Quote James Perrett:

As a user I'd offer a different perspective - I want stability more than absolute low latency.




Ditto




I don't see the argument unless I missed something in the mix.

If there is no instability for those requiring and using the lower latencies with the better interfaces , the question of stability is not even on the table , nor has it been in any of my testing.






I said that because of some comments that were made by reviewers when Focusrite replaced their BridgeCo interfaces with the DiceII based interfaces. Apparently the Focusrite DiceII interfaces and drivers were more stable than their older siblings when they were first introduced. I'm not amazingly impressed with the reliability of my Bridgeco based Saffire Pro26.

James.

--------------------
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http://www.jrpmusic.net


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #921746 - 21/06/11 08:56 PM
Quick heads up,

Ran up the RME HDSP AIO card which is to be used as the base reference with the latest 3.24 driver and found some interesting performance variables in behaviour and results compared to the 3.08x drivers I had been using in the past.

The behaviour of the driver when pushed to the limit is very different to the previous, where the 3.08x drivers could be pushed right into and past the red on the ASIO meters and still deliver without causing too much garbage on playback , the new driver is a very different beast which will simply bork completely when tipped and also has an issue on loopback more so than the previous when close to the limit.

Results were down 5-10 % points which kind of scuttled my baseline idea a bit, so I have used the earlier 3.08x driver as the baseline and even the newer 3.24 driver is then gauged against it in regards to the LLP Rating.

I'll post the results later in the week for all of the interfaces tested on the new round using the new table format.

* On a side note, I went back and retested all of the FF800 results again just to be 100% sure, the one that stuck out a bit was the 064 results under legacy on the newer driver , and of course I couldn't repeat the 140 RXC result again, so I have amended the result back to 130 - which is the same as the 2.999x results under legacy. *

Stay Tuned

Peace

V:


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #921846 - 22/06/11 12:12 PM
So, it’s fast looking as if to get optimum performance at very low latency you need a particular make and model of interface running a particular driver version on a particular day when the wind is in the right direction


Martin

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #921927 - 22/06/11 08:59 PM
Hey Martin,

LOL, its not quite as drastic as that.

Re the latest RME 3.24 driver , there are new included features in regards to WDM streaming, so obviously some tweaks have been made to the driver as well as a new firmware to accommodate. I tested the driver first with the old firmware and the new one, and results were identical. Also the previous 3.085 driver works as before under the new firmware, which is a relief to be honest as re-flashing to a previous firmware could be a problem.. :-(

I need to be 100% transparent in this testing, so even RME themselves despite being the base reference, will have their new drivers gauged against their earlier work.

V:


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922080 - 23/06/11 02:07 PM
O.K, here are the results for the first batch of audio interfaces on the new round of testing.



Explanation of how the LLP- Low Latency Performance Rating is derived.

The results for the DAWbench DSP RXC across the latencies of 032 thru to 256 ( which has been the M.O for the last 5 years ) are added and the total is then % wise gauged against the result for the RME HDSPe AIO baseline card. The same is then calculated for the DAWbench VI CV/ NCV tests for 032-512.

Those 3 % results are then added and divided by 3 to give an average % .

I thought it important for the I/O and RTL figures to be an influencing factor on the rating as some cards have a lot lower overall latency than others, so the average % results is then multiplied by the last % result for the RTL.

How the RTL % is calculated is I combine the total of the RTL's across the specific available buffer settings for the cards ( all cards past the base reference RME were calculated using 064-512 except the M-Audio FastTrack which had a range of 128-512. You will notice 2 values listed for RTL for the RME HDSPe AIO , first being for 032-512, second being 064-512 ) , and then calculate the % variable against the baseline. You will see that some cards are actually better than the RME AIO card in regards to I/O /RTL i.e RME Babyface and the AVID MboxPro 3.

I think that is a fair appraisal using the collated data, and it gives deserved credit and advantage to those cards that do have lower individual In/Out and Round Trip Latencies.

Some cards do not report the correct I/O values to the DAW Hosts , only reporting the nominal value for the respective buffer sizes. In those instances I will make a note and default the RTL rating to .95 which is below the average of the other cards RTL % rating of around .97 , not perfect but a reasonable compromise.

A few side notes - the baseline reference is the RME HDSPe AIO card using the 3.08.5 driver. You will notice that even the new version 3.24 driver is also tested and gauged against that baseline result as the performance has dropped. If a later RME HDSPe driver levels or betters the 3.08.5 driver, then it will become the new baseline, unless of course another card has better overall performance and becomes the new baseline.

The performance of the M-Audio Fast Track Ultra was actually quite good for a USB2 interface at the available latencies, but its rating suffered due to the lowest available buffer being only 128. The results for the RME Babyface was very impressive for a USB 2 interface , edging out the FW - Profire 610 in the rating due to its lower overall I/O and RTL. The Result for the Focusrite unit using the dreaded Dice II FW controller , speaks for itself !

More results and analysis coming in the next few weeks

Stay Tuned.


V:

Original text edited at vin's request by MW

Edited by Martin Walker (26/06/11 12:03 PM)


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922081 - 23/06/11 02:12 PM
Woo - these look good Vin!

Will the manufacturers with the best-performing interface get a specially-presented DAWbench trophy of excellence?


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922082 - 23/06/11 02:15 PM
Hey Martin,

The think the better performing will place me on their X-Mas card list, the not so better performing will probably send out the hit squad... LOL

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922189 - 24/06/11 12:36 AM
Hi Vin,

Excellent work as ever, however the ESI latency results are not credible.

If you look at the figures, you find that they add up exactly to the size of the nominal buffers: 1.451mS = 64 samples (63.989, to be foolishly precise), 2.902 is obviously double that (128 samples), 5.804 is 256 samples, and 11.610 is 512 samples (all at 44.1KHz).

Only the buffer size is being reported. The real giveaway is the fact that the RTL at any of the four buffer sizes is identical to the 'in' or 'out' latency at the next buffer size up.

None of the other interfaces with symetrical i/o buffers exhibit this behaviour.


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922191 - 24/06/11 01:13 AM
Hey John,

Yeh, I think you are right about the ESI only reporting the nominal buffers, I did notice that each successive buffer was double the previous when I was finalizing the numbers and raised an eyebrow with a note to revisit.

I'll double check in Reaper and StudioOne see if its reporting the same numbers , if so then its obvious reporting within the ASIO driver, and not something to do with Cubase.

I may even break out the CEntrance Latency Tool and see what it spits back via the AD/DA.

In the meantime I do to have a rethink about the RTL % results and adjust back to a say a 1.00 %, which will drop its rating.

Thanks for the catch and the prod.. :-)

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922192 - 24/06/11 02:16 AM
Quick heads up,

All DAW hosts are reporting the same values on the ESI so its obviously inherent in the driver to report back only the nominal values, which is disappointing.

I didn't have much luck with the CEntrance Utility , only managing to get one reading which was so high I dismissed it and on subsequent attempts via both analog and digital was not getting any readings at all. Even if I manage to get an accurate reading, I am not entirely sure how it actually correlates to the values given by the other cards :-(

In the interim I did an average % calculation of all of the RTL ratings across the other cards and it came to .97% , so I have decided to assign a .95% value for cards that are not properly reporting the latency values, I know its not entirely accurate but it will allow a more reflective LPP Rating. The ESI has dropped to 7.50 using the latest calculations.

I'll update the tables and repost.

@ Martin and the other mods, its frustrating that I cannot go back and edit and amend my earlier post after the time out has expired, whats with that ??

This is the only audio forum that I know of that does have that timeout/lockout.

I digress..

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922343 - 24/06/11 04:21 PM
Quote TAFKAT:

@ Martin and the other mods, its frustrating that I cannot go back and edit and amend my earlier post after the time out has expired, whats with that ??

This is the only audio forum that I know of that does have that timeout/lockout.




Drop me a PM - I can edit your previous post for you if you let me know exactly what you want to change.


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922395 - 24/06/11 09:45 PM
Done.

Tables and LLP Ratings already amended.

Thanks again to John Roberts for the catch... :-)

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922617 - 26/06/11 12:04 PM
Text now updated as per your request


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922714 - 26/06/11 08:23 PM
Thanks Martin.. :-)

I have finished testing the AVID HD Native card under ASIO and to say it was an interesting experience is an understatement. Input buffers are double the usual, Output are reporting Nominal , overall performance isn't too bad right down to 032 samples, but there were some major hoops to activate the ASIO driver in Cubase . i.e : needing to toggle off the ASIO driver back to the Generic and then back again to simply wake the driver on every session load . Interestingly I ran into a similar issue with the ESI on any latency below 512, where I had to reset the driver at each session reload to wake it up.

I am wondering how many interfaces are experiencing the same or similar issues. The only other interface over the years that I have stumbled on the same issue was an early Sapphire a few years back.

The 2 interfaces having the issue on the recent testing both have nominal values ( at least for playback ) , so I am wondering if that is a contributing factor. Also, the ESI had no issue at any latency in Reaper , so it is something to do with the way Cubase is making the call to the driver or vice versa.

With the HD Native card, its obvious to me that the card is specifically geared to Protools , the ASIO driver really being only for convenience , so its not really a player in the current comparative. I will upload the results later , LLP rating was 7.55 - it took a hit in the rating due to the double Input buffers.

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922892 - 27/06/11 07:29 PM
My pleasure Vin!

This is the first time I've heard of an interface with double input buffers - most do it on the output to keep things smooth when folk are aiming for low input latency


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #922916 - 27/06/11 10:05 PM
Hey Martin,

Yeh the double input buffers are a bit weird, but we need to remember this card is specifically designed for PTHD , where the I/O and Mix Engine are controlled directly off the FPGA , so the I/O and RTL of the ASIO driver is not really telling the whole story with the card in its designated playing field, so to speak.

I had a chat to my AVID contact yesterday and discussed the encountered issues and anomalies and unlike some other companies that play ostrich as soon as any negative comments are expressed, the feedback was welcomed and will be passed along to the appropriate dev channels. I am trying to get some detail on the reported I/O and the ASIO driver activation issue , as well as some clarification of some other under bonnet aspects for the FW interfaces as well.

Stay Tuned

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #924966 - 06/07/11 09:58 PM
Quick heads up,

I have finished the testing on the Steinberg MR816CSX and the USB2 Audio component of the SSL Nucleus as well as retesting the RME HDSPe 3.24 driver that was performing below that of the earlier 3.085. Going on a suggestion from M.C from RME that the new Multiclient option could be the cause of the slight performance drop , I retested with the option off, and sure enough, performance is identical to the earlier.

I'll post a summary next week as I have a few more interfaces to test before I post the updated charts.

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #924984 - 06/07/11 11:52 PM
Quote TAFKAT:

Going on a suggestion from M.C from RME that the new Multiclient option could be the cause of the slight performance drop , I retested with the option off, and sure enough, performance is identical to the earlier.




Isn't it reassuring when manufacturers like Matthias Carsten not only have some idea of why their driver efficiency has changed, but can also get it back as low as it was before with a simple tweak

Not many like that around


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #924991 - 07/07/11 04:05 AM
Quote Martin Walker:

Isn't it reassuring when manufacturers like Matthias Carsten not only have some idea of why their driver efficiency has changed, but can also get it back as low as it was before with a simple tweak

Not many like that around




Definitely not many at all.

I'd suggest M.C has a pretty good handle on the under bonnet fine print , and with no fundamental changes to the driver / firmware except the inclusion of the expanded Multiclient aspect , it was the variable most likely.. :-)

RME have the advantage of being in total control of their drivers from top to bottom , unlike the manufacturers dealing with OEM whitebox drivers that they throw a different GUI on and hope for the best..

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #926359 - 14/07/11 02:07 PM
O.K, here are the final results for the current series of testing.



Some notes on the latest interfaces tested.

The MOTU 424 PCI / 24 I/O delivered exceptional performance right down to 032 samples , and despite the overall numbers being below that of the RME reference card, if you take a closer look the I / O and RTL are measurably lower as the MOTU uses very minimal padding on the buffers. By taking that into account using the RTL % rating of 1.17 , it drew level with the reference RME HDSPe.

The MOTU's have always drawn mixed reports on Windows and I know first hand how flaky the cards can be having 3 different revs of 424 cards across the bench in the last few weeks, and its a crap shoot whether the respective revs will work on current systems , but the drivers themselves have always been good with the PCI/PCIe cards, ( as long as you don't use external clocking- LOL )

The ESI U46XL was the USB audio interface that I had hinted at a while back that surprised me and the results speak for themselves. Yes the reported latencies are nominal and it has the oddity of having the highest buffer setting of 256 ( actually one step higher to 288 ) , so I had to do some juggling with calculating % ratings , but seeing that this whole exercise is about LLP , I gave ESI some rope with the interface not having a 512 setting. Great little interface , especially for it being USB2.

Staying with ESI, the Maya 44e is a mixed bag, respectable performance although beaten by the U46 , but the card has an issue where the driver has to be reset in Cubendo every time you open a project with any buffer setting below 512 ??? !!

PITA - Still waiting to hear back from ESI who have initially said they can not reproduce it.

The Steinberg MR816 delivered decent overall performance, latency settings available all the way down to 032 samples, mind you that's with the help of some extra padding on the output buffers, which resulted in higher RTL values and subsequent lower comparative RTL %. The 032 setting is more Window dressing IMO.

SSL Nucleus - the results don't tell the whole story as the unit is so inconsistent across multiple systems and even respective USB ports. The results are pretty decent but it took some major arm wrestling to settle the system to achieve those results. The driver panel still looks like the vanilla OEM breadboard and the odd non standard buffer sizes are a sore point if you want to try and use it with Protools 9 - which it is advertised as being compatible with - as PT9 simply will not accept the odd buffers.

The Mackie Onyx actually performed O.K for a Mixer/FW combo, DAWbench DSP were quite good, but it collapsed under the VI testing - the OEM controller definately feels different to some of the Dice variations, so not sure what Mackie are using there.

The Midas Venice was an interesting one, basically a mid size live desk with a Dice FW hacked into the back - I/O - RTL were identical and performance was close enough to identical to the Focusrite Sapphire , so its obviously using the same ( poor performing ) controller.

AVID Mbox 3 is identical to the M-Audio FTU range with slightly better I/O and RTL , not too hard to see that AVID are using the same under bonnet components across the AVID Mbox 3 Pro / Mbox 3 / Profire / FTU ranges.

AVID HD Native card under ASIO , well to say it was an interesting experience is an understatement. Input buffers are double the usual, Output are reporting Nominal , overall performance isn't too bad right down to 032 samples, but there were some major hoops to activate the ASIO driver in Cubase . i.e : needing to toggle off the ASIO driver back to the Generic and then back again to simply wake the driver on every session load .

Its obvious to me that the card is specifically geared to Protools , the ASIO driver really being only for convenience for some compatibility with other DAW hosts , so its not really a player in the current comparative.

Thats it until the next round..

Peace

Vin Curigliano
AAVIM Technology
DAWbench.com


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #926403 - 14/07/11 07:36 PM
Quote TAFKAT:



Thats it until the next round..





Simply not good enough - i demand a graph!

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #926418 - 14/07/11 08:42 PM
Quote robinv:

Simply not good enough - i demand a graph!




Ya just can't please some people... LOL

You mean graphs, each for RXC, CV, NCV, I/O , RTL and LLP rating.

I'll see what I can muster up... :-)


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #926972 - 16/07/11 09:24 PM
By popular demand , the LLP results in graphical form.. :-)



* Quick Note - The AVID Mbox Pro 3 results and rating have been amended as I discovered I had a slight miscalculation in the RTL % results which has been amended from 1.04 to 1.05 , not huge but enough to bump the rating a notch. The tables have been amended as well *

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #927168 - 17/07/11 01:51 PM
Dammit man, i wanted it in green!

Ok, trying to get a handle on what your performance rating means. Here's what you said earlier:

Quote:

Explanation of how the LLP- Low Latency Performance Rating is derived.

The results for the DAWbench DSP RXC across the latencies of 032 thru to 256 ( which has been the M.O for the last 5 years ) are added and the total is then % wise gauged against the result for the RME HDSPe AIO baseline card. The same is then calculated for the DAWbench VI CV/ NCV tests for 032-512.

Those 3 % results are then added and divided by 3 to give an average % .

I thought it important for the I/O and RTL figures to be an influencing factor on the rating as some cards have a lot lower overall latency than others, so the average % results is then multiplied by the last % result for the RTL.




So, just to get it straight in my own head the graph shows an overall performance figure in relation to the RME Ai/o which scores the maximum 10. So cards with good low latency performance get near to the 10 whereas those with poorer performance (in comparison) get a lower mark. Is that about right?

Can you conclude then that if you were using a Focusrite Saffire 56 at 128 samples and reached the limits of your system, you could spend 100 quid on an ESI U46XL and actually be able to do a load more tracks, or run more effects or more instruments while staying at the same latency? Is that the reality of what you're saying here?

(please note i'm not trying to be clever, sarcastic or ironic, i'm genuinely asking if this is saying what i think it's saying and how awesome is that?)

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #927180 - 17/07/11 03:04 PM
Hi Vin,

Now that WILL put the cat among the pigeons - make sure you mount extra guards from now on, and have the searchlights on full power in case of reprisals. I’d better buy some extra barbed wire as well

Numerical results can be ignored, but once you print them in a graph everyone can see at a glance how things stack up.

I suspect you’ll now get some flak on how you calculate your LLP (Low Latency Performance) figures next.

Stay strong when it arrives!


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #927182 - 17/07/11 03:08 PM
Quote robinv:

So, just to get it straight in my own head the graph shows an overall performance figure in relation to the RME Ai/o which scores the maximum 10. So cards with good low latency performance get near to the 10 whereas those with poorer performance (in comparison) get a lower mark. Is that about right?




That's it exactly Robin!

Quote:

Can you conclude then that if you were using a Focusrite Saffire 56 at 128 samples and reached the limits of your system, you could spend 100 quid on an ESI U46XL and actually be able to do a load more tracks, or run more effects or more instruments while staying at the same latency? Is that the reality of what you're saying here?




Right again!

Of course you wouldn't expect the audio to 'sound' as good as from a more expensive interface, but ESI have for many years been releasing some extremely capable and versatile drivers that I've always been impressed with in my interface reviews - I'm pleased to see that they are finally getting some recognition for this


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #927247 - 17/07/11 06:59 PM
Quote Martin Walker:



Of course you wouldn't expect the audio to 'sound' as good as from a more expensive interface, but ESI have for many years been releasing some extremely capable and versatile drivers that I've always been impressed with in my interface reviews - I'm pleased to see that they are finally getting some recognition for this


Martin



ESI and Audiotrak before them have always made bonkers drivers, they just do stuff that no one else does like looping GSIF drivers back through ASIO all within the driver model. Always worked great except when they go wrong when they blow your speakers and freeze up solid

You could use the U48 to keep the creative flow going and when you were mixing down switch back to the Saffire once you've bounced a few tracks or instruments - just a thought.

