Low Latency Interfaces

Hello all,
I'm new to Logic. (and computer recording in general) Although I have been in studio's that used pro tools and stuff like that and the biggest problem I have is latency when doing overdubs.
I recently purchased a Macbook Pro (2.5 penryn, 2 gigs ram, 667 frontside bus, hdd@5400rpm)
I'm wondering if anyone could suggest a firewire interface, preferably with at least 4 inputs with phantom power, but no less than 6/8 inputs altogether. There would be alot of track by track recording. I'm looking to spend about $700usd or $900cdn.
So any help with the following questions would be great.
1. How detrimental is the rpm speed of my hdd, If I use a fast firewire external is it even an issue?
2. The dsp mixer software that motu uses for zero latency,
(a) Will that work in real time with logic?
(b) Can i use plug ins while that is running?
(c) What other similar hardware comes with these latency solutions?
I'm not going to lie to anybody, I'm really green when it comes to this stuff. If anyone could help with any of these concerns I will buy you a virtual shot of Jagermiester.

In your price range, +1 for MOTU. If you want 4 preamps, check out the Traveler.
1) 7200 rpm should be fine for most needs.
2a) Yes
2b) If you use direct-monitoring, you can't hear Logic's plugins on the tracks you're recording
2c) Almost everybody now has a direct-monitoring solution, whether it is Apogee, RME, MOTU, etc. Hey, even M-Audio do that!

Similar Messages

Low latency audio interface, firewire or usb? and wich one would u recommend?

Low latency audio interface, firewire or usb? and wich one would u recommend?

Around 600, only need 2 xlr inputs.

Best soundblaster solution for low latency recording/monitor

hi, i have an emu 0404 that i use for my small desktop project studio, and i am happy as can be with it. i wanted to get a soundblaster to use for my laptop, and i've been looking at the audigy 2 nx, or the newer audigy 2 zs notebook. my question was, as both of these use asio drivers, is there any difference between them, performance-wise? does one have generally lower latency than the other? im not so concerned about the i/o on these cards so much as the latency. also, i would like to be able to use them for 5. surround mixing, which is the better bet for me? i specifically want a soundblaster and not an emu or some other higher-end usb interface because i want to use the soundblaster for gaming, music, movies, etc. so which should i get, the nx or the zs?
thanks in advance
mike

El_Pepe,
You will get the best performance from the Audigy 2 ZS Notebook. The PCMCIA connection has a lot more bandwidth than a standard USB 2.0 connection and allows for more features. The Audigy 2 NX is a great sound device, but if given the option I would select the notebook over the NX.
Jeremy

Low-latency multi-I/O - Is It Essential?

Hi,
On Apple's Logic Express 8 Technical Specifications page, it says "Low-latency multi-I/O audio hardware and MIDI interface recommended".
My question is, is a Low-latency multi-I/O audio hardware and Midi interface essential to use Logic Express properly? Or is it only necessary if you're doing live recording of instruments? If you're just using Logic's internal virtual synths do you still need one of these interfaces? I was under the impression Logic Express worked just fine on a MacBook Pro with no extra hardware required.

I don't know if Apple specifically recommends any i/o device, but I use products by Echo Audio. You can look them up at http://www.echoaudio.com. I personally have the Audiofire12, but if you are wanting a smaller i/o they have a few different models, Audiofire2, Audiofire4, and Audiofire 8. They are great products, I have been using them for years.
-Tyler

HP Elitebook 2530p RAM - Kingston HyperX low latency -- not compatible?

Hello, I would be grateful for help! I tried to install the following RAM in two different 2530p Elitebook:
Kingston HyperX --- KHX6400S2ULK2/4G --- These are two moduls, each 2Gb.
I tried it out with only one of 2Gb or both of them, but the notebooks did not boot. Both tested HP 2530p motherboards are model 30E1, version F12, Intel chipset GS45, revision 7. -- One has SL9600 processor, one has SL9400 processor. The operating system is Win XP Prof SP3 / 32-bit.
This is DDR2-6400 RAM with similar specifications to the original RAM (which works fine), but ULTRA LOW LATENCY.
>>>> QUESTION: Is there any compatibility problem known with ultra low latency RAM and HP 2530p?
>>>> OR is the problem that I am using a 32-bit operating system? And this is ONLY for 64-bit OP?
KHX6400S2ULK2/4G
4GB (2GB 256M x 64-Bit x 2 pcs.)
PC2-6400 CL4 200-Pin SODIMM Kit
Thanks a lot, I will try to help with other issues I already experienced, now we have three 2530p in my family.
Christian

