There you go! That's more what we want to hear after our long and dreadful eulogies in that sad latency thread. I have one Digitech driver, an RP500, that gets 8 ms round trip. Now, that's loaded into REAPER or Zynewave's Podium in the usual manner on a standard audio track.
Unfortunately, that ASIO does not always play nice - can lock up my soundcard or whatnot and the DAW will sometimes freeze on exit - so I most often use ASIO4ALL v2 in its stead with that favourite interface. Using ASIO4ALL, round trip is anywhere from 12 to 16 ms I think.
Focusrite is a brand that is well known for their high-quality studio equipment, with models ranging from entry level to professional. Their Scarlett 2i2 audio interface offers an easy and straightforward approach to recording audio for everyone looking to make their own home studio.
I would greatly prefer all latencies remaining below 10 ms round trip; however, I can live with 16 ms if I have to, since I don't anticipate using it professionally live for months and months, if ever. Now, I recall clearly saving a minimum of 1 ms each direction in 64-bit standalone. Obviously the signal doesn't have to pass through or deal with any aspect of the DAW. So, I should say, Congratulations! I know your latency troubles have been a real bugaboo for you. I have no FireWire, and as I said some time ago, will undoubtedly be skipping that and heading for USB3 with the next PC build.
Won't you be heading ASAP for Thunderbolt when the opportunity arises? Thunderbolt is supposedly the new state of the art for Apple, correct?
Have you been able to put your hands on it for a test? I wonder how much better and faster it is over next-best FireWire.
I'm all questions here because most folks in the know think that Apple will license out Thunderbolt sooner than later, and they will proceed to rake in yet another fortune in doing so. Please do keep us updated on this new technology, if you will, and if you hear anything on this. I would jump on the band wagon, only if it becomes available on a PC. Meanwhile, because it sounds like you have the 'headroom' to do it, I strongly suggest using that 96k if it will be accepted throughout and your DAW/Mac can handle the higher processing hit. Anytime I've heard from anyone who has come out of an interface in 96 has had only praise and good things to say about the setting. I hear remarks such as, 'really opens up the instrument's sound' and 'it lets my guitar breathe!'
The Telenator wrote:Have you been able to put your hands on it for a test?Not as of yet. But probably will later on. When time arises, and spare time allows me.The reason I had this amount of time, is that on my day job, one week each month is on 'duty call', i e I have to be at standby mode at home, and can't do anything that can't be aborted within a split second. A perfect waste of past time is something like these nerdy tests. If someone calls and I have to rush to work, I can abort whatever I am doing, since it is - definitely not - that important. Sort of 'I have to do something boring anyway, because if the call comes.'
The Telenator wrote:I hear remarks such as, 'really opens up the instrument's sound' and 'it lets my guitar breathe!' As for 96 K, I can't hear it on guitar that much, but maybe some other things. When doing WAV recording and tracking I record on 96 K and 32 bit. And then work on it internally. Just at the last minute I bring it down to 44.1 Khz and 16 bit. I have not heard any real acoustic advantages within my hearing range with 96 and especially not from S-Gear, which I think sounds great at 44.1.I think it depends on the soundcard. Good quality soundcards can sound just as great on 44.1 as on 96.
If I have to use S-Gear as a VST plugin when tracking, of course, everything has to be set up like the rest of the tracks accordlingly. My hearing range isn't above 14 Khz anymore (at all, at no level) so the highest top end is gone anyway.
I can hear and do mixing and mastring quite well, but I have trouble in hearing if anything is 'wrong' with the track, say a stray signal from something that shouldn't be there, is above 15 K I can't hear it. I have to look at it instead. ON USB3: 'These delays may be due to problems in the CMOS manufacturing process,a focus to advance the Nehalem platform, a wait to mature all the 3.0 connections standards (USB 3.0, PCIe 3.0, SATA 3.0) before developing a new chipset, or a tactic by Intel to favor its new Thunderbolt interface. Announced laptops with USB 3.0 ports on June 11, 2012, nearly four years after USB 3.0 was finalized.