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #927263 - 17/07/11 09:24 PM
Quote robinv:


So, just to get it straight in my own head the graph shows an overall performance figure in relation to the RME Ai/o which scores the maximum 10. So cards with good low latency performance get near to the 10 whereas those with poorer performance (in comparison) get a lower mark. Is that about right?




Correct.

Quote:

Can you conclude then that if you were using a Focusrite Saffire 56 at 128 samples and reached the limits of your system, you could spend 100 quid on an ESI U46XL and actually be able to do a load more tracks, or run more effects or more instruments while staying at the same latency? Is that the reality of what you're saying here?




Well the interfaces you selected aren't really comparative feature wise, but speaking purely in regards to LLP , that is exactly what I am saying. The ESI U46 XL easily outperforms the Sapphire at 128 across the board whether its the number of plugins or even more so, VI polyphony , which is even more dramatic at 256 where the ESI doubles the Sapphire result.

Quick edit - just noticed that Martin had responded in almost exactly the same manner... LOL

@ Martin,

I am expecting some resistance but I think the methodology of calculating the LLP is pretty clear , open and fair. The RTL % is an important variable that gives the advantage to the interfaces with the better I/O,RTL that may not have the respective numbers for the RXC/CV/NVC, case in point being the MOTU.

I am more than happy to converse with the developers if they have an issue with how the data has been presented , but I expect them to have something in the tank in the form of a tangible rebuttal with some empirical data to back in up... :-)


V:

Edited by TAFKAT (17/07/11 09:40 PM)


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #927402 - 18/07/11 12:03 PM
Feeling a little annoyed with Focusrite for the lame performance of the Saffire, I upgraded from an Echo card and noticed worse performance, but had been putting that down to problems with my system. Is this something that could be improved with better drivers or is it a hardware limitation?

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: OD23]
      #927559 - 18/07/11 08:36 PM
Quote OD23:

Is this something that could be improved with better drivers or is it a hardware limitation?




That is a good question and it is something that I am endeavouring to get some clarity on, but its a really sensitive area for the manufacturers using the OEM chips and also the developers of the chips themselves to be too open about the under bonnet detail.

Some of the manufacturers that have not faired so well using one of the variations of the Dice chips by TC Applied Technologies ( A division of TC Electronics ), which I initially thought was Dice II , but have since found out it may not be as Dice II has been long gone , will be less than energetic about disclosing what is being used.

Those performing well will probably be even less so as to not tip off the competition.

From my recent investigations I have learned that Dice currently have 2 available chips, Dice Mini - up to 16 channels of audio and Dice Jr - up to 64 Channels. Presonus / Focusrite/ Midas , etc are using one or both of the Dice variants with close enough to identical poor results , M-Audio/AVID are using one or both of them with incredibly good results.

The question is who is using what, and if its the same chip, why have AVID/M-Audio managed to extract exceptionally good performance, while others are not.

The only answer is drivers , unless of course there are some other variables in play..

How deeps the rabbit hole..



V:


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #928102 - 19/07/11 01:22 AM
Hi Vin,

There are hardware variables, ie, the particular SDRAM and flash ROM packages used, as well as the choice of 1394 PHY chip.

Then there's the efficiency and extent of any third-party code executing on the ARM core (which implies the assumption that any TCAT-supplied OEM code is sleeker than a jaguar, of course!).

Finally, the presence or otherwise of an external DSP/FPGA (and the use to which it's put) could be a factor.

Regards,

John

Edited by John Roberts (19/07/11 01:28 AM)


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: John Roberts]
      #928103 - 19/07/11 02:03 AM
Hi again,

Just had a look at the final table.

Aren't ESI getting a tremendous boost here by 'not telling the truth'?

They're receiving praise for performance/LLP metrics that you and I both know are inaccurate in the latter regard.

And if all the other manufacturers, encouraged by this, follow their example, won't your estimable project have reached a very sad end?

Regards,

John


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: John Roberts]
      #928105 - 19/07/11 02:49 AM
Hey John,

The ESI's may not necessarily be getting a boost as we really don't know whether the .95 figure I have assigned to the nominal latency figures is actually being a benefit or if it is in fact detrimental.

I needed to come up with something for the final RTL result for the cards reporting nominal figures after you brought it my attention, and I did that as I explained earlier by working out the average % of all the other interfaces and working off that. When I initially did the calculation it was .97, with the added list of cards its .98 , so .95 is still below the average , not perfect I know but not sure of what else I can do past make the note that figures are nominal.

I could drop it to 90% as a further penalty if its feels more warranted, or if anyone has a better idea of how to assign a figure to the nominal cards ,I'm open to being guided by the opinions of the others reading in.

I hope manufacturers would not try and bend some rules to get a better slightly better LLP rating by bluring the RTL, instead focus on actually getting better respective performance over all.

P.S : Thanks for the heads up on the other possible hardware variables.. :-)

Peace

V:


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #928107 - 19/07/11 03:18 AM
Quote TAFKAT:

Hey John,

The ESI's may not necessarily be getting a boost as we really don't know whether the .95 figure I have assigned to the nominal latency figures is actually being a benefit or if it is in fact detrimental.

V:




That's the whole point. You could assign any "handicap" you like, but it will be pure guesswork. The only thing you know for certain is that the latency values reported by the ESI devices cannot be true.

In my opinion, the only option which upholds the credibility of DAWbench is to 'suspend' the ESI devices 'pending further inquiries'.

Do bear in mind that among the devices in the final table, ONLY the ESI devices report nominal buffers.

Regards,

John


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: John Roberts]
      #928110 - 19/07/11 04:28 AM
Quote John Roberts:

In my opinion, the only option which upholds the credibility of DAWbench is to 'suspend' the ESI devices 'pending further inquiries'.




. . . With which, putting my money where my mouth is, I'd be happy to assist.



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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #928115 - 19/07/11 06:19 AM
The Mackie Onyx and the AVID HD Native on the output are also reporting nominal... :-(

I have a member at the DAWbench forum currently developing a RTL measuring utility for me that he has tested on his FF800 , and preliminary testing is very close to the reported , so we may have a resolution to the nominal readings shortly.

V:


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #928142 - 19/07/11 09:09 AM
Try this:
MIDI drum pad, close mic'ed, connected to trigger a drum sample in a virtual instrument. The output speaker is also close mic'ed, ideally isolated from the pad. The two mics are connected to the soundcard on seperate channels and set to record. Hit the pad, record the sound of the stick hitting pad and sound of drum sample output. Match them up together in Sound Forge or something and make an exact measurement of the delay between the pad hit and the sound coming out. It's a bit heath robinson but it would measure actual latency ...... i guess.

--------------------
Molten Music Technology - Computers for doing music on


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #928335 - 19/07/11 03:48 PM
Hey Robin,

Thanks Mate, there is only one vital aspect to that method which I don't have a lot in reserve... time... :-(

The Applet being developed is similar in concept to the Centrance utility , but is far more accurate and will take a few minutes across all latencies to get accurate RTL's to those reported in Cubendo/Reaper/SONAR.

V:


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Martin WalkerModerator
Watcher Of The Skies


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #928730 - 20/07/11 07:28 PM
Now that I'd love to have a copy of

The CEntrance LTU is usable, but you do need experience on reading its somewhat varying output readings. Having said that I've used it to sample accuracy on some occasions.

I also used modified MIDI cables and audio loopback tests when I wrote a 2-part investigation into Real World Latency way back in SOS September/October 2002 (yes, I have been doing this stuff for that long )

www.soundonsound.com/sos/sep02/articles/pcmusician0902.asp
www.soundonsound.com/sos/oct02/articles/pcmusician1002.asp


Martin

--------------------
YewTreeMagic


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dickiefunk



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #928873 - 21/07/11 10:50 AM
Wow this has been a very interesting read!

I replaced an old Echo Layla 3G with a Focusrite Saffire Pro 24 DSP and have to say I preferred the audio of the Saffire but noticed a dip in performance!
I didn't realize how bad the Saffire fared against other interfaces on the market ( performance wise )!

Frustratingly when I replaced the Echo I was torn between getting the Saffire Pro 24 DSP and MOTU Ultralite mk3. I picked the Focusrite because I preferred the workflow of the DSP mixer, price and because there wasn't any reviews of the Motu at the time.

Really hoping the Focusrite's performance can be improved with driver updates!!

--------------------
www.richardpenrose.com


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Dee
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #929081 - 22/07/11 12:24 AM
Can anyone separate the wheat from the chaff for me!!

I am currently using a 0404 - but want to upgrade and looking at a few firewire interfaces.

Anyone recommend a unit around the £150-£200 mark pre owned? If there is one around that price highly regarded new then by all means let me know.

I'm seriously struggling to choose an audio interface - I have been looking for ages but just can't nail one main choice.

I will buy a cyber pint to anyone that can recommend me a model or three

--------------------
Hail Hail and up the VSTs


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #929419 - 22/07/11 10:39 PM
Interesting that all of the new audio interfaces released at the Summer NAMM are USB2 - Akai/Focusrite/Presonus.

Plenty of air about "low latency " all over some of the blurbs , hmmmm, O.K.. :-)

Lets see where dust settles

V:


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #930149 - 25/07/11 08:41 PM
Quote Martin Walker:

Now that I'd love to have a copy of

The CEntrance LTU is usable, but you do need experience on reading its somewhat varying output readings. Having said that I've used it to sample accuracy on some occasions.




Hey Martin,

I could never get any consitancy out of the Centrance unfortunately , the new utility can also be finicky on some interfaces, but gives a reading very close to that reported by Cubase when it syncs, which I have managed to get it to do on all interfaces tested so far with it.

The utility will eventually be available to all , the initial build has already proven valuable in that its uncovered that both of the ESI units - MAYA44e and U46XL, employ double buffering on playback judging by the reported RTL's.

A bit cheeky, but it looks like they report nominal to mask the fact they use the double buffering. It could well be an oversight on their part , but still disappointing after my initial impression of the units.

Their respective LLP ratings have been adjusted accordingly down to 5.45 - the RTL % balancing the results against the units that use the double buffering.

Echo are another manufacturer who employ double buffer, but they at least do not try and mask it by reporting nominal figures. I'll also be removing the Mackie unit from the charts/results until I can get some accurate RTL figures.

V:


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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #930230 - 26/07/11 04:24 AM
Any chance you could get your hands on a Roland Octa-Capture USB2 interface to compare with the rest you have tested?


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Jorge]
      #930466 - 26/07/11 08:00 PM
Quote Jorge:

Any chance you could get your hands on a Roland Octa-Capture USB2 interface to compare with the rest you have tested?




That unit looks interesting, I'll dust off my old Roland contact list and see if I can wrangle one to test..

V:


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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931026 - 28/07/11 05:51 PM
That would be great. It seems like the Octa-Capture would be a very useful USB2 unit for both studio and live recordings with up to 8 mics. Anecdotal reports have been generally positive and a review on Harmony Central was very positive but did not quantify latency in a standardized manner like you are doing. For some of us doing live percussion recordings, latency is a particularly important consideration.

Edited by Jorge (28/07/11 05:52 PM)


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931052 - 28/07/11 09:11 PM
I did read that "review" after my attention was drawn to it earlier this week by some others who were reporting it had impressive low latency performance.

I didn't see anything in regards to low latency performance from the reviewer , but I did note this.

Quote:

...... 10 ms roundtrip and 64 samples...pretty darn good!




??

That is pretty much double the RTL the RME USB2 devices deliver , so I suspect not only large added safety buffers but also double buffering on playback.

Personally, after reading that I'd take anything the reviewer reports with a grain of salt.., or 3... :-)

I'll try and get a hold of a Octo/Quad unit to put thru its paces.

V:


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Anderton



Joined: 13/04/10
Posts: 2
Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931264 - 29/07/11 07:36 PM
Quote TAFKAT:

I did read that "review" after my attention was drawn to it earlier this week by some others who were reporting it had impressive low latency performance.

I didn't see anything in regards to low latency performance from the reviewer , but I did note this.

Quote:

...... 10 ms roundtrip and 64 samples...pretty darn good!




??

That is pretty much double the RTL the RME USB2 devices deliver , so I suspect not only large added safety buffers but also double buffering on playback.

Personally, after reading that I'd take anything the reviewer reports with a grain of salt.., or 3... :-)

I'll try and get a hold of a Octo/Quad unit to put thru its paces.

V:




I am the reviewer of whom you speak. A review involves many more components than the focus on a single spec. For some, if not most, musicians and recording engineers audio fidelity is the primary consideration. For others it's I/O, and for those with vintage condenser mics, whether the +48V phantom power is truly +48V (that is not true of all interfaces, particularly bus-powered ones). Others want to be able to use an interface as a stand-alone mixer for field recording, while for others, the nature and quality of onboard DSP might be essential (particularly when recording vocalists who say "Can I have some reverb in the cans?").

Perhaps you did not notice the set of screenshots of audio specs that revealed, for example, the presence of greater amounts of second harmonic distortion than some competing interfaces. For some musicians, that is a positive attribute. For others, it is a negative one. I present that data knowing people are generally capable of drawing their own conclusions, and capable of comparing those screenshots to screenshots from other interfaces I've tested (FYI lowest THD so far of any interface tested: Mbox 3 Pro).

Now, to latency.

I don't care what absolute latency values are, whether double- (or octuple, for that matter) buffering is involved, or what the card reports as latency because these figures are not guaranteed to be accurate due to a variety of reasons. What I care about is the following:

* My primary instrument is guitar. Guitar is a percussive instrument. I use amp sims. Can I set the latency low enough that there is no perceptible delay that would annoy me during the recording process?

* Same with electronic drums and percussive keyboard patches. I absolutely need to monitor through the computer due to the extent to which plug-ins influence the final sound and performance, so the inclusion of "zero-latency" monitoring is not sufficient for 90% of my needs.

* If I can set latency low enough to accomplish comfortable real-time recording - can I do this in the context of a real-world project, with effects, virtual instruments, at least dozens of tracks, etc.? If not, do I need to raise the latency? If so, by how much? Subjectively speaking, I find *total* latencies - regardless of how they're accomplished - of under 10ms to be perfectly acceptable. I monitor about 3 ft away from my speakers, resulting in my "default" latency being 3ms. If I put on headphones, my "default" latency compared to normal monitoring therefore becomes 7ms. I do not have a problem with that. I play concerts all the time where I'm 15 to 20 feet away from monitors or amps, but am not the kind of person who will pout in a corner because the latency is not under 10ms.

I have evaluabed numerous interfaces over the past few decades. My primary concern is audio fidelity. The second is stability. The third is functionality. The fourth is lowest possible latency, although there is an exception where this is a gating issue: if I cannot obtain stable operation using real-world projects without objectionable delay, then the other characteristics become moot.

You are welcome to take my review with a grain of salt, although bear in mind that excessive sodium is not good for your heart. However, by dismissing the plethora of topics discussed in that review that matter to real-world musicians trying to decide how best to spend their money in a tough economy, and the fact that you have not identified anything in the review that is factually inaccurate, with all due respect I believe you do not fully grasp the function a review is supposed to serve.

Also note that if you do find anything that is not factually accurate, the review to which you refer is designed intentionally as an "open source" review where all are invited to participate and present their opinions, even manufacturers. You are welcome to comment on the Octa-Capture in that thread, as is anyone with an interest in the topic. I am always open to corrections; the point of a "pro review" (the name of the open-source format) is to reveal the truth about a product, and I am not sufficiently deluded to think that I can know everything there is to know about every possible application of a product. For example, I cannot say with certainty if a product will work with laptop X unless I have a laptop X for testing. But someone with a laptop X can contribute that information, thus increasing the relevance and usefulness of the review.

Bottom line: Compared with other interfaces I've tested, the Octa-Capture has given me sufficiently low latency, with consistency at a listed buffer figure of 64 samples, using real-world projects, to allow for real-time monitoring while recording. What else matters? Well, of course, aside from all the other evaluations contained in the review regarding audio performance, functionality, suitability for various applications, efficacy of the applet, whether having onboard compression is worth having if it doesn't precede the A/D (my personal opinion: it's not), ease of use of the operating system, the usefulness of unique features like auto-sens, expandability and ability to cascade even when using ASIO, etc...

My mission is to present a factually accurate overview of a product from a real-world perspective, to which potential users can relate, and which they can understand. I believe I have have accomplished that with my review, I stand by the information prseented unless and until there is a need to stand corrected, and make no apologies for the credibility of the review, or for that matter, any other reviews or articles that I write.

(PS - I am enough of a gentleman not to mention a different web site in Sound on Sound's own web site, but I think I've presented enough information that anyone could find the review in a minute or two spent with Uncle Google.)


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Anderton]
      #931283 - 29/07/11 09:58 PM
Quote Anderton:



I am the reviewer of whom you speak.




So you are, thanks for dropping by.. :-)

Quote:

I don't care what absolute latency values are, whether double- (or octuple, for that matter) buffering is involved, or what the card reports as latency because these figures are not guaranteed to be accurate due to a variety of reasons.




Hmm,

Well you may not care Mr Anderton, but those following this thread and reading in here certainly do.

Unlike you, my primary focus of the recent work is on LLP- Low Latency Performance and I have enough resources at my disposal to get a pretty accurate figure if the units are not reporting correctly to the ASIO driver.

Now you can debate all you like how that translates in Real World for you and I'm sure you have more than enough experience to warrant your opinions, but I also have more then enough experience to warrant mine , I have no interest in a pissing competition about your past resume, I know who you are have read plenty of your work and I am pretty familiar with the whole medium of mainstream reviews and where to place them in context.. :-)

Quote:

You are welcome to take my review with a grain of salt, although bear in mind that excessive sodium is not good for your heart. However, by dismissing the plethora of topics discussed in that review that matter to real-world musicians trying to decide how best to spend their money in a tough economy, and the fact that you have not identified anything in the review that is factually inaccurate, with all due respect I believe you do not fully grasp the function a review is supposed to serve.




Slow down there Mate, the only thing I have commented on is the LLP , nothing more . 10 ms round trip at 064 samples is not "pretty darn good " as you put it !

You think you are the only one trying to help real world musicians in how best to spend their money in this tight economy with your reviews , and Seriously, save the anaemic swipe at me not fully grasping what a review is supposed to serve. You may have covered the usual net of S/N , feature sets, etc, etc, which is typical of most "reviews" , I am extending what most reviewers do not want to tackle, and that is the very sensitive aspect of the comparative low latency performance of these interfaces. This is increasingly important if not most important to "Real World" for many end users who get dazzled by the shiny baubles only to find the units fall flat when used in their respective working environments.