After reading over the unofficial guide more carefully, i cam accross a link to kingston homepage about a related issue with the KHX3200/xxx modules: http://www.kingston.com/support/faqs/memory/intel_3.asp
This has led me to believe that the KHX3200UL/xxx modules will infact not work, as they are more vigerous than the KHX3200/xxx modules. According to the second solution of that link, i have purchased 2x KHX3200A/512 modules (same ram with slower timings) which, according to kingston, should work with i865 boards. Is anyone able to confirm this?
Thank you again

Low latency mode doesn t work?

If I enable "Low latency Mode" nothing happens, Ioaded tons of Plugins and set the limit to 1 ms....Shouldnt it bypass some plugins? Someone who knows the problem

does it work for you?

Ping from lower security interface to a higher

Hello,
I have a Cisco 5520 ASA firewall with a direct connection to a Checkpoint firewall. On the inside network of my ASA i have a server that needs to ping a server on the dmz on the Checkpoint and vice versa. So i have the correct routing and firewall rules on both devices.
I can successfully ping from my server on the INSIDE interface on the cisco asa to the server on the DMZ on Checkpoint but i cant ping in the other direction.
Q Is this because i am trying to go from a lower security interface on the asa to a higher one?
I cant be sure if the error is on my asa or the checkpoint because neither is showing anything in the logs?
Everything else on both firewalls is fine.
regards,
Kevin

Hi,
Its hard to tell what the actual problem is at the moment.
With regards to the "security-level" value, the situation is if the interface doesn't have an ACL configured on it then traffic sourced from networks behind it will be allowed to networks located behind interfaces of lower "security-level". If the source interface for the direction that is not working doesnt hold an ACL and has lower "security-level" than the destination interface then you will have to configure an interface ACL to allow this traffic.
Then again, the problem might be as simple as the server simply rejecting the ICMP Echo but allowing itself to ICMP Echo some remote destination and receive an Echo Reply for that. In other words, the server can ICMP remote hosts but wont accept ICMP Echo from remote hosts. It might reply to hosts on the directly connected network. So if there is no clear reason for the traffic to not go through I would consider checking the server software firewall.
It might also be that the working direction has been configured with Dynamic PAT and there is no correct translation for the other direction to enable sending ICMP to the server.
You can easily test the ASA configuration with the "packet-tracer" so that would be the first natural step to determening the reason of the problem or atleast narrowing it down.
packet-tracer input icmp 8 0
In the above command you would use the interface nameif behind which the ICMP Echo is coming from (8 0 = ICMP Echo). The source IP address is obvious. The destination IP address should be the NAT IP address of the server IF there is NAT being performed. If NO NAT is done for the destination then you naturally use the real IP address.
Hope this helps
- Jouni

Why do I have to check low latency every time I want to record 1 track

I-Mac, 3.06 GHz Intel core 2 duo. 4GB RAM. Logic 9.1.1 Mac 10.6.4, Apogee Duet.. Theres no need for this in my main studio. Mac Pro, Motu 896. Even with 8 tracks recording.

OK here ya go. 2 different versions of logic and 2 different versions of SL. Every time you close and reopen a project. You have to check low latency again. I guess I just never noticed it in the main studio because I never needed it, Using a Mac Pro, Motu 896, Buffers set @ 512 Medium With multiple fire wire devices, Multiple midi devices, Recording 8 audio tracks at the same time. Never a problem. On the I-Mac With Apogeee duet. 1 Track with no plugins. Buffers @ 128 I need it. Go figure. Thanks for your input everyone. Bill
Message was edited by: cowbell bill

Low latency ram...worth it?

Hey everyone,
I'm thinking about upgrading my RAM, and one of the options is RAM with low latency. Does it make that much of a difference? Cutting down on latency for recording music was the main reason I was thinking of upgrading my RAM in the first place.
Thanks...

To save you the trouble, CL 4 versus standard CL 5 RAM wins every benchmark, though not at amazing numbers. So if the CL 4 RAM is only a small amount more then yes there are slight advantages lower latency. I use CL 4 RAM and things are much snappier, though mine is a MacBook which relies heavily on RAM due to the integrated graphics.