Because Apple computers use only Intel processors and 'bridge' chipsets, Intel's lack of integrated support for USB 3.0 may have proved to be a primary reason why the company didn't add support sooner.' And further reading: 'There have been many reports of USB 3.0 equipment only transferring data at USB 2.0 speed, usually with a message 'This USB Mass Storage Device can transfer information faster if you connect it to a Super-Speed USB 3.0 port'. This has been due to several causes, including drivers, certain cables specified as USB 3.0 (problems disappeared when a different cable was used), order of starting equipment, equipment needing to be disconnected and reconnected, and overclocked computers. All major test equipment vendors offer electrical compliance test tools meeting USB 3.0 electrical compliance. Electrical testing requires USB 3.0 test board provided type A, B, mini AB electrical compliance test breakout adapters' The order of starting equpiment, and restarting, and be reconnected pisses me off big time, as for now. We've all been there done that.
Better wait until all things have been sorted out. As for Thunderbolt, their hybrid copper/optic cables are way too expensive. The main thing with thunderbolt was ability to carry POWER over it, 10W, and connect units. However, it seems that you can't use long cables anyway.
If you power your hardrives with dedicated power supply, length is not an issue. USB 3.0 sends just 4.5 Watt over the cable, FireWire 40 W, and Thunderbolt 10W. As I see Thunderbolt, it's a sort of PCIe performance moved out to peripherals instead, with an appropriate cabling and connecting.
I do still think that Thunderbolt has huge improvements in transfer speed rates over both FW and USB3 but I think that it matter most when trying to record 48 tracks at once into any computer. You're always hindred by the internal roundtrip latency, should you waste your guitar playing on an external thunderbolt soundcard interface.
WHile USB3.0 definitely has its advantages, it is still loading the cpu, no matter how little. FW does not tax CPU no matter how much. As for now, I think Firewire still suits the music production the most in my opinion. At least on a MAC.
I agree that the market will probably shift in a year or two and doesn't allow anything to be built with FW anymore. It's USB3 or TB anyway. Right now it's still in its infancy.
Well, thanks for all of that! I see a couple of points I was completely unaware of, and a couple I'd forgotten. I know I could have looked all of that up myself, but I am still about a year away from making a serious move, so I tend to not research until the time comes nigh. I completely agree concerning the virtues of FW, but as you yourself went on to include, it is the market and the makers that will tell us what we will be using in future times.
Nothing new, of course! All right then. Now that you have just about scared the pants off of me with the gruesome details, I suppose I'd do best to ignore the whole mess until I have no choice.
It's all to do with the desktop build - looking at probably this time next year. As I understand it - upgrade to a USB 3.0 or Thunderbolt won't actually improve latencies. Sure, usb 3.0 and Thunderbolt can afford a lot more in terms of BANDWIDTH, but that doesn't mean that they are faster in terms of response and latency.
As for the bandwidth - Apogee even stated on their site about Apogee Quartet - ('Why not usb 3.0? Why not Thunderbolt?' ) that most interfaces (I don't speak of 64in 64out studio madness here) just doesn't need that much and usb 2.0 is enough. Imagine it as compared to the internet connection. Yeah, sure you can upgrade from 5 Mbps to 10 Mbps, but that won't improve your PING. So, as I see it - the only way to improve RT (roundtrip) latency in external devices this days - is try to get rid of buffers and latencies inside the interface.
Like AD-DA converters and such. As I understand - that's exactly what RME are trying to do with their later solutions. Upd.: Someone from Texas Instruments stated, that, actually Thunderbolt CAN lead us to a latency improvement. Said it's about providing multiple streams, a thing that USB 3.0 can't do. Anyway, even if external interfaces won't get better in terms of latency - there's a chance that we'd be able to connect like Laptopthunderbolt to PCI-e dock PCI-e interface.