Quote:

Bottom line: Compared with other interfaces I've tested, the Octa-Capture has given me sufficiently low latency, with consistency at a listed buffer figure of 64 samples, using real-world projects, to allow for real-time monitoring while recording. What else matters?




Now you see this is where you are failing to grasp certain aspects.

Most interfaces will give some usability at 064, but for those that require extended performance at those latencies then it matters a great deal how the respective interfaces perform, also when using double buffering on playback , it increases the round trip latency sufficiently enough for it to be a problem for some , which is compounded even further when nudging up to the next level(s)

So your "Real World" , doesn't translate for all.


Quote:

My mission is to present a factually accurate overview of a product from a real-world perspective, to which potential users can relate, and which they can understand. I believe I have have accomplished that with my review, I stand by the information prseented unless and until there is a need to stand corrected, and make no apologies for the credibility of the review, or for that matter, any other reviews or articles that I write.




Well my mission is to tackle areas where most regular reviewers fear to tread. :-)

Re credibility of your review, Wow Man, I made a passing comment that I "personally" take what you reported with a grain of salt ( as I do all mainstream reviews ) and you felt warranted to go into this long winded rebuttal .

You need to Chillax.. :-)

Quote:

(PS - I am enough of a gentleman not to mention a different web site in Sound on Sound's own web site, but I think I've presented enough information that anyone could find the review in a minute or two spent with Uncle Google.)




You've lost me, what are you on about ?

V:


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Anderton



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931312 - 30/07/11 01:06 AM
Actually, what you said was "Personally, after reading that I'd take anything the reviewer reports with a grain of salt.., or 3... :-)"

"Anything?" Putting "review" in quotes earlier in your post to imply it somehow wasn't a real review? I thought your meaning was quite clear, but I do always allow for the possibility that trying to derive meaning solely from text, which may have been written in haste or without adequate forethought, is not always 100% accurate.

However, I think your more recent comment about my review, regarding how you can place "mainstream reviews" "in context," also speaks for itself.

The review you're referring to is anything but mainstream, which is why I still don't think you grasp the concept. It is open to all to make comments. It is part blog, part forum, part 24/7 fact-check. There is nothing else like this review format in print or, to the best of my knowledge, on the web so for you to dismiss it as a "mainstream" review you can put "in context" is at best, puzzling.

Think of a pro review as similar to open source code, where someone creates a structure and others build upon that structure. You can go in there and say you think the Octa-Capture causes brain damage and has a latency measured in hours if you want. Of course, people will think you're an idiot for saying something like that, but you have the complete freedom to say it. No post in a pro review has ever been deleted in order to be politically correct. In fact if you want to see a real bloodbath, I invite you to check out the pro review of the TC Konnekt.

I did not denigrate your efforts in any way, I merely explained my methodology and priorities in light of your saying you personally would "take anything the reviewer reports with a grain of salt.., or 3... :-)." I don't "fear" to go anywhere, but I may or may not have the resources to go everywhere I want to go. A Pro Review invites those who have additional resources to contribute. I would certainly welcome any test results you, or anyone else, can contribute on the Roland Octa-Capture. All we have is anecdotal evidence from people who have the unit, compared it to other interfaces, and presented their own conclusions regarding the drivers and latency.

So it appears it all comes down to a difference of opinion - I think 10ms of latency "is pretty darn good" (you'll note I didn't say "excellent," or "best in class," or anything that would imply it's superior to anything out there), and you don't. Fine. Not sharing your opinion doesn't invalidate "anything" I say.

As to the very last sentence you didn't understand, I was simply saying that I wasn't going to link to a (possibly) competing web site while in the SOS forums. It's just a matter of courtesy. It has nothing to do with you, but with explaining why I did not link to the review so people could see for themselves what a non-mainstream review looks like in its true context.

I urge you to carry on with your work as I do find it interesting, and hope you will contribute your findings to the pro review. I also hope you understand that calling into question "anything" I report in what you refer to as a "review" (in quotes) is not exactly something I felt could go unanswered. Perhaps you did not intend to be dismissive and/or derogatory, but personally, I could see no other way to interpret the text you wrote, which is why I felt I needed to respond.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931321 - 30/07/11 03:39 AM
You are making waaay to much out of this.. !!

Lets put this into context and into perspective.

What I stated was in light of your comments that 10 ms round trip latency is pretty darn good, when it is actually nowhere near good, IMO , so personally after reading that - I said I will take anything with a grain of salt.

There was no further intention past that.

I still stand by that , especially in regards to LLP after your added commentary here, as we definitely are not on the same page.

Re my comments re placing reviews and reviewers in context , LOL, thats a can or worms you may not want to open - I don't see your "review" being any different to the multitude of others in regards to how you approached the low latency performance aspect , or should I say didn't approach it. I may pop on your respective thread if/when I get a unit to test and have something concrete to offer , until then I see no point , nor do I see the point of you getting so worked up about me stating an opinion.

Re the last misunderstanding , I admit I missed that in the mix, Mate it was already mentioned it was Harmony Central, its not a big deal , I seriously doubt Martin or the other admins here would care if you linked to it , especially since you are a contributor to SOS as well.

Lets get back to doing what we do best..

V.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931332 - 30/07/11 06:42 AM
Just checked in on your Octa-Capture thread..,

I'll stand corrected about the Roland using double buffering.

From your last post and screen cap -



Roland Octa-Capture:

Buffer Setting : 064 Samples - Nominal 1.5 ms

Input Latency - 5.1 ms (227 samples )

Output latency - 4.3 ms ( 190 samples )

Round Trip Latency - 9.5 ms ( 417 samples )

You don't see something screwy there ?

There is an added 168 sample buffer on Input and an added 126 sample buffer on Output , now of course that also includes the AD/DA inherent sample buffers, but those figures are exceptionally high.

In comparison the new baseline is the RME HDSPe AES/ADI8QS combo which interestingly only reports the driver latency minus the AD/DA to the ASIO host despite being from the same manufacturer , but I managed to calculate the AD/DA inherent delays and the correct RTL with my Utility

064 : Reported I/O - 1.519 / 2.245 : Calculated RTL - 191 Samples : 4.3991

There is a 63 sample buffer above nominal , that includes the AD/DA, an added 32 samples on playback + a few samples for the FPGA - 064 RTL is 191 samples total / 4.3391 , so the AD/DA are about 14 samples respectively.

Now I admit that is an exceptionally good 064 RTL , but even the RME Babyface which is also a USB2 interface has an I/O of 2.472 / 3.243 and RTL of 5.578.

9.5 ms is far from being on the same playing field and on those figures the unit would do extremely poorly on my LLP Rating simply due to the extreme safety buffers being employed.

V:


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Anderton]
      #931395 - 30/07/11 11:23 AM
Hi Craig, and welcome to the SOS Forums!

Haven’t seen you for ages - hope you’re well and thriving, and thanks for posting in such detail here

Don’t get too wound up about Vin - his heart’s most definitely in the right place, and his measurements nearly always spot on.

‘Low latency’ is indeed an emotive subject that stirs up a lot of arguments - in my experience many musicians are attempting to run their audio interfaces with buffer sizes of 64 samples to get a real world round trip latency of 5mS or less at 44.1kHz (A/D + 1.5mS buffer in + 1.5mS buffer out + D/A). Some even work at 96kHz primarily to drop this even further.

Personally like you I’m happy working with a RTL of around 10mS for keyboard-controlled softsynth playback, but a lot of drummers tell me they demand tighter timing for pad work, while those wanting to listen to incoming performances through cans really struggle above 5mS.

In fact I recently started a thread on exactly this topic to gather other musician’s opinions entitled ‘What audio interface buffer size do you use, and why?’. Here it is if you fancy a quick read:

www.soundonsound.com/forum/showflat.php?Cat=&Number=919292

Currently the results are reasonably evenly split between 64, 128 and 256 samples, although some musicians have said they would move to lower buffer sizes if their audio interfaces didn’t prevent it by being so inefficient at lower latencies and clobbering their plug-in capabilities

In the past plenty of studios accepted the much higher overheads of running at 96kHz and 64 (or even 32) samples so that their clients got the ‘monitor mixes with plug-in effects’ they demanded. However, Vin’s work identifying the sometimes HUGE variations in performance between audio interfaces at buffer size below 256 samples (100% or more in some cases ) has highlighted the fact that choosing an audio interface for low latency work can currently be somewhat of a lottery.

I agree with you that many people are quite happy with a RTL of 10mS. However, others aren’t, so I welcome all discussion on the subject. As you’ve said “it appears to come down to a difference of opinion”. Spot on, but this difference can get quite heated at times, as I know to my cost

As Vin points out above from your screenshot, the Octa-Capture does appear to add plenty more latency to the total than many other interfaces do, which could seriously annoy some potential purchasers (trust me - it’s a really sore point with some of them )

It’s great to see you here by the way!


Martin

--------------------
YewTreeMagic


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931516 - 30/07/11 10:09 PM
* Just to clarify on standing corrected on the Roland using double buffering , I originally thought it may have used it only on playback which is common on some other units, instead it is actually using extended safety buffers on both Input and Output *

Moving on,

As mentioned in a previous post , the new baseline is the RME HDSPe AES/ADI8QS combo , which returned identical results across the respective benchmarks , but has significantly lower I/O and RTL due to the tighter AD/DA . Proving the drivers and performance are identical for the HDSPe cards , RTL values will differ depending on AD/DA.

With the new baseline I had to recalculate all of the cards RTL /LLP ratings , I also dropped any card reporting nominal which I have not yet managed to calculate the RTL via the utility, to a pending status (following a suggestion by John Roberts)

I have an RME UFX which I need to test on both USB2 and FW as well as an Echo Audiofire12 to add to the growing list.

I'll post the updated results as soon as I find the time to do the testing.

Peace

V:


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931526 - 30/07/11 10:57 PM
Hi,

As the person who posted the request for latency figures on the Harmony Central forum, I can't help feeling some slight sense of embarassment at the hornets nest it appeared to stir up.

This was certainly not my intention. Likewise, if I'd thought for one moment that referring to another forum might be regarded as "ungentlemanly" I wouldn't have done it. None of the forums I routinely participate in operate such an embargo, either explicitly or implicitly (there are countless references to HC - and many other such sites - all over this forum, for example), so it simply never occured to me that one might.

I have edited my post at HC accordingly. I do think, however, that Harmony Central should really consider including it in the terms of use, to prevent such inadvertent faux pas in future.

Having said all that, my view is that Vin's perspective and Craig's perspective are equally valid. It would be nice to think that the coming-together of these two cultures could be a meeting rather than a clash, and that some good might come of it. With that in mind:

Vin - you're looking for a reliable way of measuring RTL. I personally can't see much wrong with Craig's simple yet elegant solution. Results obtained via this method strike me as very clear and unambiguous - and most importantly, immune to reporting 'trickery'.

You could set up an extra channel (per buffer setting) in the DAWbench project so that the RTL test recordings (of one of the DAWbench audio tracks) can take place whilst running the benchmark. This way the RTL measurement is directly integrated into - ie, adds functionality to - the benchmark.

Perhaps you could automate a drop-in on the appropriate channel(s) so that the test measurement is always taken at the same point in the benchmark run. Using this method would also allow the testing of discrete A/D D/A converters, by comparing the digital-to-digital RTL of digital-only cards with the RTL of the same card through various converters.

Craig - I disagree with Vin entirely with regard to your review of the Octa-Capture. I do think, however, that adding the latency measurement turned an excellent review into an indispensable one.

http://acapella.harmony-central.com/showthread.php?2746804-Roland-Octa-Cap ture-Interface-Now-with-Conclusions

Regards,

John


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931530 - 30/07/11 11:29 PM
John,

Firstly you should feel no responsibility for anything that transpired and secondly , why would you feel it necessary to delete the link over at HC to the conversation that transpired here.

I think its important that the guys reading in over at HC actually see the dialogue that transpired as its important to where each of us stand. I am not interested in posting over at HC btw.

Re measuring the RTL, I have a Utility that is delivering results within a few samples , I will make that available when the developer gives me the go ahead to offer the final build.

Re disagreeing with me in regards to the review , more power to you, but lets be clear , my qualm is purely on the discussion of latency and the ongoing stance that 10ms (9.5ms) RTL is actually acceptable at 064. So out of interest , does that mean you also disagree that 9.5ms RTL at 064 samples is excessively high comparatively to other units ?

V:


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931533 - 30/07/11 11:47 PM
Hi Vin,

No - I only meant generally. I agree with you entirely about the latency. But while it's a vitally important metric, it's obviously not the only one - and I thought Craig Anderton covered the other stuff very well.

After all, you could have an i/f with exemplary DAWbench stats and terrible sound quality, build quality, frequency response, SNR etc, etc.

I'm fascinated by your new utility. What's the theory behind it?

Regards,

John


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931536 - 30/07/11 11:52 PM
Quote TAFKAT:

So out of interest , does that mean you also disagree that 9.5ms RTL at 064 samples is excessively high comparatively to other units ?
V:




That's the point - I don't know, because there is doubt over the validity of the data (to my mind).

That's what appealed to me about Craig's method. It's obviously valid. I'd like to see all of the units tested the same way.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: John Roberts]
      #931540 - 31/07/11 12:04 AM
Quote John Roberts:

Quote TAFKAT:

So out of interest , does that mean you also disagree that 9.5ms RTL at 064 samples is excessively high comparatively to other units ?
V:




That's the point - I don't know, because there is doubt over the validity of the data (to my mind).

That's what appealed to me about Craig's method. It's obviously valid. I'd like to see all of the units tested the same way.




Hey John,

But the reported latencies via the ASIO driver were accurate, all Craigs method did was confirm the reported values.I have found in all of the interfaces ( that don't report nominal ) that I have double checked so far with the utility, to be reporting accurately as well.

Re the Utility , the methodology is similar to the Centrance utility where you connect a physical cable to an I/O and it measures the RTL latency.

Not any different to what Craig did, but more convenient.

Send me a PM with some contact details, I'll let you have a play.. :-)

V:


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931545 - 31/07/11 12:27 AM
PM sent.

What did the utility say about the i/f's that DO report nominal?


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: John Roberts]
      #931549 - 31/07/11 12:57 AM
Hey John,

I noted in an earlier post but you may have missed it.

Both the ESI's are using double playback buffering.. :-(

Their ratings have dropped accordingly... :-)

V:


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John Roberts



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931551 - 31/07/11 01:25 AM
Hi Vin,

You're right . . . I missed that.

Are we talking about a double RAM buffer, or additional buffering in the interface hardware?

On a connected note, I did notice in Craig Anderton's screenshot that there is a menu item labelled 'buffers in playback queue', which is set to two.

Cheers,

John


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931554 - 31/07/11 02:06 AM
Hey John,

I suspect the added buffering on the ESI units is the driver itself , its hard to tell with just the calculated RTL value whether its added to both I / O or just the Output , but the RTL is quite high.

It took some work to get the results as well, whereas on the RME units it was relatively easy. Having another look at the results they are exceptionally high , I doubt the AD/DA's / FPGA/DSP could account for such a measurable increase. I do have to revisit them again before posting the results , but needless to say they are close to the bottom of the table listing now.

Re the "Buffers In Playback Queue" setting in Sonar, from my understanding that is the amount of buffers that Sonar reads ahead before playback, it has no correlation to the actual buffer size/ latency, but can be used to smooth out playback.

V:


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #931700 - 01/08/11 04:57 AM
Just to add a few more curves into the mix..,

Doing some double checking of the data I suspect the MOTU 424/24 I/O and RTL values may only be for the driver/safety buffers and DSP, excluding the AD/DA. I am having that checked for me by a client with the MOTU unit as I no longer have it in my possession.

The fun continues.. :-)

V:


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #932817 - 06/08/11 10:25 PM
Quick update,

As suspected the MOTU was only reporting the driver/safety buffers sans AD/DA , which added another 98 samples to the RTL across the board at each buffer setting.

I have completed the UFX testing with some interesting and rather pleasing results on the FW side considering that the unit no longer has a dedicated FW controller , simply uses the FPGA for all on the control/arbitration now. Slightly better performance than the FW800 with lower I/O and RTL due to the tighter AD/DA.

I am just holding off posting the results for the time being as I need to confirm one last variable which is the new MultiClient Mixing option on the USB side, that I need to cross reference against a driver that does not have that feature. Of course I skipped a couple of driver versions so just waiting on a copy of the last revision of the driver sans the MCM. I had the appropriate version for the FW side and it is the same version I had tested the FW800 with, so all good there. I am told by M.C from RME that the next version of the FW/USB drivers will have the available option check box the same as the HDSPe drivers.

Also in light of the recent discovery on interface drivers sometimes not reporting AD/DA , I need to go back and double check the AVID/M Audio FW and USB units that are reporting identical values for I/O , which is a little suspicious.

What have I got myself into ... LOL

Stay Tuned

V:


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Dishpan



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #932902 - 07/08/11 09:42 PM
>Also in light of the recent discovery on interface drivers sometimes not reporting AD/DA , I need to go back and double check the AVID/M Audio FW and USB units that are reporting identical values for I/O , which is a little suspicious.

A recent discovery? This has ALWAYS been the case for some models! I did similar research on this years ago and hardly any of the interfaces then reported convertor latency. I resorted to brute-force testing to get precise figures. Remember it's not just AD/DA either, certain cards (such as MADI etc.) can add extra samples for conversions to these protocols (although to be fair this isn't usually significant) and again, this is often NOT reported so ensure you include this too!. So you have that plus buffer plus safety to contend with! It really is a minefield


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Dishpan]
      #932908 - 07/08/11 10:37 PM
Quote Dishpan:



A recent discovery? This has ALWAYS been the case for some models! I did similar research on this years ago and hardly any of the interfaces then reported convertor latency.




Care to share that data ?

BTW - I didn't mean it as some global revelation, it was meant from a personal perspective only.

Also I was quite familiar that when using external AD/DA's via ADAT/AES/MADI in some cases the AD/DA were ignored , especially when mixing and matching cards/converters from different manufacturers. I was a little surprised that the RME HDSP AES/ADIQS combo also didn't report the AD/DA tho.

FWIW: the Lynx AES/ Auroro combo I am told do report the AD/DA latencies via the ASIO driver to the Host via the standard AES connection.

I was more surprised by the fact that the MOTU units that use a proprietary protocol would also ignore the AD/DA and will be even more disappointed if the AVID/M-Audio are also ignoring the onboard AD/DA.

My comment needs to be taken in that context.