Low latency, high throughput RMI, Any suggestions welcome.

I am working on a module which is designed to support low latency synchronous RMI and high throughput asynchronous RMI.
It uses TCP, NIO, custom serialization, custom RMI, Java 5 concurrency thread pools.
Its current latency is 5 to 15 micro-seconds slower than ping times (depending on the arguments/return values). For throughput, it gets 80K-140K calls per second on a single channel.
I have looked at JGroups and Stream.
I was wondering if any one has tried this sort of thing, what suggestions you might have and what open source libraries I might use for compassion.

ejp wrote:
Using custom serialization most messages are under 256 bytes, if I send smaller packets of 128 bytes, it doesn't make much difference.I wouldn't expect it to make any difference. An Ethernet packet is around 1500 bytes. Whatever amount you sent below that is still an Ethernet packet.The maximum packet size is around 1500 bytes, however it won't send more data than needed. If you send ping with different packet sizes you will see different average latencies.
I have used the generic RMI Proxy class, and use simple Reflections to lookup methods to call on the server. I believe I can cut 2-4 microseconds if I generate my own proxies and method callers. This is a bit of work generating byte code for not much saving.You will save time by not using reflection at all.Correct, but for a generic implementation I need to use something.
I was also wondering how much work had been done using non-blocking RMI calls.RMI calls aren't non-blocking. Even if the methods are void they still have to return a a success/failure indication, e.g. 'true' or an exception. You can make them asynchronous by using more threads, but that's more cost.This is a custom implementation, not the built in Java RMI which is much slower (and a pain to use IMHO). When a callback is provided as the first argument, the result is processed in another thread as the results becomes available. In this way it can support many concurrent requests over the same channel. Yes, an additional thread is required and there is a small impact on latency, however asynchronous calls can reduce the impact of latency on an application.
CORBA supports async calls and you may be able to get at that via Java IDL and idlj. Not with RMI/IIOP though.
This is how I get the high throughputs on a single channel.But you're not in control of the number of channels. RMI will use as many channels (i.e. TCP connections) as it needs to execute whatever remote calls are in progress. Making your own calls 'nonblocking' or async doesn't really affect that.I wrote the custom RMI so I make sure I control the number of channels. :) In fact I use different channels for synchronous, asynchronous calls, and asynchronous events (where the server pushes events to the client) as required.

Low level interfaces to JVM

Hi. I'm trying to write a java debugger in java. Sun mentions publicly available low level interfaces to the JVM for getting complete info on classes (ie things like private variables)
see the end of http://java.sun.com/docs/books/vmspec/2nd-edition/html/Overview.doc.html
I've been searching this site for "low level interface" but I haven't been able to find any information on this topic. Has anyone had any experience with this? Can anyone help me out?
Thanks,
Tom

There are a number of free publicly available
debuggers. You can get hold of their source for a
few ideas. e.g. Eclipse, NetBeansThanks for this peter. I think you're missing the point though. I am trying to build my own debugger so that I can precisely control the environment, not to do typical debugging. Even using an open source project, I think it would take me longer to try to fit in what I want to do into that than to build my own.

When I press 'Low Latency' button I only have audio coming through left channel ?!

When I press 'Low Latency' button I only have audio coming through left channel !?
When you take it off 'Low Latency' mode the audio is fine and comes out Right and Left speaker.
Can anyone please advise why this happening and how I can fix it ?

I think Pancenter is trying to get you to read and understand the "Working in Low Latency" section of the manual as the results or problems with working in Low Latency mode are variable.
This section, for instance:
Note: The sound may change in Low Latency mode. Depending on the plug-ins in use, the changes can be anything from subtle to dramatic. If plug-ins being used do not exceed the total latency limit, there will be no audible difference.
If it was me posting about this issue I would now come back with what exactly is inserted/bussed on the tracks because it could be an accumulative problem based on what plug-ins you have active and if they are on audio tracks (stereo or mono), instrument tracks etc. Lots of variables. The section of the manual quoted above won't give specific instances as any user could be using any combination of any amount of plug-ins.
So more information is needed before anyone can help you any further.

Low-latency audio on Windows Phone 8 (hint: forget it for now...)