Hi Guys, I'm about to replace my trusty Yamaha n12 with a dedicated firewire audio interface. My main requirement is super-low latency for running BFD and other virtual instruments, along with monitoring through Logic directly sometimes (to add effects to vocals in real-time, etc). I've narrowed down my choice (I think!!) to: - RME Fireface 800 - Apogee Ensemble - MOTU 896 Mk3 I have been told that the RME drivers are incredible, and hence it has the lowest latency. I'd love to hear from other Logic users who own any of the above! What kind of latency do you get (round trip in ms)? What is the lowest buffer setting you can use?
Any downfalls or problems you've had with any of these interfaces? Thanks heaps guys, can't wait to hear some opinions!
Cheers and thanks, Mike. I've used several MOTU interfaces and the Apogee Ensemble. If you're looking for low-latency with BFD, forget the Ensemble. It sounds great, but latency is a weak point.
I can run a 64 sample buffer with BFD and MOTU Firewire interfaces at 96k, but with the Ensemble, 96k required a 256 sample buffer with BFD on the same machine (older Mac Pro 2.6 quad). Not to mention that Ensemble has significantly higher latency at any given buffer setting. On the other hand, I'm not a big fan of the MOTU interfaces because 1) they have flaky S/PDIF inputs at 96k (confirmed this problem with tech support at Universal Audio), and 2) MOTU has an aliasing bug with SMUX ADAT at 96k (I've reported this to MOTU multiple times with no reply). However, for low-latency Firewire, MOTU is definitely a contender.
But I've never compared directly with the Fireface 800-I'd be very interested if anybody has directly compared the latency of a MOTU device with RME using BFD as the VI. (Seeing as how the V.P. Is such a V.I.P., shouldn't we keep the P.C.
Lowest Latency Fw Audio Interface For Mac Mac
'Cause if it leaks to the V.C. He could end up M.I.A., and then we'd all be put out in K.P.) -James.
Yeloop wrote: Interesting to hear that the Ensemble is so ordinary when it comes to latency. For some users this would not be important, but for something like BFD, latency is a massive issue. I spoke with a tech at Apogee about the Ensemble's latency-I was surprised there was so much and wondered if there was a problem with my system. The answer I got was that Apogee doesn't write the driver for the Ensemble-it uses Apple's generic (tweaked for Apogee?) OS X Firewire audio driver, and Apogee is aware the latency is high. I was told there were no immediate plans for Apogee to write their own driver, and they weren't aware of any impending improvements on Apple's side.
Their official word is to use Maestro for low-latency audio, and if you need low-latency VIs, buy the Symphony system instead. Is your experience of the MOTU devices the latest 896Mk3, or the Mk2 versions? The Mk2 versions. Would be surprised if the latency is significantly different (the Mk3s appear to use the same drivers), but can't confirm that. Sampleconstruct wrote: Monitoring through the RME itself gives you 0 latency (could be one sample or so) and the Total Mix Software is extremly versatile for all routings you can think of. Thanks for the info on the Fireface 800, and I'm sure it's a great piece of gear!
But, I think you're probably measuring the Total Mix latency incorrectly-the Fireface converters almost certainly have more than a few samples of latency. For instance, I did some latency measurements comparing the MOTU Traveler Mk2, Metric Halo 2882, and Apogee Ensemble, using 'direct monitoring' on all three devices: 2882 (MIO): 75 samples (1.70ms at 44.1k, 0.78ms at 96k) Traveler (CueMix): 73 samples (1.66ms at 44.1k, 0.76ms at 96k) Ensemble (Maestro): 22 samples (0.50ms at 44.1k, 0.23ms at 96k) The Ensemble's 22 samples is the lowest analog in-analog out latency I've seen on any digital device, so for direct analog audio monitoring (i.e.
Not monitoring through software), it's great. When I compared using external A/D/A converters routing via ADAT or S/PDIF, I got similar values for all three devices, so it appears the Ensemble's converters have lower latency than the other two devices, and the latency of the mixing itself is similar. For comparison, ProTools HD reports 105 samples of latency at 44.1k (software monitoring through ProTools). AFAIK direct monitoring is not available with ProTools HD. Check out the RME's latency again, if you get a chance-I'm guessing when you measured the latency, you maybe weren't getting the full roundtrip analog in to analog out path.