Quote:

Remember it's not just AD/DA either, certain cards (such as MADI etc.) can add extra samples for conversions to these protocols (although to be fair this isn't usually significant) and again, this is often NOT reported so ensure you include this too!. So you have that plus buffer plus safety to contend with! It really is a minefield




The RME units with the onboard AD/DA that I double checked with the RTL Utility report all of the latencies from my calcs via the ASIO driver to the hosts, that includes the safety buffers/FGPA as well , the AES card reported all minus the AD/DA.

The further I get into this the more I realise I have opened a can of worms of huge proportions, a minefield it is , but one I am right in the middle of carefully navigating as best I can. It is also harder when I am doing this on my own and against the grain of some of the manufacturers as well , who would be less then ecstatic about the laundry being aired , but there is no turning back now... :-)

V:


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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #932920 - 08/08/11 12:51 AM
I think it is great that you are independently doing these measurements. From the perspective of live recording of percussion and other rhythmic instruments, measuring and minimizing latency is an important issue.
Your methodology does seem to be getting more complicated, and my naive question is, why not measure a simple total round trip latency directly? For example, record a rim shot or some other short sound with steep initial slope of the waveform, play it back through a headphone with your measurement mic inside the phones while recording the played back sound, then read the total round trip latency directly off of the time axis of the recorded waveform.
It may not answer all questions for those who use synthesizers, midi, etc, but I don't see any potential sources of significant errors with that method (which replicates a live recording situation), and it would seem to be reproducible and usable unmodified with any interface and hardware/software setup. Perhaps you are doing this already and I just don't know how to interpret your reported data. Am I missing something?

Edited by Jorge (08/08/11 12:55 AM)


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Jorge]
      #932996 - 08/08/11 12:19 PM
Quote Jorge:

I think it is great that you are independently doing these measurements. From the perspective of live recording of percussion and other rhythmic instruments, measuring and minimizing latency is an important issue.
Your methodology does seem to be getting more complicated, and my naive question is, why not measure a simple total round trip latency directly? For example, record a rim shot or some other short sound with steep initial slope of the waveform, play it back through a headphone with your measurement mic inside the phones while recording the played back sound, then read the total round trip latency directly off of the time axis of the recorded waveform.




Hi Jorge!

Vin is effectively doing this, but creating a dedicated utility makes things even quicker and less prone to user error.

For instance, I’ve measured a load of interfaces using the ‘record the click and look at the time difference’, and once you’ve introduced a sequencer for the recording you also have to double check that any auto-quantise is disabled, as well as any other latency compensation tweakery.

With a dedicated utility everyone can measure EXACTLY the same thing for themselves without worrying about any of this stuff


Martin

--------------------
YewTreeMagic


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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #933002 - 08/08/11 12:52 PM
Thanks, Martin. It does seem that there are potential confounding factors I had not considered, and a utility that accounts for these and gives reproducible results would be useful. The comparative results for a range of popular interfaces tested under identical conditions using such a utility would be very useful for end users like me. I can certainly see the potential for resistance from manufacturers. Vin, I give you great credit for dealing with that and to SOS for supporting a project like this on your forums.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933103 - 08/08/11 09:12 PM
Hey Jorge,

Martin beat me to the punch.. :-)

The Utility will be available to all once we settle on a final build and it will definitely be of benefit to many I believe.

IMO - the only manufacturers that will have resistance to this project are those that have something to hide. This could be of enormous benefit across the board for all involved if taken in a positive manner , but some are not really open to taking the data in that light, only seeing it as criticism. How they actually respond to the available data ( if they do at all ) is the real litmus test to their respective approach , lets see where the dust settles.




V:


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robinv



Joined: 31/08/04
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933378 - 10/08/11 08:36 AM
Quote TAFKAT:



IMO - the only manufacturers that will have resistance to this project are those that have something to hide. This could be of enormous benefit across the board for all involved if taken in a positive manner , but some are not really open to taking the data in that light, only seeing it as criticism. How they actually respond to the available data ( if they do at all ) is the real litmus test to their respective approach , lets see where the dust settles.




V:



For the sake of being a cheeky devils advocate - you are assuming that manufacturers have even heard of you and your little test or give a toss about your opinion or findings. You do talk as if they are hanging on your every word, desperate for your approval and that not to do so means they are evil people hoping to screw people out of latency while wringing their hands and cackling. You've already posted results and graphs and then had to change them which doesn't really do the seriousness of your claims any good (i have personal experience of this sort of thing). Also you are pinning everything on a single attribute. I don't think anyone would like their product compared and rated on a single factor. I wouldn't like to see a Rain computer put along side computers from every competitor and measured/judged purely on maximum number of plug-ins run because it doesn't take into account the whole environment in which that measurement exists - there are always a number of factors involved in choosing a product.

However, all us freaks on this forum are eagerly fascinated by what you're doing and all hoping it helps to improve the performance of interfaces in the future

--------------------
Molten Music Technology - Computers for doing music on


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #933448 - 10/08/11 01:57 PM
Quote robinv:


For the sake of being a cheeky devils advocate - you are assuming that manufacturers have even heard of you and your little test or give a toss about your opinion or findings. You do talk as if they are hanging on your every word, desperate for your approval and that not to do so means they are evil people hoping to screw people out of latency while wringing their hands and cackling.




I don't assume anything Mate, and as for whether they are aware of me or give a toss , thats two totally different things. I am pretty sure some are aware of my work over the years , whether they give a toss is another matter.

Hmm, maybe I should have a signature with a direct link to my websites to help them find me if need be... :-)

I can safely say I know of more than a few who respect the work and are very conscious of the comparative LLP performance of their units.

As to my little test , you are dismissing the volume of work I have assigned to this endeavour as little ?

As to the manufacturers hanging on my every word , I doubt it, nor do I care. I am simply reporting as the chips fall , whether they read in , ignore , dismiss or other wise isn't the primary concern. Those that take issue know where to find me, those that want to work with me know like wise, and those that choose to ignore me, well, more power to them :-)

Interestingly I do get some nice toys sent over to play with , qualify and report back warts and all. I have a healthy qualifying list of hardware on site at all times , it does keep things interesting and in perspective.

Quote:

You've already posted results and graphs and then had to change them which doesn't really do the seriousness of your claims any good (i have personal experience of this sort of thing).




So you are questioning the credibility of the data because I am being totally transparent and amending it as I progress further , wow that's interesting I have to say. Also the amendments haven't changed anything in regards to the bulk of the work, only those interfaces that either reported nominal or needed further clarification with the AD/DA values.

This is work in progress and I am amending the results as further variables come into play, how can you possibly be finding fault with that. I am publicly disclosing and amending any results that have a question mark over them. As to your personal experience , please don't compare my current contributions to your instance.

You posted blatantly skewed data that you never followed up on or retracted despite myself and numerous others pointing it out to you very early on , reasonably politely I might add. Then 12 months on you posted an amended report with results that were the total opposite of the previous report. The results coincided with data that I and others had previously presented you, there was absolutely no mention , apology or retraction of the previous report which was conveniently deleted.

I think our experiences differ there Mate, I am happy to eat crow if I have screwed up on any data presented , I didn't see the same courtesy from you. BTW: Its quite tasty baked with a touch of salt & lemon.. :-)

Quote:

Also you are pinning everything on a single attribute. I don't think anyone would like their product compared and rated on a single factor.




I made it abundantly clear right from the get go that I was focusing purely on the LLP aspect , the rating is pretty straight forward , its a Low Latency Performance Rating, nothing more, there are plenty of other reviewers doing the comparative reviews focussing on other attributes.

But as you suggested earlier, why would the manufacturers even care about my LLP rating ?

Quote:

I wouldn't like to see a Rain computer put along side computers from every competitor and measured/judged purely on maximum number of plug-ins run because it doesn't take into account the whole environment in which that measurement exists - there are always a number of factors involved in choosing a product.




Right, I bet you hated all the positive commentary from that last SOS review system of yours based on the benchmarks you are now suggesting are pointless. ;-)

Of course there are plenty of factors involved in choosing an appropriate solution , one variable that is extremely important is matching the end users respective working environment with an audio interface solution that will best suit that environment. Perfect example, a client working with large sample based VI's at low to moderate latencies will not be well served with an interface that does not perform as well as others at those required latencies. All of the bells whistles and branding mean squat if it fails in that one single variable !

Quote:

However, all us freaks on this forum are eagerly fascinated by what you're doing and all hoping it helps to improve the performance of interfaces in the future




I'm confused, you seem to be trying to have a little bit each way :-(

On one hand you dismiss the work as being irrelevant to the manufacturers , but on the other hand you are hoping it may result in better drivers in future ??

Oh I get it, what your are saying is that you don't mind someone getting their hands bloodied for the benefit of all, as long as their not yours... :-)

V:


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933614 - 11/08/11 09:11 AM
No you nut, i'm being "cheeky" for the sake of it - i'm poking a bit of fun, not making defining statements about you or your "little" test (that's meant as cheekiness, not sarcasm). I've said on here before that your work is brilliant, it's just sometimes it comes across a bit over-important and so that makes me want to tease you a little - it's just teasing mate. Your transparency is great but people do often take things at face value and don't read what was said and what i mean is that someone may take your results and buy a certain interface because you reported it as good only to find that you change the results later based upon new data. You're not measuring an actual latency at the moment you're calculating one. But none of that would be your fault, you're simply presenting your data. I'm not calling your results into question or disrespecting them in any way - all i'm saying is that caution is required.

Quote TAFKAT:



I don't assume anything Mate, and as for whether they are aware of me or give a toss , thats two totally different things. I am pretty sure some are aware of my work over the years , whether they give a toss is another matter.




Well, no, you said that if they didn't respond they must have something to hide so therefore you do assume they give a toss. And so i was teasing you about that assumption.


Quote:


This is work in progress and I am amending the results as further variables come into play, how can you possibly be finding fault with that. I am publicly disclosing and amending any results that have a question mark over them.



I'm not finding fault mate, i'm saying that people don't always read what you say and take many things to be gospel that perhaps weren't not intended as such - be cautious.

Quote:



I made it abundantly clear right from the get go that I was focusing purely on the LLP aspect , the rating is pretty straight forward , its a Low Latency Performance Rating, nothing more, there are plenty of other reviewers doing the comparative reviews focussing on other attributes.




Absolutely, but that's my point. You said that manufacturers would have something to hide if they didn't act on this single attribute whereas i'm just saying that i think it's a bigger picture than just that - for some people this is the only thing they care about and that's fine. It's the assumption you made about the manufacturers that i was poking fun at

Quote:



Right, I bet you hated all the positive commentary from that last SOS review system of yours based on the benchmarks you are now suggesting are pointless. ;-)



Oh it was so awful i almost left the country.

Quote:

Quote:

However, all us freaks on this forum are eagerly fascinated by what you're doing and all hoping it helps to improve the performance of interfaces in the future




I'm confused, you seem to be trying to have a little bit each way :-(



Both ways absolutely - i love your work man, i'm just poking a bit of fun at you, and y ou know that i have different opinions on these things and so i present a slightly different view - not to knock yours, just to offer some more flavours.

I don't dismiss your work in the slightest - i didn't ever say that and i am always happy to contribute to the discussion and to the work.

Obviously my post and teasing was misjudged and so i'm sorry about that Vin, didn't mean to rub you up the wrong way - unless it was funny, which it seems it wasn't.

You keep doing what you're doing and feel free to ignore me

--------------------
Molten Music Technology - Computers for doing music on


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #933640 - 11/08/11 10:55 AM
Quote robinv:

No you nut, i'm being "cheeky" for the sake of it - i'm poking a bit of fun, not making defining statements about you or your "little" test (that's meant as cheekiness, not sarcasm).




Sorry Mate, I totally misinterpreted, sometimes the intent is lost due to the non verbals that are the nature of the cyber beast... :-(

Quote:

Your transparency is great but people do often take things at face value and don't read what was said and what i mean is that someone may take your results and buy a certain interface because you reported it as good only to find that you change the results later based upon new data.




I hear ya, and its why I immediately came back and retracted my earlier praise of the ESI units when I discovered they were hiding huge safety and playback buffers by only reporting nominal. It has made me extremely cautious as I go forward with any and all interfaces now.

Quote:

You're not measuring an actual latency at the moment you're calculating one. But none of that would be your fault, you're simply presenting your data. I'm not calling your results into question or disrespecting them in any way - all i'm saying is that caution is required.




The ASIO driver protocol has the ability to report very accurately to the Host , something that Craig Anderton discovered when he went thru the manual exercise to find that he was within a few samples. The Utility being developed is using the same principle but in a more convenient manner which will hopefully minimise discrepancies that can be introduced trying to zoom in , select and measure manually.

Quote:


Well, no, you said that if they didn't respond they must have something to hide so therefore you do assume they give a toss. And so i was teasing you about that assumption




Ahhh yes assumptions are definitely a dangerous thing, but you are assuming that I was assuming, when actually I wasn't assuming anything at all .. LOL

What I said was that IMO the only manufacturers who would feel resistance to what I am doing would be those with something to hide , that could be anything from less then stellar driver efficiency to hiding extended safety and playback buffers. Thats not really an assumption its simply an opinion based on some behind the scenes communique. I am hoping to lift the veil as much as possible so that we are all more informed.

I know this is extremely sensitive for some parties dealing with the 3rd party OEM controllers being highlighted here , but the simple fact that all are dealing with the same deck , but some are performing substantially better than others. That alone should be enough for those not performing as well to look into why the variable exists at their end and improve on it, well at least I hope it would.

Quote:

Obviously my post and teasing was misjudged and so i'm sorry about that Vin, didn't mean to rub you up the wrong way - unless it was funny, which it seems it wasn't.




Its cool Mate, I usually pick up the intent on your posts pretty well, unfortunately I was reading the last one a little too diagonally , lets put it down to the coming full moon... :-)

All Good and thanks for responding in the manner you have.

I'll try and complete the testing on the current batch and get the amended charts/graphs up ASAP

Peace.

V:


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933667 - 11/08/11 12:42 PM
Quote TAFKAT:



Ahhh yes assumptions are definitely a dangerous thing, but you are assuming that I was assuming, when actually I wasn't assuming anything at all .. LOL






Quote:


I'll try and complete the testing on the current batch and get the amended charts/graphs up ASAP

Peace.

V:




We're all looking forward to that
And remember i like graphs so that i don't have to actually read anything.

Cool mate.

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933760 - 11/08/11 10:27 PM
So we won't need a group hug after all


Martin

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933791 - 12/08/11 03:31 AM
Ahhh what the he.., why not



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vinark



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #933857 - 12/08/11 11:23 AM
Hey everybody,
I'm one of the people you write your reviews for (and a hopefully helpfull member of Dawbench, hey Vin, interesting discussion here). From my perspective, I have to agree 100% with Vin (Tafkat). LLP is an absolute, not a perception at all. It may not be easy to measure and quantise, but it is a value anyway. Just like for example mic preamp noise, what would "it is rather silent" mean exactly, I need figures and maybe a warning like yes the figures are low but it's an awfull digital kind of noise....
Without Vins help choosing an audio interface for me as an in the box composer is a complete nightmare. You spend lots of money on a fast machine and might achieve nothing but flawless iTunes playback .
Please reviewers don't underestimate the importance of LLP. I would guess it is the biggest difference between products apart from features (like how many in and outputs etc). Much more so then sound quality, which is much more related to price/budget IMHO.
BTW I own a HDSP9652 with ADI-8AE. Super performance....Great sound (yes these are absolutes )

--------------------
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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: vinark]
      #934004 - 13/08/11 12:09 AM
Hi vinark, and welcome to the SOS Forums!

Thanks for your comments. I totally agree about the LLP test, and have been measuring real-world latency myself during many SOS audio interface reviews over the years (and been shocked by all the ‘hidden extras’ found on some models )

Many musicians don’t seem too bothered about low latency performance except to complain that their computer can’t do it successfully, without realising that it may be their choice of audio interface to blame rather than the computer


Martin

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Hardwinte



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #934010 - 13/08/11 12:56 AM
I would just like to thank Vin for his work so far it has been very helpful for me in looking for an interface/soundcard for my itb work, and I'm sure it will be helpful to many like me who are beginners and are looking to compose with mainly virtual instruments and samples.

However, I settled on a soundcard outside your list - the perfect M-audio delta 66 (perfect for me), enough outs for multiple references, enough ins for a few synths and s/pdif! The latency on this thing is very good (anecdotal evidence suggests RTL below 5ms across various forums), it is stable, and the only thing not nice for me is the budget-level dac.

Your research saved me some time in looking for a low-latency gear, thankyou!


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Hardwinte]
      #934289 - 15/08/11 12:00 AM
Hi Hardwinte, and welcome to the SOS Forums!

The Delta 66 has provided sterling service to a host of musicians over the years - I remember reviewing it for SOS way back in 2000. The audio quality of its AK4524 Codec chips has since been surpassed by quite a few other models, but it remains a budget workhorse for a lot of people.


Martin

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Hardwinte



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #934384 - 15/08/11 12:13 PM
Thanks Martin Walker

Funny side thing, but my previous english teacher's name was Mr Walker. (The **** actually kept on walking out of class and only turning up for half of our lessons... )

Not that he reflects on you at all, by the mere chance that you share a surname! No good sir, no, but if you could kindly talk some more about those better codecs I would appreciate it, made me all curious as I haven't actually bought my interface yet (no time, going through exams)


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Hardwinte]
      #934451 - 15/08/11 03:01 PM
Well I don't want to hijack this thread - let me just say that the Delta 66 is a great interface, but was brought out ten years ago now, so newer models are likely to have more recent technology that might possibly have slightly better audio quality.


Martin

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YewTreeMagic


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #934536 - 15/08/11 09:01 PM
Hey All,

Thanks for the support and ongoing interest.. :-)

Just a few notes , re the Delta 66 ( or any PCI interface ), with the current Intel chipsets moving away from a legacy PCI buss unfortunately they will be less of a focus moving forward. My X58 test/development system luckily still has native PCI so I can achieve full performance on the older cards that are compatible, but current systems - P67/H67/Z68 on may not achieve full performance with PCI cards due to the 3rd party bridged PCI implementation.

It would be great if M-Audio updated the Delta range to PCIe as they have always been decent performers , I would also add Lynx to that list with their Lynx 2 range , but unfortunately it seems that most manufacturers are focusing on external connectivity as that gives a wider spread across both desktop and notebook markets. However its not all smooth sailing as witnessed with the variance with the FW controllers and now with the growing trend of manufacturers moving more and more to USB2 ( supposedly for greater compatibility ), its becoming even more of a crap shoot. Perfect case in point is the issue I encountered with the ESI U46XL that simply refused to enumerate and function correctly on H67/P67 , and the SSL Nuclues unit that was extremely inconsistent depending on the system and even USB port. I am also keeping a close eye on numerous reports of issues with other USB2 interfaces also being inconsistent.