My business specializes in audio and music apps for the Windows ecosystem. For this new project that I’m considering (a virtual instrument of sorts), I need to achieve the lowest possible audio latency from capture to render.
A measured latency below 3 ms would be ideal, 20 ms would be ok, and anything above 30 ms would be a deal breaker. And just to be precise about my definition of latency, I’m strictly talking about “mic-to-speaker” latency (includes the processing in-between).
I’m not talking about “glass-to-speaker” latency, although this topic is also of interest to me.
I figured that I should be using WASAPI, as this is apparently the API that sits at the bottommost of the audio stack in WinRT / WinPRT. So I spent a couple of days re-familiarizing with C++ (happy to meet an old friend)
and I coded a prototype capturing, processing and rendering audio using WASAPI having in mind the goal of achieving the lowest possible latency. After trying and comparing various methods, workflows, threading mechanisms, buffer values and mix formats, the
best I could achieve using WASAPI on Windows Phone was ~140 ms of latency. This is using 3 ms buffers. And although the audio is glitch-free, I could have been using 30 ms buffers and it wouldn’t have made the slightest difference! The latency would still
be around 140 ms. As I found out, the API/driver is extremely defensive in protecting against audio starvation, so it adds quite a bit of buffering at both ends. This is very unfortunate, because it basically disqualifies real-time audio/musical
applications.
I’d love to be able to provide quality audio/musical apps for the platform (both Windows Phone and Windows 8), but right now, this latency issue is kind of a deal breaker.
I've been pointing out the importance of low latency audio to Microsoft for quite a while, and I know I'm not the only one, and I know a lot of people at Microsoft realize this is important. But in its execution,
it seems Microsoft constantly fails to deliver a truly low-latency audio stack. In the pre-XP days, I've had talks with the sysaudio devs about this, and I was told, "yeah, we're working on a new architecture that will come out after XP and
it will solve the latency problem for all audio and musical applications." Fast forward to mid-2010 (pre-Windows Phone 7), and I was still there pointing out the horrible latency figures one would get from the APIs that were about to ship. And
now that WASAPI is available on WP8 (our best hope yet for low-latency audio), I discover the overly defensive and buffer-happy architecture of WASAPI (even though one of its promises was precisely low-latency audio).
So, the question is…
Is Microsoft aware of this issue? If so, is Microsoft giving up and simply conceding the pro-audio territory to iOS? If not, I’d be glad to discuss this issue with an engineer at Microsoft. I’m serious about bringing audio apps
to the platform. I just need some assurance that Microsoft is taking action on its end, so that I can sync my development with the next product cycle.
Thanks in advance for any help, advice, or insights!
/Antoine

Any update on this issue? Is there a roadmap for a fix?
I ported a realtime sound analyzing application (requiring both audio input and output) to WinRT and was quite surprised at the high latency from WASAPI. I then searched around and posted some questions regarding this on various MS forums and got to the
following conclusions based on feedback from people including MVPs and MS employees:
A lot of those people are under the impression that WASAPI qualifies as a low-latency API.
Apparently WinRT apps are not supposed to have high CPU usage, and this is purposefully baked into the framework (minimal thread priority control, async/await everywhere which results in thread priority inversion, and other issues).
Why would anyone ever need high-priority threads? They must use a lot of CPU, right?
A lot of people think low-latency audio means high CPU usage. You can see where this is going when you look at the previous point.
Async/Await is being forced down to all levels, even though it should only be used at the UI level. What some people are actually now calling "old-school" multithreading is now being pushed out (lock, etc). Async/Await has horrible overhead and
results in thread priority inversion amongst other issues. For example, witness that there is no StorageFile.Read, just StorageFile.ReadAsync. Do some IO benchmarks with some of these async methods and you will see some horrible performance compared to desktop
file IO.
To get an understanding of what low latency audio means and why it is important, see this video. It compares the latency of Android to iOS using the exact same music app. Ever wondered
why there are no quality music apps for Android? Well now you know. And then realize that WinRT has
twice as much latency as Android.
And if anyone thinks this is a niche use case, consider that Apple created adds showcasing "musicians" playing their iPad "instruments". A use case that is essentially unavailable for WinRT apps. Why would Apple create
adds for a niche use case?
This should have been one of the high priority issues solved from the start in WinRT. MS solved this issue in Windows (desktop) long ago, with the ability to get insanely low latency there (0.3 ms in some stress tests,
see here), even beating out OSX. It is as if there is a new generation of architects at MS that know nothing about this previous work and are doomed to make the same mistakes over again. I really don't
understand why these pre-existing APIs can't be exposed in WinRT. No need to re-invent the wheel. But I guess it just isn't important enough.
I find this situation really sad since otherwise it could be a great and powerful platform.