Thomjinx wrote: If yeloop is using BFD, then direct monitoring isn't going to help him. That is for audio in. FWIW, on MOTU Ultralites, I get a little (less than 1 ms) less latency with the Mk 1 rather than the Mk 3. Perhaps the built-in DSP slows the Mk 3 down? Interesting-thanks for the info (one more reason not to upgrade MOTU, frankly.)! It is still 4.3 ms at 32 samples with the Mk 3.
For MIDI controlled VIs, it would be less than that. I think I measured 3.9ms with my Traveler mk2 at 44.1k with a 32 sample buffer. So, it's a bit faster than the mk3. At 96k I can use a 32 sample buffer to get 2.5ms latency, but I can't run BFD2 with a 32 sample buffer.
Best I can do with BFD2 is a 64 sample buffer at 96k. But, you're exactly right, Thomjinx: with a VI, you won't hear that full amount of latency. Assuming the MIDI latency is negligible, the VI response time will be roughly 1/2 the full analog audio latency. (Logic reports both of the these figures-output latency and full roundtrip latency-if you click on the latency info in the Audio preferences. Be aware, though, that some devices report this information correctly, while others do not. For instance, with the Apogee and Metric Halo products I've used, the numbers reported by Logic are exactly correct, while with MOTU, the numbers shown in Logic are lower than reality: with a 64 buffer at 44.1k the Traveler mk2 indicates 4.3ms roundtrip latency with Logic, but I measured it as 5.6ms. So you may be misled if you try to use the latency values reported by Logic to compare products.
In that regard, MOTU seems to cheat, while Apogee and Metric Halo follow the rules 🙂 -James. Thanks for that info, James. I was not sure how accurate Logic was reporting, but I was assuming it would be at least be consistent. But apparently not. In any case, if one needs more than two ins and/or outs, the MOTUs offer a lot of bang for the buck. What I would love is an Apogee Mobile Symphony!
Currently I am running the Ultralite in a small 3 space rack, with two or three extra mic pres when I need to run more than two mics. For most of my live gigs, I just use two mics and use MainStage to change the settings on the mics. That way I can use one mic for Sax and harmonica, and the other for flute, pennywhistle, cornet and steel pan. EWI runs into a 1/4 inch line in. With the two-mic setup, I don't even need the rack. Thomjinx wrote: Thanks for that info, James.
I was not sure how accurate Logic was reporting, but I was assuming it would be at least be consistent. But apparently not. Technically, I think Logic itself is probably being consistent. The problem is that Logic relies on the latency reporting from the device drivers, and the MOTU drivers seem to underestimate their latency. On a related note, this is why Logic is able to place recorded audio from the Apogee and Metric Halo devices with sample accuracy automatically, while the MOTU devices will record audio slightly late (40-100 samples depending on device) when you first plug them in. The MOTU devices can record perfectly in Logic, but you'll need to do a loopback test and set the delay compensation manually.
Currently I am running the Ultralite in a small 3 space rack, with two or three extra mic pres when I need to run more than two mics. For most of my live gigs, I just use two mics and use MainStage to change the settings on the mics. That way I can use one mic for Sax and harmonica, and the other for flute, pennywhistle, cornet and steel pan. EWI runs into a 1/4 inch line in. With the two-mic setup, I don't even need the rack. I'm using the MOTU Traveler for live instruments, and it's working well, too. Definitely not the best-sounding box you can buy, but for size, features, and latency, it's a good match for live performance.
Jnashguitar wrote: Cool! I'm using the MOTU Traveler for live instruments, and it's working well, too. Definitely not the best-sounding box you can buy, but for size, features, and latency, it's a good match for live performance.James As a side note. Some of the touring Broadway shows, have been using MBP's, with Mainstage and a MOTU Traveler for the orchestra pit keyboards. This is what I would call mission critical as the show depends on these keyboard parts for efx, intros, musical cues and main backing keyboards. Apple Footer.
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