Focusrite which has had a negative focus on my recent testing with the performance of the FW interfaces has recently introduced USB2 interfaces ( as has Presonus) , and already there are reports of inconsistencies and instability posted on this forum Here

Also take note of the reported latencies :-(

If any of the other DAW builders or end users with one of these new Focusrite units want to chip in with some testing/reports that would be great, not sure if/when I will have access to one.

I am currently buried with work unfortunately but I will drop back with some fresh results ASAP, I even dusted off an old Lynx2 PCI card to do some testing after a test loaner PCIe config fell thru , that should give a good indication at least of the current driver - AD/DA will not be as tight tho.

Stay Tuned

V:


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #934652 - 16/08/11 01:17 PM
Quote Martin Walker:

Well I don't want to hijack this thread - let me just say that the Delta 66 is a great interface, but was brought out ten years ago now, so newer models are likely to have more recent technology that might possibly have slightly better audio quality.


Martin



I'm still recording through my Delta 44/Omni io combination that i bought when they were released. In terms of features i haven't been able to find anything to replace it with. Still sounds ok to me

--------------------
Molten Music Technology - Computers for doing music on


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Skyline
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #934761 - 16/08/11 11:16 PM
Quote Martin Walker:

Well I don't want to hijack this thread - let me just say that the Delta 66 is a great interface, but was brought out ten years ago now, so newer models are likely to have more recent technology that might possibly have slightly better audio quality.


Martin




Care to name those 'newer models' Martin, just for reference?

I used a Delta 66 for a long while but had fairly frequent 'fizziness/crackles' when I recorded sometimes. It's not an uncommon issue with the Delta range. I wasted a lot of time on the MAudio forums, and others, searching for the Holy Grail solution, including trying arcane tweaks like changing power setups in the BIOS screen, etc. All to no avail.

I bought a new Sandy Bridge DAW in April and bought a Focusrite Saffire PRO14 for a change. I haven't experienced any of the 'Delta fizzies' and am fairly pleased with it. Then I read your recent SOS piece saying that Firewire isn't all it's cracked up to be and an internal card (a la Delta) is probably better! Do I assume from your comment above that as the Pro14 is much newer than the Delta 66 I nevertheless have some advantage?

John

--------------------
When I'm sad I sing, and then the whole world is sad with me.
Band / Songs


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Skyline]
      #934768 - 17/08/11 12:17 AM
Quote Skyline:

Quote Martin Walker:

Well I don't want to hijack this thread - let me just say that the Delta 66 is a great interface, but was brought out ten years ago now, so newer models are likely to have more recent technology that might possibly have slightly better audio quality.


Martin




Care to name those 'newer models' Martin, just for reference?




Well let me just say that in terms of audio quality alone I would personally rate the Saffire and Konnekt ranges highly, although as we've been discussing here latency may be another matter to consider.


Martin

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Tombot



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #935059 - 18/08/11 08:41 AM
Hi Vin, have been testing a few units including the fore-mentioned Focusrite.
What do we do when the buffer size is in ms instead of samples?

Also, i'v e done head to head sound quality tests on a few units and can report minimal difference at 48k, at 96k it starts to get a bit different, but it pales into insignificance compared to choice of monitor speakers. The differences between the Mackie hr824, Neumann kh120 and Genelec 8030 are probably 50 times greater than between any of the interfaces ive tried so far.


Edited by Tombot (18/08/11 08:49 AM)


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ef37a



Joined: 29/05/06
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #935077 - 18/08/11 09:52 AM
"It would be great if M-Audio updated the Delta range to PCIe as they have always been decent performers"
It would indeed TAFKAT but it isn't going to happen. I have had stirling service from my 2496's. They have worked flawlessly in XP and W7/64 and even with WMCE which is supposed to be a no-no!

So when I got my W7 PC my aim was to buy a pair of AP 192s to get 4 in 4 out with what were reckoned to be excellent new converters (Martin W said so!) I was crushed then to find my new PC had but one PCI slot. Ah well,I wil just wait for the PCIe versions and dropped an E to M-A asking about when. Never was the reply. They are not and will not be upgrading any soundcards to PCIe format, out of MOBO sound altogether.

Bloody outrageaous and commercially stupid IMHO.

Dave.


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Tombot]
      #935243 - 18/08/11 10:32 PM
Quote Tombot:

Hi Vin, have been testing a few units including the fore-mentioned Focusrite.
What do we do when the buffer size is in ms instead of samples?




Hi Tombot!

You can convert farily easily.

For instance, at a sample rate of 44.1kHz, 1ms = 44 samples, at 96kHz, 1mS = 96 samples

As for your comments about audio quality, loudspeakers certainly vary a hell of a lot more than interfaces, but they can still be heard, particularly once you've got some acoustic treatment in place and some decent loudspeakers in place.


Martin

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YewTreeMagic


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: ef37a]
      #935245 - 18/08/11 10:35 PM
Quote ef37a:

"It would be great if M-Audio updated the Delta range to PCIe as they have always been decent performers"
It would indeed TAFKAT but it isn't going to happen. I have had stirling service from my 2496's. They have worked flawlessly in XP and W7/64 and even with WMCE which is supposed to be a no-no!

So when I got my W7 PC my aim was to buy a pair of AP 192s to get 4 in 4 out with what were reckoned to be excellent new converters (Martin W said so!) I was crushed then to find my new PC had but one PCI slot. Ah well,I wil just wait for the PCIe versions and dropped an E to M-A asking about when. Never was the reply. They are not and will not be upgrading any soundcards to PCIe format, out of MOBO sound altogether.

Bloody outrageaous and commercially stupid IMHO.

Dave.




Hi Dave,

PCIe just isn't as popular as USB2 and Firewire, despite all the extra hoops you have to jump through in many cases (controller chips, hidden safety buffers and so on).

This I suspect is largely because laptops have become so much more popular, and USB/Firewire is usable on both these and desktops, whereas PCIe cards have a much smaller market


Martin

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YewTreeMagic


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Tombot



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #935248 - 18/08/11 10:48 PM
Hi Martin
Should have explained a bit better, was a half asleep this morning!
I understand what to do with ms, but on that basis, if it is measured in ms, the intervals would be
44, 88, 132, 264, 528
I guess it can be argued that the higher buffers are quite close to 128,256 and 512 samples, but the 44 and 88 are completely in the middle of the 32 / 64 range.
Didn't want to post any numbers that could be determined to be unfair to a units measurements.
Have got a few units that seem to be working this way and wondered what should be the way forward?

--------------------
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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Tombot]
      #935381 - 19/08/11 01:49 PM
Quote Tombot:

Didn't want to post any numbers that could be determined to be unfair to a units measurements.
Have got a few units that seem to be working this way and wondered what should be the way forward?




Wait until Vin has perfected his real world latency utility, and then you can report the entire round trip latency of any interface, without any jiggery pokery of hidden safety buffers, missing converter latency, onboard SRC latency...


Martin

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Pete Kaine
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #935396 - 19/08/11 02:42 PM
Vin - If you've got code at a beta stage then errr... we'll apply for beta testing positions.

Can't have our Tom sitting around idle in his hovel

--------------------
ScanProAudio & 3XS Audio Systems
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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #935454 - 19/08/11 08:51 PM
Hey All,

@ Tom & Pete at Scan, no problem for you guys to test the utility in its current state, it is actually quite usable.

I'll contact you via email... :-)

Re the in between buffer settings , yeh that will be a tougher one to try and do head to head comparatives. I also had similar odd buffer sizes from the SSL Nucleus USB2 interface , but they were actually listed as Samples - 066/132/265/529 , judging by your upper 3 settings, I suspect it maybe the same OEM controller.

Maybe we can use the upper 3 as a cross reference against the Focusrite FW line , that will give us an idea.

Out of interest what values do Cubase/Reaper report for those respective latency settings ?

@ Dave,

As much as it disappoints me I can understand why most manufacturers will not develop PCIe interfaces , as I have noted in the past and Martin has just reiterated, its far easier for them to develop FW and USB2 interfaces to be able to cater for both the desktop and mobile markets. M-Audio are not the only manufacturers who have bypassed development on PCIe, Lynx who in the past have been on the top of the pile in regards to LLP with their Lynx 2 range haven't made the move to PCIe past the AES card either. Maybe with PCI no longer being native on the Intel chipsets , we may see some added incentive for more Lynx PCIe cards .., I can hope.. :-)

So I do understand the difficulty faced by a lot of manufacturers as its definitely a different market landscape and it wouldn't be a problem if we didn't have to navigate the current minefield of the inconsistency of the OEM controllers / drivers. Now some may say its more the fault of the controllers ( Dice ) being used, and the manufacturers can only play with the hand they are dealt. When I first started this endeavour I was also one under that impression, but the fact is this , all the current manufacturers have access to the same line of Dice chips , AVID / M-Audio perform exceptionally well , Focusrite, Presonus, etc do not , it can't be so easily dismissed that is purely the controller , the onus has to come back to the manufacturers doing more than simply supplying the units with the base OEM ASIO driver.

V:

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DanKaplans



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #935671 - 21/08/11 10:24 AM
I had the same problem.


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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #935674 - 21/08/11 10:38 AM
Hi Martin.
PCIe "just isn't as popular" because there just isn't any! ("No call for size 13's sir, you're the 11th person to ask today!)

I am a total pc numpty but maybe someone could tell me if it would been a big problem to have just transplanted an AP192 as is, onto a PCIe PCB?

Then: There are PCIe to PCI converter boards but the sellers don't know if they will work with PCIe soundcards and M-A won't test and tell. Something SoS might look into?

Dave.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #936025 - 22/08/11 10:17 PM
Hey All,

Here are the latest results with some added interfaces, a new baseline and amended ratings.





Notes -

First off we have a new Baseline Interface - RME HDSPe AES : ADI8-QS - results across the 3 respective benchmarks were identical to the previous baseline AIO card , proving that the performance of the driver is consistent across the HDSP/HDSPe line. The I/O and RTL was significantly lower due to the tighter AD/DA. A few points, the reported I/O values in Cubase were sans the AD/DA , ( common amongst interfaces when using AD/DA connected via standard AES ), I calculated the added latencies with my RTL Utility which reported an extra 28 samples across the board - 14 samples respectively for the AD/DA ( RME spec is listed as 12 ). The RTL Values are including the AD/DA and all other interface RTL ratios have been amended.

MOTU 424 : 24 I/O - I discovered was not reporting the AD/DA , so the correct RTL was calculated using the Utility and values amended.

LynxTwo C - I was hoping to test a current AES16e and Aurora combo but the loan units didn't eventuate so I dusted off an old PCI Lynx TWO C card to at least get a heads up on the performance of the current driver. Double Playback Buffering has been disabled ( still have no idea why Lynx thought it needed implementing ). The card performed extremely well only being bettered by the 2 PCIe RME units , its a shame that these PCI legacy cards are not suitable for the current Intel chipsets with the non native PCI, hopefully Lynx will eventually update the Lynx TWO line to PCIe as well.

RME Fireface UFX - This was a unit that I know a lot of end users are interested in seeing the results for and I was keen to get it back on the bench after my preliminary experience with a very early unit was less than stellar under FW. In the meantime there has been numerous firmware updates as well as drivers , but I actually used an earlier driver sans the MultiClient option to remain consistent with the earlier Fireface 800 testing. I was pleasantly surprised that the unit under FW had improved measurably to the point of it actually bettering the FF800 results. Thats quite an achievement IMO considering the unit no longer has a dedicated FW controller. USB2 performance was also exceptional , however slightly below the FW results , which may surprise a few people considering the greater focus has been on the USB2 aspect. All in all, a big thumbs up to the RME devs on ironing out the early quirks with the FW on this unit , an exceptional performer on both FW and USB2.

ESI U46 XL - The initial good impression I noted on the preliminary testing changed significantly when I discovered that ESI by reporting the nominal values only, were actually masking large safety buffers as well as double buffering on playback. The calculated RTL via the utility showed significantly higher values than most other interfaces at the respective latencies. RTL % and LLP ratings have been amended and have dropped accordingly. There are also issues with the unit not initialising correctly on current H67/P67 chipsets.

Echo AudioFire 12 - This unit has large safety buffers as well as double buffering on playback to the point that the nominal values reported in the Echo Control Panel are almost comical. I did attempt to get a confirmed reading via my own RTL Utility as well as the Centrance and the results were all over the place unfortunately. My Utility was having issue ithe the excessive safety and double buffering, the Centrance simply refused to give a result. Performance was solid enough but the high I/O and RTL need to be taken into account when weighing up the figures. The excessively high RTL dropped the over all LLP rating substantially, as will be the case with any interface employing large safety and double buffering.

I have also removed some interfaces from the results pending further testing.

I am currently further investigating some of the listed interfaces which I suspect may also not be reporting the AD/DA. Once I have confirmed those results I'll amend the LLP ratings accordingly. Interfaces under investigation - AVID/M-Audio/Focusrite/Midas.

Peace

Vin Curigliano

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Sian G



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #937350 - 30/08/11 04:05 AM
Hey peeps. I was directed to this thread by Martin in response to my thread. As it is there is far too much jargon for me to understand. What i have concluded is that at least 2 of the interfaces i was interested in are no longer on my shopping list. What i need to know in a nutshell before 12o'clock today is how best to spend my £400 to £500 for an interface? My requirements are fairly simple. I need to fully mic my drum kit to record it over pre recorded music, so 8 mic inputs i think are needed. Also i will eventually be recording guitars and bass and eventually some electronic/synth stuff albeit on a smaller scale than probs most of you folks. A midi i/o on the interface would be nice so i can use my old midi keyboard as a controller for any soft synth i may eventually get, though a Moog Slim Phatty is on my shopping list so i might not get the soft synth anyway. I'm very much a novice at all this and not really a very good musician, only competent as a drummer. I won't be filling my laptop with loads of virtual instrument software as i like to (try to) play real instruments. Any advice would be appreciated before i go and line the pockets of some saleman somewhere and take home something i don't want or need. As for my laptop, it's only running Vista x32 but i will have the use of my mothers Win7 x64 laptop if mine proves to be incapable of handling what i may throw at it. A Macbook Pro is in the pipeline though. Btw, what is the general concensus on the Roland Octa-capture then? That is just out of budget but i could stretch to it if it means i get the best interface for my money and needs.
Ta Sian.

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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Sian G]
      #937356 - 30/08/11 05:58 AM
Hi Sian.
Just my 2penn'oth but that Roland looks pretty good according to this..
http://www.musicradar.com/gear/all/computers-software/peripherals/input-de vices/audio-interfaces/octa-capture-362668/review (sorry SoS, don't think you have done it?)and it looks just under your budget?

One thing. I would try to get rid of Vista, you say you will eventualy go mac but XP would AFAIK be a better bet in the interim, the laptop MIGHT just be fine with the Roland but all evidence is against it.

Dave.


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The_BPP
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: ef37a]
      #937367 - 30/08/11 07:57 AM
I own the Roland Octa-Capture, and I'm very happy with it, the latency is certainly improved on my previous EMU 0404 USB, and it sounds better too.

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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: The_BPP]
      #937370 - 30/08/11 08:11 AM
Quote The_Big_Piano_Player:

I own the Roland Octa-Capture, and I'm very happy with it, the latency is certainly improved on my previous EMU 0404 USB, and it sounds better too.




That ^ is praise indeed FYI Sian since the 0404USB recieved high marks for the quality of its converters almost everywhere.

Dave.


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The_BPP
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: ef37a]
      #937722 - 31/08/11 03:08 PM
I actually upgraded to get more I/O, as I was very happy with the 0404's sound, but the Octa-Capture trumps it for quality, too.

Latency seems fine. People get a little pretentious about latency, sometimes, IMO. There's latency between the end of a drum stick, and the drummers ear.

Let's see - sound runs at around 343m per second, depending on conditions, and the length between a drummer's ear and the end of his stick is (for the sake of argument) a metre, with his arm stretched out, so that makes the latency nearly 3ms. That's 3ms before someone records it.

I haven't read the rest of the thread, but I'm presuming someone's already pointed this out.

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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: The_BPP]
      #937781 - 31/08/11 06:05 PM
Finite speed of sound in air is old news. Musicians learn to adjust for this by making their attack as coincident as possible with what they hear. A good listener can distinguish a single sound from a flam with 1-2 ms separating two sharp rapidly decaying impulses. Much more delay than that is disturbing to percussionists and causes us to try to compensate.

So in overdubbing some live instrument percussion tracks, if the delay between the already-recorded tracks and the newly recorded track after the RTL is more than 1-2 ms, some adjustment needs to be made. The RTL includes many component latencies in the computer sound/signal path that in simple terms is computer > DA conversion > headphone to ears/hands > played note > instrument to mic air delay > AD conversion > software recording to DAW. If the delay is a constant time interval, some DAWs can automatically adjust for that, or we can slide the new track a few ms earlier so it coincides with the rest without noticeable delay. If the delay is variable over time, due to the many changing processes going on inside the CPU and other processing systems, then it all becomes much harder and we need to minimize RT latency as much as possible. This is one of the reasons some of us musicians, especially percussionists, make a big deal of RTL. Singers and many other instrumentalists and electronic composers do not require as much precision in timing, so minimizing RTL is not a significant issue. I can understand your dismissing 3 ms latency since it is of a similar magnituded to other unavoidable delays, but please understand that your argument does not apply to some of us.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #937811 - 31/08/11 08:42 PM
Just to add to the last post, good post btw Jorge.. :-)

What also needs to be noted is that any additional latency imposed by distance is added onto the actual RTL of the audio interface , so I never really get the continual attempts at dismissal via the finite speed of sound argument. For those not needing to process the input and can make do with direct monitoring, then the listed total RTL information is not that important, but the actual inherent latencies of the AD/DA will come into play as some are nowhere near as tight as others. However that's only one side of the coin, the other 1/2 is actual performance of the interfaces at designated buffer settings. If the same end user is using lots of Virtual Instruments and are needing efficiency at the lower buffer settings then its extremely important.