Corsair twinx3200 low latency or twinx4400

which mem best do you think
i had 3200 in sys for a while o.c only to 210x12
tight timings on mem 2.3.3.6
good b/mark 5600+
put in 4400 clocked to 220x12 stable
looser timings 3.4.4.7.
benchmark drops to 5100 argh !!!!!!!
now i know 3dmark 05 is gpu not cpu intensive
but with gcard at same mhz why am i dropping scores ?
i would have thought fsb at 220 would give more speed than 210 i.e
440 mhz on ram / 420 mhz on ram
and
880 fsb / 840 fsb
tried upping cpuv to 1.7
mem v @ 2.8
agpv 1.6
agp locked @68 mhz
and yep kingfisher have read your guide again thx
( but i think im still missing something )
cant seem to get over 225fsb stable.
mem should take 550mhz i.e 275 fsb.
what is holding me back ?
tried 11x multi in bios get higher fsb but cpu wont go over 2700mhz
what effect will or what does 4xht do ?
am i right to think it is multiplying fsb speed i.e 4xht = stock 200fsb 800 mhz
3xht = stock 200fsb 600 mhz
if so if i drop this to 3 when i o,c cpu im not going beyond my boards fsb of 800 stock by to much .
confused you soon will be arrrrrgh pulling my hair out
ok kingfisher gonna read it again i know i know rtfm.

Well, I'm not AMD64 user,
but I think if you can go to FSB 275 with AMD64... woww, it's so amazing! (am I right?) Do you really need PC4400?
Just go for Twinx XL, as it is rated with tight timings (2-2-2-5) at stock and can overclock very well (I think timing by SPD @ FSB 250 is 2.5-3-3-6, pretty good huh? But mine 2.5-3-3-5 ).
So you have wide range of FSB with PC3200 price.
You may not only choose Corsair TwinxXL as other also have release their "extreme low latencies and wide range FSB", like Crucial Ballistix and other, check them at Anandtech, Memory section, DDR 2-2-2... or just click this
NB: yes, as I know latency can play some role in 3DMark. And FSB 220 with timing 3-4-4-7 is verrryyy bad! It'll be very very slowww.

ASIO Low Latency Driver sound output is bad?

My Notebook comes with "build-in" asio drivers called "Generic Low Latency ASIO Driver". Further there is ASIO DirectX FullDuplex Driver and ASIO Multimedia Driver.
Sound card is: IDT High Definition Audio CODEC.
Running Win7 x64 (HP ProBook 6550b).
All drivers are the most recent ones.
Problem: 'Generic Low Latency ASIO Driver' sound output is much worse than using the other ones (there the sound is really great in both of them, but they have way to much latencies, so I cannot use them in cubase or any other music daw).
So I'd like to use the (standard build-in) Generic Low Latency ASIO Driver. But the sound is much worse in contrast to the other drivers! It sounds a bit like a phaser or gramophone.
Just to clarify: there are no glitches or stuttering in any case. Just the sound-"quality" is that messed up...
I'm getting really upset about this. Tried the recent version of the ASIO4All 2.10 driver, but the problem remains! The sound with asio4all is exactly as bad as with the buildin Asio Low Latency Driver.
How on earth can I achieve low latency with good sound in Cubase? Has anyobne experienced things like this before, or might there be any fix for this?

Well, it's just the standard driver that comes with my notebook, named HP ProBook 6550b.
http://h20000.www2.hp.com/bizsupport/TechSupport/SoftwareDescription.jsp?lang=en&cc=us&prodTypeId=32...
I did not take control over the buildin audio driver installation, because win7 installs these drivers innately.
The problem does not only occure in Cubase, but in any music tool/program/player where I set the driver to low latency (which obv does not make sense in others than cubase, but for testing I found out that it is not an application related problem).
It is really strange. The soundcard seems to provide real asio, but the low latency sounds as bad as the "pretending" asio4all. hm...

Low Latency Interfaces

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