I think the focus here is getting lost in the mix for some... :-(

V:

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Ishar



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #937941 - 01/09/11 10:40 AM
This is only partly to do with the title of the article. I'm just starting out in computer music and have been informed that I need an AI. From the article it seems that M Audio are good products. Has anybody used/ uses a M Audio Fast track or Fast Track USB and which is the better? Are there any other good AI's around the £100 - £150 mark? I am on a very limited buget, so I was considering the Behringer Guitar Link UCG 102 as a starting point,for now, for my guitar as it is much cheaper, and going for an AI later on for mic and/or keyboard. Is the Behringer Guitar link any good?
Thanks

Ishar

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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #938114 - 02/09/11 03:50 AM
Vin, will your utility measure RTL in a comparable way for inexpensive digital interface/mixer combinations such as the A&H Zed series or the Presonus StudioLive series? Some of us are considering using that format in studio for overdubbing and it would be good to compare with the straight interface units. My sense is that these are usually used for live sound where RTL is not really the critical parameter, and I suspect less engineering effort has been put into minimizing latency than for recording studio interfaces.


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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Jorge]
      #938117 - 02/09/11 05:12 AM
Jorge,
I have had some experience with the A&H ZED10 and there is a problem with the usb link and Win 7. A&H are aware of this and indeed very rapidly came up with a fix so if either yourself or others find the signal levels far too high with these unit talk to A's (or indeed myself).

Note too that the usb system is 16bit only, it is a very GOOD 16 bit link! But 16bit nonetheless.

Latency I cannot comment on since I just use the unit for tracking down to a laptop sometimes. Mostly the ZED feeds into a 2496 in two desktops.

Dave.


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Jorge]
      #938213 - 02/09/11 12:31 PM
Quote Jorge:

Vin, will your utility measure RTL in a comparable way for inexpensive digital interface/mixer combinations such as the A&H Zed series or the Presonus StudioLive series?




If they have an audio input and output that appears as an option in Windows then Vin's utility should work nicely


Martin

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Jorge]
      #938313 - 02/09/11 09:35 PM
Quote Jorge:

Vin, will your utility measure RTL in a comparable way for inexpensive digital interface/mixer combinations such as the A&H Zed series or the Presonus StudioLive series? Some of us are considering using that format in studio for overdubbing and it would be good to compare with the straight interface units. My sense is that these are usually used for live sound where RTL is not really the critical parameter, and I suspect less engineering effort has been put into minimizing latency than for recording studio interfaces.




Hey Jorge,

As Martin noted the utility can be used with any interface.

Re the Mixer/Interface combo's, the performance will depend on the controller /driver implementation employed. The Midas for example, performance was close enough to identical to the Focusrite unit using the same OEM controller/Driver. I have a Presonus DL unit on its way for qualification on a clients system shortly and I am hoping that there has been some work done on the drivers since I last tested one last year, as the drivers were in a league of their own in regards to LLP , and I don't mean in a good way. Despite using one of the now all too common Dice OEM variants as a controller, the driver control panel/applet was different to their studio interfaces , the lowest buffer setting was not as low either ( can't remember exactly whether it was 128/256 ), so the focus is definitely not the Studio IMO.

I'll post the results when I have them.

V:

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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #938449 - 03/09/11 05:19 PM
Thanks, Vin (and Dave and Martin). I look forward to hearing more about the SL to see if the latency with updated driver is anywhere near as low as the better dedicated recording interfaces.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #939891 - 10/09/11 05:40 AM
Quick one to touch base.

@Jorge,

I'll have a Presonus Studio Live on the bench over the next few days so I'll drop back once I have managed to give it a run thru.

On a side note , as much as the initial focus was on the OEM FW controllers , with a lot of manufacturers now moving their hand to USB2 I'll be making a specific effort to get as many of the new USB2 interfaces across the bench in future. I have some preliminary results from some of the new USB2 interfaces and to say they are disappointing would be putting it mildly. There seems to be a significant shift towards double/triple buffering which needless to say is extremely detrimental to I/O and RTL values.

No amount of spin, polish and marketing hype will detract from the fact that the extended I/O and RTL on these units would not be suitable for end users requiring decent LLP , so I am even more determined to highlight the units exercising the practice.

Stay Tuned.


V:

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Jorge
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #939938 - 10/09/11 03:01 PM
@TAFKAT I am very much looking forward to hearing what you find, both with the SL and the USB2 units. Over the next few months I plan to upgrade my old Delta 66 to a USB2 or PCIe interface, for 2-4 channel multitracking and overdubbing. RTL is one of the key parameters I am looking to optimize.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #940156 - 11/09/11 10:25 PM
Hey all,

I ran the Presonus StudioLIve 16.4.2 thru the test suite and its feeling like ground hog day with any of these units using the Dice OEM solution.

First note re the lowest latency setting , by default when the control panel is first called up default is set to 256 , which is also the lowest setting offered. However on toggling the safety buffer option from Normal to Safety 1 and back again to Normal again, the lower settings of 064/128 are enabled. That behaviour has been experienced on 3 systems so one to take note for those using the units.

Reported I/O and RTL is identical to the Focusrite Saffire 56 and the Midas Venice F previously tested which use the same Dice OEM controller , hinting further that the driver is not reporting the AD/DA, only the buffer/safety buffer and the DSP. I am having some trouble trying to get a reading from the RTL utility with the routing of the desk , but it is something I am keen to clarify for all of the Dice units.

Performance between the 3 units is close to identical, some small variations in respective plugins/polyphony but its pretty safe to say the performance between the 3 units is the same.

The deeper I get into the investigation its evident that there really isn't a wide choice for the end user in regards to selecting an interface for LLP. The units using the Dice OEM solution and maintaining the base driver supplied with the DDK are all at one level. AVID/M-Audio have put in some work with the driver to be measurably better using the same controller so its not rocket science that the others aren't. The only other FW OEM controller available on the market for manufacturers is ArchWave - ( formally marketed as Bridgeco but has splintered off from the main arm ). There is a list of Manufacturers using the chip on their website but I haven't had the opportunity to test any as yet so it would be interesting to get a head to head with their latest v the Dice powered units and see where the dust settles. I have a Mackie Onyx 1640i here again for a configuration during the week which I noted on earlier testing had a different style of buffer settings , I/O , RTL values and overall feel, so I am wondering whether that is in fact a ArchWave controller despite the rack/desktop Onyx interfaces using Dice.

Which brings me to the new crop of USB2 interfaces, lets just say that from the units tested so far apart from AVID/M-Audio and RME , I am not optimistic about those using various OEM controller solutions , more misty optic... :-(

I'll post some charts for the latest interfaces when I get some clear air

Peace

V:

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #940182 - 12/09/11 04:57 AM
Keep up the good work Vin - if you were being paid by the hour for this work you could have retired by now


Martin

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #940847 - 14/09/11 08:46 PM
Hey Martin,

Thanks for the ongoing support, No retirement plans just yet.. :-)

The further we get into this project the deeper the rabbit hole . I had an Echo Gina3 user post a concern that Cubase maybe reporting incorrect playback buffer values ( double ) as other Apps were not reporting the same value. He had initially tested in Abelton Live and Reason, we dug a little further using the Centrance Utility and of course the RTL showed single buffering on playback not double as being reported by Cubase.

I quickly double checked and cross referenced the Echo AF12 in Cubase, Reaper and SONAR X1 and StudioOne, also checked with the Lynx 2 and RME HDSP cards that are still installed on the system as well.

RME/Lynx report identical results across all 4 applications , the Echo reported the double buffering on output in Cubase but not in the other DAW hosts. Digging further , there is an Audio Priority setting in Cubase which balances processing for audio and non audio events , Normal ( default ) gives Audio / MIDI equal priority , Boost gives MIDI priority.

By toggling that setting and hitting reset ( reinitialising the audio driver ), on odd occasions I would see the output buffer drop to the single buffer setting identical to the input as reported by the other DAW hosts , it would only stay on the setting for a few seconds and then revert back to the double buffer value. After a few attempts I couldn't get the buffer to monetarily change to the single figure, so I got a short glimpse and then it settled back to reporting only the double buffering.

What that means exactly I am not entirely sure except that there is something specific within the Echo ASIO driver and the Cubase ASIO implementation that is assigning the double buffering. If it is reporting an incorrect value then it would compromise any audio process dependent on accurate I/O and RTL values , so I don't think that is the case.

BTW: I tried again to get a reading for the AF12 via the 2 RTL utilities and the results were all over the place , neither settled on a consistent value and both were above values being reported by Cubase with the double buffering and way above the other DAW Hosts. With most other cards I can get a pretty accurate result within seconds, so something screwy happening in the back end with this unit... :-(

So if its only Cubase reporting/implementing double buffering on output with this specific audio interface/driver, question then arises how many other interfaces reporting/implementing double buffering on output in Cubase, do not in other DAW hosts ?

Which leads me to the next request/call to arms if I may.

I would like to sort a manual measuring procedure as has been suggested by a few on this forum and also used by Mr Anderton on that Roland review of his , but I seriously do not have the time to set up a session and methodology at the moment. If anyone reading in has the time and energy to configure a Cubase session or even Reaper if thats easier or more convenient, using a short percussion sample configure a quick RTL loop / recording / measuring procedure so we can use it to double check the interfaces that are throwing up some added curve balls , that would be greatly appreciated.

The fun continues...

V:

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #941119 - 16/09/11 01:41 AM
Quote TAFKAT:

Hey Martin,

Thanks for the ongoing support, No retirement plans just yet.. :-)






Quote:

If anyone reading in has the time and energy to configure a Cubase session or even Reaper if thats easier or more convenient, using a short percussion sample configure a quick RTL loop / recording / measuring procedure so we can use it to double check the interfaces that are throwing up some added curve balls , that would be greatly appreciated.




The easiest way to do this in Reaper is to use the bundled ReaInsert plug-in, which lets you add external hardware in a send/return loop, but instead just plug in a couple of loopback cables.

ReaInsert includes a Delay Compensation 'Auto Detect' button that works well for me, with numbers repeatable to sample accuracy.




Martin

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #941357 - 16/09/11 10:27 PM
Hey Martin,

Thanks Mate , so simple .

In Cubase we should be able to use the same principle via the ping test for external HW.

FYI : Next build of the RTL Utility will also list the values reported by the ASIO driver to the Hosts as well as the AD/DA RTL.

V:

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Nightredt



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #941570 - 17/09/11 10:37 PM
Hey,

I had a question, which was bothering me for a while. Does the sound quality differs from different Audio interfaces? And if the sound quality differs between USB and Firewire devices?
Thank you

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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Nightredt]
      #941623 - 18/09/11 07:29 AM
Quote Nightredt:

Hey,

I had a question, which was bothering me for a while. Does the sound quality differs from different Audio interfaces? And if the sound quality differs between USB and Firewire devices?
Thank you



I think your question is outside the ambit of the latency tests but it is none the less a very good one!
You are a new poster and perhaps a new SoS reader? If so you will have missed many AI and a few (precious few!) soundcard reviews in which the sound quality was carefully compared with the reviewers' rig or/and any other converters he had about at the time. These tests seem to have been abandonded? We get a subjective "good clean sound, good LF heft..."Di Da...but rarely a comparison. I well remember MWs' review of the M-Audio AP192 in which he said, IIRC, it was an improvement, albeit a small one, on his existing rig (I have wanted one ever since but never seem to get A in gear!).
There are also quite a few PCI and a few, VERY few! PCIe soundcards around that have never been reviewed. Is there a reason for that other than the obvious "They don't send 'em!" ?

Dave.


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: ef37a]
      #943147 - 25/09/11 07:01 PM
Quote ef37a:

You are a new poster and perhaps a new SoS reader? If so you will have missed many AI and a few (precious few!) soundcard reviews in which the sound quality was carefully compared with the reviewers' rig or/and any other converters he had about at the time. These tests seem to have been abandonded? We get a subjective "good clean sound, good LF heft..."Di Da...but rarely a comparison. I well remember MWs' review of the M-Audio AP192 in which he said, IIRC, it was an improvement, albeit a small one, on his existing rig




Hi Dave!

Hugh (Robjohns) has done plenty of A/B comparisons in his audio reviews, but as far as I know I do seem to be the only SOS reviewer to consistently compare each audio interface I review against a benchmark comparison so that readers get an idea how much better/worse it is than that benchmark.

Having said that, it IS tricky to do properly. I employ a passive comparator with the output levels of each audio interface being matched to less than 0.1 dB and specially set up software that lets me hear the same music tracks being played in sync through multiple ASIO drivers simultaneously

However, this arrangement does mean that I can perform double blind switching back and forth between multiple interfaces in a fraction of a second to potentially hear tiny differences in audio quality. I note down any difference I can discern, and then find out at the end of my comparator tests which interface is which - sometimes with interesting results


Martin

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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #943242 - 26/09/11 08:15 AM
Thank you Martin.
Yes, tricky! Audio A/B tests have ever been so! (no reason to dispense with them though?).

Then, for all your great care and attention it is only YOUR (afaik) impressions that are recorded.

It would be very interesting to have a statisically valid number of people of various backgrounds, some audio experts, some pro musicians but "technically challenged" and others from a totaly different walk of life involved in your tests and running the top 5 say, cards/AI's (you could run a comp' for readers to pick the 5!).

THAT, I suspect might be even more interesting! Might disjoint a few noses too?

Dave.


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Dishpan



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: ef37a]
      #943924 - 28/09/11 07:17 PM
It's simple, instead of a single AD or DA stage, run a loop so you're going A-D/DA maybe 16 times (obviously levels need to be accurately calibrated). Then you're clearly hear the difference between the combined convertor stages of different interfaces. You can clearly hear noise and harmonic distortion and clearly hear the differences between different models.

I've done this but not sure why reviewers don't seem to think of doing it; It's quite easy to do and doesn't take too long to setup.

(of course cynical/clever manufacturers could detect this and make the input look more like the output!!)

Edited by Dishpan (28/09/11 07:31 PM)


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Dishpan]
      #944045 - 29/09/11 12:16 PM
That’s a useful approach Dishpan, particularly to highlight differences between interfaces by exaggerating them to the nth degree.

However, I’m not sure how the comparison relates to real-world results when what most people really want to know is whether or not you can hear a difference between interfaces after one pass through the converters


Martin

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Frederick



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #944452 - 01/10/11 03:17 PM
I'd just like to say I've been through four M-Audio Profire 610's now, all had developed looseness in the volume control within a month (if it wasn't there to start with).

I'm going to try the Avid Mbox Pro next (I assume this is the same as Mbox Pro 3 you tested TAFKAT and 3 refers to version), in the hope that is of better build quality (for £500 one would hope so).


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Frederick]
      #944499 - 01/10/11 10:05 PM
Quote Frederick:

I'm going to try the Avid Mbox Pro next (I assume this is the same as Mbox Pro 3 you tested TAFKAT and 3 refers to version....




Correct , 3 refers to the new 3rd generation units :-)

V:

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #948649 - 23/10/11 05:39 AM
Hey All,

I haven't had a lot of time for testing lately but I thought it would be good to post the latest batch of results and amendments that I have collated since the last posting.



Interfaces Added :

MOTU - 828 Mk3 - FW : Respectable performance across the board, only caveat being that it uses double buffering on playback which effects its RTL.

Presonus Studio Live 16.4.2: Another unit using the now familiar Dice OEM controller/driver solution , performance being on par with other interfaces using the identical controller /driver.

Focusrite Saffire 6 USB : Worst performing and highest I/O and RTL of any interface tested , needless to say this did not rate well. * Testing and results by Scan U.K *

Amended Results:

AVID Mbox Pro 3 , Focusrite Saffire 56 , Midas Venice F - all units were not reporting AD/DA , RTL has been calculated using the utility, RTL and LLP Ratings have been adjusted accordingly.

Peace

Vin Curigliano

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #948900 - 24/10/11 11:31 AM
Thanks Vin!

I found even in its ‘magnified’ form the results were still too small to read with my aged eyes, so I right-clicked on the magnified version and used ‘Save Image As’ so I could explore it at any size at my leisure


Martin

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #949017 - 24/10/11 08:17 PM
Quote Martin Walker:

Thanks Vin!

I found even in its ‘magnified’ form the results were still too small to read with my aged eyes, so I right-clicked on the magnified version and used ‘Save Image As’ so I could explore it at any size at my leisure


Martin




Yeh I noticed that to, however there is a simpler fix than downloading the file.

At the top of the magnified image there are some icons, click the full size tab ( square ) or hit the F key and it expands to full size... :-)

Vin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #949214 - 25/10/11 01:45 PM
Doh!! Why haven't I noticed that before?




Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #962634 - 08/01/12 10:11 PM
Hey All,

Here's the final list of interfaces tested in 2011 , with some added detail of the new interfaces added/amended to the list and a summary.





New Interfaces added.

NI Komplete Audio 6 : Respectable performance , very solid drivers, higher than normal RTL due to added safety buffers. * Testing and results by Scan U.K *

Presonus Audio Box 44VSL : Good I/O , RTL results, 032 latency setting is unusable being more window dressing than anything else , performance close enough to equal to their FW interfaces.

Steinberg UR28M : High I/O and RTL , one of the worst performing interfaces tested so far. It did at least manage a result/rating. * Testing and results by Scan U.K *


Amended Results.

Echo AudioFire 12 : I went back and retested this interface with the latest 5.7 version driver on the request of a few members reading in ( chart has a typo and still shows driver version 5.5 ) . Performance was identical to 5.5 , but this time I managed to get a results using the latest build of the in house RTL utility. There had already been some confusion with the reported I/O and RTL, as they would report differently in Cubase to Reaper/Sonar , so I wanted to get an accurate reading using the RTL utility to get a result independent from the DAW hosts. Heres where the funs starts as the unit reported huge added safety buffers per respective setting which would increase as the settings went up - ie. added 256 samples from 064 to 128 setting, 512 samples 128 to 256 , I couldn't get the final figure as the RTL utility instantly crashed trying to get a reading for the 512 setting.

The inconsistency that this unit displays in reported and actual RTL's is quite disturbing and I am way past spending any more time on it considering the product line is DOA here locally , literally , I had the only working unit from the old distros' and there will be no further units available from that distro at least. I won't be revisiting the Echo FW units.

Pending Results.

Focusrite Scarlett 2i2 : This one is an interesting case , we did manage to get testing and results completed but the respective settings are so far off the standard , - i.e 44/88/132/264/528 , that is was difficult to correlate against the baseline. The upper 3 settings I could let slide, but the 44/88 settings were the tipping point.

Mind you I/O and RTL are thru the roof and have absolutely no correlation to the listed settings - i.e Reported I/O @ 44 - 3.356/4.376 : Calculated RTL 11.1 ms , all the way to Reported I/O @ 528 - 25.351/37.370 : Calculated RTL 66.00 ms , yep, you read that right.

Needless to say this would not not rate well even if we could get a final rating.

M-Audio C400/C600: This unit tested very close to the results for the Fastrack Ultra 8R, but had a strange quirk where the performance dipped at 512. I have reported the issue to the devs and am awaiting resolution before posting final results.

Tested but No Rating Achieved :

Lynx Aurora 8 - FW : Unfortunately this unit proved extremely unstable and unpredictable that testing was abandoned , still hoping to test the unit with the AES16 PCIe card.

EMU 0404 - USB2 : This unit was extremely inconsistent , lower results at higher buffer settings than the previous, vastly different results at respective buffer settings on consecutive runs , etc.


That closes off the testing for the last year , and its been interesting to say the least. A quick summary on my part would be that there are only very few manufacturers who really focus on their drivers to deliver good Low Latency Performance , and its obvious from the charts who they are.

The shift to USB2 that has been gaining momentum is not going to guarantee better performance/compatibility , especially for those manufacturers again relying on 3rd party OEM controllers/drivers. There are some stand outs at both ends of the spectrum listed in the charts: I am not going to go into further detail or commentary as the results speak for themselves..

I will be writing a more detailed report for the main DAWbench website when I get some clear air.

Peace

Vin Curigliano

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Edited by Martin Walker (09/01/12 11:20 AM)


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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #962687 - 09/01/12 07:29 AM
Greeting TAFKAT,
Nice to see I have picked a winner for once in the K6! I was very close to getting a Scarlet, "something".
I have now tried the interface on 3 very different desktops and a laptop and 3 os'ess, XP, pro and home and W7/64,no worries.

I have just read of a guy who bought a K6 but said he could not get audio to run on his pc, MIDI but not audio! He has since switched it for a Scarlet 2i2 which he says works but I have asked him to "Speccy" his pc, give me the details and intend to ask NI if they know of any problems with same.

Dave.


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Pete Kaine
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #962713 - 09/01/12 10:11 AM
Quote TAFKAT:


Steinberg UR28M : High I/O and RTL , one of the worst performing interfaces tested so far. It did at least manage a result/rating. * Testing and results by Scan U.K *





Just to note, Steinberg were notified back in November about this and when I had a chat with one of thier sales guys, he mentioned that the testing guys had also noticed something was up in certain situations and they were already working on a resolve. An update to fix was promised Dec/Jan so we should be due an update soon and we'll retest when they can get one over to us.

Quote TAFKAT:



Lynx Aurora 8 - FW : Unfortunately this unit proved extremely unstable and unpredictable that testing was abandoned , still hoping to test the unit with the AES16 PCIe card.





We also tested and found the same here. Just switching buffer sizes with a sequencer open would cause a reboot each time and that aside it just wasn't all that stable. Considering the end user price of the unit we were stunned.

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #962737 - 09/01/12 11:34 AM
Many thanks to Vin for all this hard work behind the scenes with little public support (apart from the occasional gem like Pete Kaine who has access to a large range of audio interfaces and is prepared to put his neck on the block )

Vin is often at the receiving end of lots of vitriol from people who think he has some axe to grind or is attempting to skew results in line with whatever conspiracy theory is currently the favourite.

However, from my studies of his results (and in various cases from repeating them for myself as and when I get specific audio interfaces in for test, when I have always got identical figures within the bounds of experimental error) I am confident that this table shows the current state of low latency performance for a good variety of makes and models.

Well done that man!


Martin

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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #962855 - 09/01/12 05:09 PM
Martin.
Re conspriracy theories: There is a ("colonial" shall we say!) forum that consistantly slates M-Audio for bad drivers and poor latency. "Well known for it" they say but never give any of the supposed wodge of complaints flooding into the forum when pressed for evidence.

Not I admit in the RME class but I have never had any problem with M-Audio kit?

Dave.


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #962891 - 09/01/12 08:08 PM
Hey Martin,

Thanks for the continuing support , much appreciated.

I would also like to thank both Pete and Tom from Scan for stepping up and actually rolling up their sleeves and contributing to the testing and database , to the point where they have set up an identical benching system to ensure the collated data is consistent.

I understand that this can be an extremely sensitive area for some manufacturers and even dealers, but in the end all we are presenting is quantifiable data , what that means to respective parties varies. For dealers this is obvious dangerous ground trying to maintain working relationships with the manufacturers/distros when the products are not performing well and you still have to manage the relationship and support for the product. Its not an easy task and its quite volatile as I have discovered to my own detriment in some cases , but the bottom line is what is in the best interest of the end user over simply selling a box, well it is in my case and I am sure with Pete and Tom.

I have to admit I was surprised by the level of resistance I received from various angles during the last 12 months, a quick overview would be that some manufacturers / distros were more than happy to send me product to test and report on , others were not prepared to unless I assured them a positive result ???, while others simply didn't deliver on the promised test units after reading the reports. Also trying to get further details in regards to the OEM controllers and the drivers from the developers was futile , this is not an area that is going to be easily approached.

Now we have to be very specific about what this database represents and where it is most viable to the end users . The term Low Latency is bandied around quite a lot but in varying context , for many manufacturers Low Latency is in terms of Monitoring , and in the case of Direct Monitoring this data has no relevance as any of the units with a DSP/Hardware mixer will allow RTL of under or around 2ms. Significantly more important is what performance is delivered at the lower latency buffer settings when using Virtual Instruments in live playing and even more importantly when monitoring with FX, which is where the RTL is extremely important.

Martin started a database here early last year asking members to list what latency they mostly worked at , problem being that simply stating 064, 128 or 256 although giving an overview, does not take into account that those actual settings on respective interfaces mean totally different things.

Perfect example , actual RTL figures for 064 ranged from under 4.5 m.s to over 10.ms , some members were reporting they were comfortable at 256 and didn't need to go to the lower extremities, but again we have RTL values ranging from 13 ms all the way to above 30 ms , and those interfaces with the extended safety buffers do not necessarily perform better at those higher values.

I'll continue to test and report on any interface that crosses my desk and I'm sure the boys at Scan will be contributing as well , so the database will continue to grow and hopefully be of benefit to the greater community , which also includes the manufacturers and developers , how some choose to navigate and/or respond to the information is the grey area.

Peace

V.C

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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #962896 - 09/01/12 08:35 PM
Looking good Vin, and with NAMM just around the corner you'll have a whole new batch to play with

Like Pete i've had a UR28M to play with since release - i was hoping to use it as my new USB benchmark interface but the results were simply too poor in comparison to anything else i'd tested - not sound quality, just plug-in counts before peaking. I've also been in conversation with Steinberg about it and i understand them to be "working on it" at the moment. I hope they can do something with it as it's a great box otherwise - great sound etc.

It's interesting how Scan has now put "LLP" stickers on some of the audio interfaces - nice touch Pete

In my own special way, being a bit special and that, i've embarked on my own more Heath-Robinson approach to latency testing. Rather than messing about with loopbacks and utilities i plug a mic into the interface, through a laptop with Cubase, out to speakers. I then place another mic near the speakers running into soundforge on another computer. I arm a track and turn on monitoring and click my fingers in front of the first mic. The second mic then records the original click plus the click from the speakers - find the attack points in soundforge and measure the difference. The idea is that i'm trying to measure what people actually experience - i hope it will prove interesting. I'll post the results once i have more than one. Currently doing it with the Propellerhead Balance - great box, fixed latency in Cubase, no control panel - very odd - that'll mess up your graph good and proper Vin.

One other suggested enhancement for the graph - you should band the results into a simple "Great - good - rubbish" - what i mean is at what point along 1-10 on the graph would you consider acceptable performance and what would be fabulous - do ya get me yeah?

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #962903 - 09/01/12 09:13 PM
Quote robinv:



In my own special way, being a bit special and that, i've embarked on my own more Heath-Robinson approach to latency testing. Rather than messing about with loopbacks and utilities i plug a mic into the interface, through a laptop with Cubase, out to speakers. I then place another mic near the speakers running into soundforge on another computer. I arm a track and turn on monitoring and click my fingers in front of the first mic. The second mic then records the original click plus the click from the speakers - find the attack points in soundforge and measure the difference. The idea is that i'm trying to measure what people actually experience - i hope it will prove interesting. I'll post the results once i have more than one. Currently doing it with the Propellerhead Balance - great box, fixed latency in Cubase, no control panel - very odd - that'll mess up your graph good and proper Vin.




Hey Robin,

Thanks for the support, much appreciated.. :-)

Your testing sounds interesting , what you are essentially doing is adding the latency over distance on top of the RTL , all good.

Re The Propellerhead unit, what do you mean it has a fixed latency and no control panel, are you sure Mate ?

Check in device manager under the listing for the driver and you may find a panel under properties , thats where the SSL PCIe panel is for example. Did my head in for a while finding that one... LOL

If you want the RTL Utility that both myself and Scan have been using to measure the units RTL ( still in BETA , will be available publicly soonish ), drop me a line and I'll forward it you. You can then use that to zero in on what the actual latency over distance is across a few interfaces compared to the RTL value to ensure its consistent.

Quote:

One other suggested enhancement for the graph - you should band the results into a simple "Great - good - rubbish" - what i mean is at what point along 1-10 on the graph would you consider acceptable performance and what would be fabulous - do ya get me yeah?




Well here's the thing , the actual acceptable working latency will be different for respective working environments , for example those simply tracking bands and not using guitar amp sims or a lot of sample based virtual instruments , compared to composers who demand the highest performance at those lower latencies , will have different levels of Fab to Drab, so to speak.

I think it might be safer for me to leave that up to individual interpretation.. :-)

Peace

V.C

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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #962920 - 09/01/12 10:32 PM
Quote TAFKAT:



Re The Propellerhead unit, what do you mean it has a fixed latency and no control panel, are you sure Mate ?




Yeah, pretty sure. Nothing in Control Panel, or under it's program menu or in any folder that i can find. I'm writing a review and am waiting to hear back from Props about it. You can change the buffer in Reason, within Reasons Audio Properties but it doesnt stick. In Cubase it's 5.8ms in and out - nothing comes up pressing the control panel button in Cubase either. Very strange - but at least you're not having to mess about


Quote:



I think it might be safer for me to leave that up to individual interpretation.. :-)




Totally - but y'know, just a "I think anything over a 6 is pretty good" type thing.
Yeah, i'll poke you about the utility - although i try to not to concern myself with anything too scientific.

Had a slightly better idea than the mic's after a facebook discussion, to use a stereo pulse from a CD or hardware synth and put one side through the interface and one not - plug both channels into soundforge on the other computer and make a stereo recording - easy to measure difference and all the measuring has been done outside the test system (which is something i like). A bit more elegant than my holding mics and snapping fingers - but less fun - and i liked the distance in air bit and huge margin of error

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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #962924 - 09/01/12 10:46 PM
Err?
Don't know if it helps but most DAW sofware, even Audacity can generate tones of almost any desired frequency, level and duration. So you could for example have a 1mS burst of 10kHz.

Dave.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #962926 - 09/01/12 10:53 PM
Quote robinv:


Yeah, pretty sure. Nothing in Control Panel, or under it's program menu or in any folder that i can find.




Did you check under the device manager listing as I suggested and see if the panel is there ?


Quote:

Totally - but y'know, just a "I think anything over a 6 is pretty good" type thing.




Sure , but then punters will be asking where the 5-6's fit into the equation, etc,

Personally I only use RME and Lynx PCI/PCIe cards in the studio , that will give you an idea of the level of performance I prefer to work with ... :-)

Quote:

Yeah, i'll poke you about the utility - although i try to not to concern myself with anything too scientific.




LOL, Hmmm, I think what you are doing with your testing would be construed by many as being pretty scientific in nature... :-)

V.C

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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #963238 - 11/01/12 11:45 AM
Quote TAFKAT:



Did you check under the device manager listing as I suggested and see if the panel is there ?





Yes, and then no
Quote:


Sure , but then punters will be asking where the 5-6's fit into the equation, etc,




Impossible isn't it? Actually maybe that's a question for Pete because Scan have put "LLP approved" stickers on some interfaces - Pete, at what point is an interface good enough to get a sticker?

Quote:



LOL, Hmmm, I think what you are doing with your testing would be construed by many as being pretty scientific in nature... :-)





Ah yes but i havent told you about the parallel testing of sitting in a room and "feeling" various latencies and then expressing that in poetry.

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #964248 - 15/01/12 10:53 PM
Quick Update.

3 years back I had a rather frustrating experience with a Prism Orpheus that I outlined in a thread at the Cubendo Forum , in short the unit was extremely inconsistent , had a convoluted buffer setting protocol , large safety buffers and mediocre performance , so not exactly impressive in respect to LLP .

With the focus on the 3rd party OEM Firewire controllers being used in some of the current crop of interfaces , this unit has my added interest because it doesn't use the more widely used Dice chip, instead using the BridgeCo/ArchWave OEM controller/driver , which would allow me to do some direct head to head against the 2 available OEM controllers.

This unit is a strange bird IMO - Prism are renown for the extreme high end converters , and this unit of course has the geneology of the higher end product line but is priced substantially lower, but not what anyone would call low mind you at around 5K, and it is an audio interface first and foremost. The full AD/DA potential of this unit is reserved to the audio interface as using it as a stand alone AD/DA it has only 1 x ADAT I/O which locks it to 48K, and no AES/MADI option which would allow it to use its 192 capability.

With the potential of the unit being so dependent on the audio interface component , I would expect that Prism would ensure that the audio interface controller and driver would deliver performance that would be worthy of such a high end professional unit.

Since my last experience which was on XP and Driver/Control version 1.05 , there has been a few upgrades to the current version of 1.07, not exactly a lot in the last 3 years but it was worth investigating and I was lucky enough to have a unit available for me to run it thru its paces again. FWIW - The client has been running his Orpheus units via ADAT the last 3 years connected to an RME HDSPe RAYDAT , as the units were simply unworkable for him as stand alone interfaces.

First off , the units control panel is identical the last time I tested the unit, there are 2 parameters that need to be adjusted to set the latency - Buffer (us) / Latency (ms) , the buffer time setting can be manually adjusted in increments of 250 us which in turn alters the available latency settings as multiples . Standard setting is 1500 us, which would then give a range of latency values from 3 ms to 12 ms. The 2 settings give a lot of flexibility to the available latency values over all other interfaces I have tested, but also an added level of complexity and there will be latency settings that will be available across multiple buffer time settings as well.

Some quick points on the latency values available on the units - the settings are listed in ms ( nominal ) , so I was able to use .75 / 1.5 / 3.0 / 6.0 / 12.0 ms for the standard test values of 032 / 064 / 128 / 256 / 512 , however the actual sample buffers as calculated by the RTL Utility were actually 032 / 066 / 132 / 264 / 528 . I tested both the 064/128 settings across 2 available buffer time settings 250/500 for 064 , 500/1000 for the 128 setting, and found in both instances that the lower buffer time setting for the respective latency values performed substantially better , in the range or 15-30%. I then used only the lowest buffer time for each respective latency value.

I hope that makes sense.

O.K, with that out of the way , how did it perform.



Performance is typical of interfaces using 3rd party OEM controllers , large safety buffers , average to below average performance at lower latencies in regards to Virtual Instruments , but there is an interesting quirk in the RXC results. Performance on that particular test was quite good all the way down to 032 samples , but the performance achieved at 064 samples could not match the 032 result - I reran the tests 3 times at each respective buffer to be 100% sure. Add to that if we factored in the 064 result using the larger buffer time which dropped the number of RXC's 30% to 89 , and the 032 results stands as even more of an oddity.

So in closing , the experience this time around was definitely more consistent than my first run on the unit , but the overall performance of the unit as an audio interface is average at best , and with no option to use the full potential of the AD/DA sans the interface, those looking at the unit as a primary interface will need to take that into account.

Of course YMMV.

Now stating the obvious before I get a wave of irate Orpheus users on here taking issue, this report is purely on the Low Latency Performance of the FW Controller/Driver , it is not in reference to the feature set , quality of the AD/DA , so please use the numerous other threads to continue that debate.

P.S : I have updated the previous charts from last week to now include the Orpheus

Peace

V.C

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #964509 - 17/01/12 04:08 AM
Quick heads up,

The Round Trip Latency measuring utility that I have helped develop is now available for public download.

RTL Utility by Oblique Audio : Here

Peace

V.C

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fito_88



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #968249 - 04/02/12 11:13 AM
guys to clear this out to me , are you stating that the rme hdspe aio is the card which has the lowest rtl latency in the market , is that why you are taking it as a baseline reference in all your tests ? and does that mean its the best PCIe card in the rme series in term of latency ? does it out perform the rme 9632 PCI card or they are both the same with the AIO is only PCIe card ? help please


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: fito_88]
      #968285 - 04/02/12 02:53 PM
Yes, RME’s HDSPe AIO PCIe card currently gives the best results at low latency of all the interfaces so far tested:

www.rme-audio.de/en_products_hdspe_aio.php

The 9632 PCI card is good, but up to the standard of the newer PCIe version as far as low latency is concerned.

Hope this helps!


Martin

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #968362 - 04/02/12 10:47 PM
Hey Martin,

Quick heads up, the base reference card is actually the HDSPe AES / ADI8 QS combo, which superseded the AIO as the base reference due to its tighter AD/DA and subsequent lower RTL.

Overall performance of the driver is identical.

The HDSP 9632 and HDSPe AIO are actually pretty much on par performance and RTL wise, its really down to the required interconnect - i.e, I would recommend the AIO over the 9632 on any current Intel chipset system using a bridged PCI implementation for example.

P.S - It would be great if we can edit the original post and add the latest results at the bottom with a link to the latest posting and summary. That may help minimise the confusion re what the current base reference card is.

Peace

V.C

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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #969999 - 13/02/12 09:13 PM
Quick Update,

I managed to finally run a Lynx AES16e / Aurora combo thru its paces.

The earlier Lynx 2 card performed very well so I was interested in getting the latest PCIe offering on the bench to see if the performance could be maintained , especially with the ultra tight AD/DA latency of the Aurora.



Overall performance of the PCIe unit did not match the older Lynx 2 card unfortunately , I also expereinced some very odd loss of buffer sync issues at 256 samples with the DAWbench VI test sessions which resulted in random garbled playback and metallic like artefacts on release tails, etc. This was more often than not triggered by loading a previous maxed out session from the 128 buffer size and trying to incrementally continue to load the session. I had to to start the 256 session from scratch each time. Where this could be a problem in Real World is if you had a heavily loaded session and wanted to bump up a buffer size to get some more headroom, you would in fact not be able to play the previous session without artefacts.

Overall I/O and RTL latency is actually lower than the RME AES/ADI8 reference unit , so potentially could have tipped the base reference interface if it had performed equally to the older Lynx 2 card, but unfortunately it didn't measure up to its older sibling. The results look quite good, but there are a few quirks that left me with the feeling that its not as solid as it could be.

I am currently in contact with Lynx support and am working thru a few angles.

I'll give you guys a heads up if anything fruitful eventuates.

* I have added the Lynx to the most recent database list *

Peace

V.C

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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #986202 - 07/05/12 09:55 PM
Hey All,

Just a small update to the database with amended results for the Steinberg UR28M which has shown a significant improvement with the new drivers.





Updates.

Steinberg UR28M : Significant improvement with the latest version 1.1.1 drivers over the original release drivers which in short were unworkable on Windows IMO, unit still has high comparative I/O and RTL for the respective buffer settings , but scaling performance at the respective latencies is now quite respectable.

Peace

V.C

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #986220 - 07/05/12 11:50 PM
Thanks for the update Vin!


Martin

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alexis



Joined: 10/01/03
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #986233 - 08/05/12 01:42 AM
Quote Martin Walker:

Yes, RME’s HDSPe AIO PCIe card currently gives the best results at low latency of all the interfaces so far tested:

www.rme-audio.de/en_products_hdspe_aio.php

The 9632 PCI card is good, but up to the standard of the newer PCIe version as far as low latency is concerned.

Hope this helps!


Martin




Warning, newbie question to follow:

"0.7 msec latency..." . Does achieving that latency depend not only on the card, but also on the computer? Or is all the heavy lifting done by the card, and any reasonably good music computer can tag along and perform well?


Cubase 6.0.5, 32-bit; XP SP2; Seagate Barracuda 80/160/160GB; Intel 550 Pentium 4 3.4GHz, 2GB RAM, HT+; Intel D915PBL mobo; Dual Radeon ATI X300SE 128MB video; Delta 66 in Omni I/O Studio; Motif8; UAD-1

Thanks!

--------------------
Alexis -Cubase7.5.20 64bit;i5-4570 3.2GHz,16GB RAM;W7SP1 64bit on Samsung SSD840 Pro256GB;Seagte 1TB SATA600 Audio;UR28M;Motif8;UAD2Solo;Jamstix 3.3;BCF2K;TC Helicon VoiceOne;RevoicePro2.5


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: alexis]
      #986453 - 09/05/12 01:14 AM
Quote alexis:

"0.7 msec latency..." . Does achieving that latency depend not only on the card, but also on the computer? Or is all the heavy lifting done by the card, and any reasonably good music computer can tag along and perform well?




Hi alexis!

It's a combination of the two - you're unlikely to achieve glitch-free low latency audio on a PC that hasn't been set up carefully, but neither are you likely to achieve it with a mediocre audio interface.

Only when the two are combined can you manage the dizzy lows of under 3mS


Martin

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alexis



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #986456 - 09/05/12 01:27 AM
Quote Martin Walker:

Quote alexis:

"0.7 msec latency..." . Does achieving that latency depend not only on the card, but also on the computer? Or is all the heavy lifting done by the card, and any reasonably good music computer can tag along and perform well?




Hi alexis!

It's a combination of the two - you're unlikely to achieve glitch-free low latency audio on a PC that hasn't been set up carefully, but neither are you likely to achieve it with a mediocre audio interface.

Only when the two are combined can you manage the dizzy lows of under 3mS


Martin




Thanks for that, Martin!

My computer has been set up properly (A Carillon, specs in this post), but what once made it a pretty snazzy machine is only ho-hum (if that) by today's standards.

I've had my eye (and heart) on maybe the Fireface UCX. I'm wondering - is there any way of "looking" inside the computer to get an idea of whether it would be able to take reasonably full advantage of that interface?

Thanks for any thoughts!

--------------------
Alexis -Cubase7.5.20 64bit;i5-4570 3.2GHz,16GB RAM;W7SP1 64bit on Samsung SSD840 Pro256GB;Seagte 1TB SATA600 Audio;UR28M;Motif8;UAD2Solo;Jamstix 3.3;BCF2K;TC Helicon VoiceOne;RevoicePro2.5


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robinv



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #986500 - 09/05/12 09:50 AM
Vin, these images dont get any bigger so it's really hard to see - should they be linked to larger versions?

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Molten Music Technology - Computers for doing music on


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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #986501 - 09/05/12 09:53 AM
Quote robinv:

Vin, these images dont get any bigger so it's really hard to see - should they be linked to larger versions?




I think there was a fix for this....Can't remember what it is tho!

Dave.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: robinv]
      #986806 - 10/05/12 10:26 PM
Quote robinv:

Vin, these images dont get any bigger so it's really hard to see - should they be linked to larger versions?




Hey Robin,

Hmmmm, they used to pop up to the full size when clicked , looks like that feature has been disabled by the forum admins.

The actual links in the body of the post are to full size images on my server , if you hit quote, you will see the direct links.

PITA, but at least a workaround for now until the admins can sort whats happening on the forum posts

Martin,

Can you please look into that for me.

Peace

V.C

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: alexis]
      #986815 - 10/05/12 11:44 PM
Quote alexis:

I'm wondering - is there any way of "looking" inside the computer to get an idea of whether it would be able to take reasonably full advantage of that interface?

Thanks for any thoughts!




Hi alexis,

Glad I could help

What you need is Thesycon's DPC Latency Checker ( www.thesycon.de/eng/latency_check.shtml ), which I discussed in this sticky thread:

www.soundonsound.com/forum/showflat.php?Cat=&Number=588140

Running this should tell you how likely you are to manage a really low latency with a suitable audio interface


Martin

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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #986816 - 10/05/12 11:48 PM
Quote TAFKAT:

Quote robinv:

Vin, these images dont get any bigger so it's really hard to see - should they be linked to larger versions?




Hey Robin,

Hmmmm, they used to pop up to the full size when clicked , looks like that feature has been disabled by the forum admins.

The actual links in the body of the post are to full size images on my server , if you hit quote, you will see the direct links.

PITA, but at least a workaround for now until the admins can sort whats happening on the forum posts

Martin,

Can you please look into that for me.

Peace

V.C




Hi Vin!

Sorry that this is seemingly broken, but over the years we've had so many large images 'breaking' our forum code (causing text to become invisible beyond the right hand margin) that it's been tweaked quite a few times now.

For the time being your 'quote' and direct link approach works well for these two images, and most other smaller ones still work fine with the current code and don't break anything

I will mention it behind the scenes though


Martin

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alexis



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #986826 - 11/05/12 03:07 AM
Martin, thank you!

I downloaded DPC and tested my system. Before I post on that thread though, I wanted to make sure I was running the test properly. Also, this is the week that I have become totally humbled by the awareness of the large holes in my basic DAW knowledge. I can write, play, record and put on youtube, but somehow I don't know so many basic things!

I ran the test with a project running as you suggested in the other thread. I ran my Delta card at its lowest buffer size (64 samples, about 1.5 msec), and something large like 2056 samples or something.

1) The DPC test was green in both cases, averaging 200-300 at 64 sample buffer size, and 50-75 at the very large buffer size. At the largest buffer size, not surprisingly, the project ran smoothly, and the VST performance meter was close to zero. "This machine should be able to handle real time streaming of audio and/or video without dropouts" it said for both buffer sizes. Great!

2) BUT: at the smallest buffer size, the DAW was unusable: crackles/pops, VST performance meter pegged at 100%, stuttering graphics, etc. (Which of course isn't surprising either).

So how do I extrapolate from these tests whether my current machine will be able to run at less than 1 msec latency if I get an RME Fireface UCX? The DPC program says no problem ... but is the limiting factor my computer in ways the DPC program doesn't identify?

I guess I'm not sure why my DAW can't run at 64 sample buffer size - is it the stuff like motherboard/processor/RAM etc., or is it the Delta card?

Thanks for helping me think through this ... <<steam coming out of ears icon here>>

P.S. Also, I don't see that you asked for the buffer size as one of the parameters of interest to be listed in the DPC checker thread ... am I missing a main point by testing it as a function of buffer size?

--------------------
Alexis -Cubase7.5.20 64bit;i5-4570 3.2GHz,16GB RAM;W7SP1 64bit on Samsung SSD840 Pro256GB;Seagte 1TB SATA600 Audio;UR28M;Motif8;UAD2Solo;Jamstix 3.3;BCF2K;TC Helicon VoiceOne;RevoicePro2.5


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Martin WalkerModerator
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: alexis]
      #986960 - 11/05/12 10:30 PM
Hi again alexis!

My gut feeling is that with DPC Latency Checker measurements of 200-300 with a 64-sample buffer size your PC is well set up, and that it's your audio interface that limits how low you can go.

Once you have that RME Fireface UCX I suspect you'll be able to run with significantly smaller buffer sizes

I've occasionally had this issue with review machines – have a read of this review and you'll see what I mean:
www.soundonsound.com/sos/jun11/articles/nimbus-project-studio-plus.htm

This is the relevant section:
“My final tests, and, as always, the most important ones, were with DAWbench running inside Cubase 5.5, since this provides a much better indication of how the machine compares with others when running audio tracks and plug‑ins. At this point, I have to admit that with the ‘well‑known Firewire audio interface’ I was sent with the review PC I got impressive results at high latency (249 Cockos Reaxcomp plug‑ins with 256‑sample buffers). However, there was a significant drop-off in performance at smaller buffer sizes, with audio glitches at just 80 percent load with 128‑sample buffers, at a paltry 50 percent load with 64 samples, and no 32‑sample option at all.

I nearly finished my review at this point, but a nagging doubt in my mind about the poor efficiency of some Firewire audio interfaces at low latencies resulted in my pulling out the elderly Echo Mia PCI interface from my own PC and running the Sandy Bridge tests again. I’m very glad I did, because they rose to new highs. With the Echo Mia and a 256‑sample buffer size, this system managed a massive 305 Reaxcomp plug‑ins — an amazing achievement for a computer at this modest budget level.
Dropping to 128‑sample buffers this time resulted in a much smaller reduction to 273 plug‑ins (instead of 200), while at 64 samples I still managed 239 plug-ins (but only 120 with the Firewire interface!). This really does prove that the performance of audio interfaces can vary hugely, especially at lower latencies, so anyone who needs to monitor incoming audio with plug‑in effects should choose their audio interface very carefully.”

Hope this helps!


Martin

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alexis



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #986981 - 12/05/12 12:29 AM
Quote Martin Walker:

Hi again alexis!

My gut feeling is that with DPC Latency Checker measurements of 200-300 with a 64-sample buffer size your PC is well set up, and that it's your audio interface that limits how low you can go.

Once you have that RME Fireface UCX I suspect you'll be able to run with significantly smaller buffer sizes

I've occasionally had this issue with review machines – have a read of this review and you'll see what I mean:
<a href="/sos/jun11/articles/nimbus-project-studio-plus.htm" target="_blank">www.soundonsound.com/sos/jun11/articles/nimbus-project-studio-plus.htm& lt;/a>

This is the relevant section:
“My final tests, and, as always, the most important ones, were with DAWbench running inside Cubase 5.5, since this provides a much better indication of how the machine compares with others when running audio tracks and plug‑ins. At this point, I have to admit that with the ‘well‑known Firewire audio interface’ I was sent with the review PC I got impressive results at high latency (249 Cockos Reaxcomp plug‑ins with 256‑sample buffers). However, there was a significant drop-off in performance at smaller buffer sizes, with audio glitches at just 80 percent load with 128‑sample buffers, at a paltry 50 percent load with 64 samples, and no 32‑sample option at all.

I nearly finished my review at this point, but a nagging doubt in my mind about the poor efficiency of some Firewire audio interfaces at low latencies resulted in my pulling out the elderly Echo Mia PCI interface from my own PC and running the Sandy Bridge tests again. I’m very glad I did, because they rose to new highs. With the Echo Mia and a 256‑sample buffer size, this system managed a massive 305 Reaxcomp plug‑ins — an amazing achievement for a computer at this modest budget level.
Dropping to 128‑sample buffers this time resulted in a much smaller reduction to 273 plug‑ins (instead of 200), while at 64 samples I still managed 239 plug-ins (but only 120 with the Firewire interface!). This really does prove that the performance of audio interfaces can vary hugely, especially at lower latencies, so anyone who needs to monitor incoming audio with plug‑in effects should choose their audio interface very carefully.”

Hope this helps!


Martin




Yes sir!!

Once again, so grateful to the nth degree for this forum, and the SOS uber-experts. My goodness, where would newbies be without it?

Thanks - all that remains is to look between all the couch cushions for the extra coins so as to afford that UCX sooner!

Thanks so much again -

--------------------
Alexis -Cubase7.5.20 64bit;i5-4570 3.2GHz,16GB RAM;W7SP1 64bit on Samsung SSD840 Pro256GB;Seagte 1TB SATA600 Audio;UR28M;Motif8;UAD2Solo;Jamstix 3.3;BCF2K;TC Helicon VoiceOne;RevoicePro2.5


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Goddard



Joined: 04/04/12
Posts: 955
Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #987047 - 12/05/12 02:52 PM
Quote Martin Walker:

I nearly finished my review at this point, but a nagging doubt in my mind about the poor efficiency of some Firewire audio interfaces at low latencies resulted in my pulling out the elderly Echo Mia PCI interface from my own PC and running the Sandy Bridge tests again. I’m very glad I did, because they rose to new highs. With the Echo Mia and a 256‑sample buffer size, this system managed a massive 305 Reaxcomp plug‑ins — an amazing achievement for a computer at this modest budget level.
Dropping to 128‑sample buffers this time resulted in a much smaller reduction to 273 plug‑ins (instead of 200), while at 64 samples I still managed 239 plug-ins (but only 120 with the Firewire interface!). This really does prove that the performance of audio interfaces can vary hugely, especially at lower latencies, so anyone who needs to monitor incoming audio with plug‑in effects should choose their audio interface very carefully.”




Hi Martin, do you recall which drivers you were using for your Mia in the above testing? I assume it was the ASIO drivers? Just curious, as Echo have been offering WaveRT drivers (for Vista and Win7) as well as ASIO for some of their PCI interfaces, but I can't find any info on how the WaveRT drivers perform (or have I missed some info on Vin's site?).

(As you probably know, the P67 motherboard in that PC you reviewed uses a PCIe<>PCI bridge, so it is nice to know that your Mia actually worked so well in that board.)

Thanks!
=Goddard


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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Goddard]
      #987454 - 15/05/12 12:44 AM
Quote Goddard:

Hi Martin, do you recall which drivers you were using for your Mia in the above testing? I assume it was the ASIO drivers?




Hi Goddard!

Yes, you're quite correct in your assumption - they were the ASIO drivers for my Echo Mia


Martin

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Goddard



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: Martin Walker]
      #987887 - 16/05/12 09:07 PM
Thanks for confirming that, Martin!

BTW, I know you had mentioned Echo's WaveRT Vista drivers back in your April 2009 PC Notes, but I'm wondering whether you (or anyone else here) have tried Echo's latest (ver. 8.6 - 12/2011) ASIO and WaveRT drivers under Win7x64?

=Goddard


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Goddard



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #987896 - 16/05/12 09:33 PM
Quote TAFKAT:

Quote Kayvon:

I never see Marian mentioned much. I'm looking into their PCIe solutions for my next desktop PC. Anyone care to venture any opinions on them?




Marian unfortunately do not have any x64 drivers at present, which kind of leaves them off the radar for many if not most.

I have access to them, fingers crossed they get the x64 driver sorted.




Hi Vin, it looks like Marian have released Win7x64 drivers for their Seraph PCIe cards. Any chance you could run some tests now?

Thanks in advance!
=Goddard


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MartinJG



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #991870 - 07/06/12 10:10 PM

Can anyone offer feedback on the following based on their own experience.

I have just replaced my old M-Audio 2496 with an RME Babyface. However, I am not entirely convinced the D/A convertors on the Babyface are that much better than my old 2496. Is this a case of diminishing returns or is my hearing up the spout? I am using AE22 monitors by the way.

MJG


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ef37a



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: MartinJG]
      #991882 - 07/06/12 11:44 PM
Quote MartinJG:


Can anyone offer feedback on the following based on their own experience.

I have just replaced my old M-Audio 2496 with an RME Babyface. However, I am not entirely convinced the D/A convertors on the Babyface are that much better than my old 2496. Is this a case of diminishing returns or is my hearing up the spout? I am using AE22 monitors by the way.

MJG



As an impecunious user of two 2496s you are warming my cockles Martin! I really can't tell ZED10 into 2496 or NI KA6 via usb!

And is it just my perspective or do we not get the careful converter/AI comparisons we used to in SoS? Perhaps like Vins latency tests some VI noses would be put out of joint?

Dave.


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MartinJG



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: ef37a]
      #991914 - 08/06/12 09:48 AM
Hi Dave

As you know, I had probs with buzz (Sandybridge issues) on my 2496 so feel my hand was forced. I guess we have to go by what we hear despite the technostuff that tells us which are good and which are not. The reason I went (reluctantly) for the Babyface is the latency which is incidently very close to the 2496. I just don't like the 'feel' of anything more than @8ms. In this respect and the audio 'bit'(which I won't really use very much), the Babyface seems to be way out in front, but the sound? I have pretty good ears on the music front but can't vouch for their technical performance after a few years of wear and tear (although I have looked after them where poss'). Am I being unreasonable? I suppose I was expecting something special given the price difference of @£400.


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #991952 - 08/06/12 01:54 PM
Hey All,

Sorry for not being around much lately.

@ Goddard,

Re WaveRT , only Sonar of the major is using WaveRT and not many audio interface developers , from my understanding performance wise its not up to the performance of ASIO, so no real reason to go there.

Re Marian, one step ahead of you, I have already asked the local reps for a test card, just waiting for an available unit.

RE the AD/DA testing, yeh thats a whole other can of worms I suspect, the SOS reviews sadly have not gone into the heavier details for quite a while now , pretty much since Martin has stopped doing the interface reviews , IMO :-(

which has been attempted over at Gearslutz with a huge Converter Shootout. I can't really comment on the methodology of the test over at G.S except to say I have my own reservations, in the end , use your own ears.

BTW: I haven't direct linked but its in the Shoot Out area of the forum, so easy enough to find.

Peace

V.C


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Goddard



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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #992161 - 10/06/12 08:37 AM
Quote TAFKAT:

@ Goddard,

Re WaveRT , only Sonar of the major is using WaveRT and not many audio interface developers , from my understanding performance wise its not up to the performance of ASIO, so no real reason to go there.

Re Marian, one step ahead of you, I have already asked the local reps for a test card, just waiting for an available unit.




Hi Vin,

Well, as a Sonar user, WaveRT is of interest here, so was kinda hoping... Of course, only relevant for PCI/PCIe interfaces, but the potential is there for some seriously low latency.

Look forward to what you find re Marian.

Cheers!
=Goddard


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TAFKAT
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Re: Audio Interface - Low Latency Performance Data Base : new [Re: TAFKAT]
      #1004150 - 18/08/12 01:19 AM