Latency UAD+EZDrummer
Hi, I'm having Latency problems with Uad-1 PCIe on Logic 7.2.3, OS 10.4.8 the Latancy issues occur when Toontrack EZDrummer is used as multi channel Audio Instrument and it's Auxilaries are processed by Uad plugs. The delays begin when I tweek the plugins on the different Aux channels. This is very strange as I've done this before with no problems. My plugin delay compensation is set to All. So I'm really a bit confused, I've also talked to UA and Toontrack and they don't seem to have a answer. Any help would be very welcome, TIA.
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Similar Messages
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UAD-1 + Logic 7.1 = Latency??
Hi all,
I just purchased a UAD-1 card, installed it and loaded UAD's latest software, 4.4.1. Upon first test it worked great, that was with one UAD plug on one vocal track. So last night I was itching to mix for real with this bad boy. But, as soon as I added a 2nd UAD track, and we're still only talking about on one track here, I start getting latency. Anyone had success with multiple UAD plugs on a single track sans the latency?
I did read on one comment that there was a bug with Logic where Logic's plug delay compensation only worked on the first insert when dealing with 3rd party plugs. Can this be true?? If so, would 7.2 fix this?
Also, anyone come across issues with 7.2 and 10.3.9?
Anyone think swapping my digi 001 card (which is in PCI slot 1) with the UAD card (slot 2) would make a difference?
Please help, right now for me to use my UAD card and have 3 simple plugs on one track, I have to bounce my track 2 times. So lame!
Oh, and when I look at the meters in both the UAD app and Logic both are around 20%, so it's not a CPU issue.I run multiple UAD plug-ins on individual tracks, and have no latency problems.
I'll assume you do have the Delay Compensation settings in Logic set appropriately...?
Also, if I read you right, you were still on Panther, and just ordered Tiger? That might very well be a contributing factor, although I'm only speculating.
Anyhow, UAD 4.4 on Tiger 10.4.8 has no problems with 2 or even 3 UAD plug-in in succession (and it was always fine on earlier versions of Tiger as well). -
Hi, Logic newbie...experiencing latency when using my UAD-1 card, Logic has ADC correct? am i missing something? thanks...
experiencing latency when using my UAD-1 card,
Yep, all UAD plugs will give latency.
Logic has ADC correct?
Yep.
am i missing something?
Firstly, you don't say whether you are talking about tracking latency, which you can't get around short of disabling UAD plugs, or mixing latency, which shouldn't be a problem for mixing as Logic will take care of it with ADC.
So, if the latter - have you got ADC turned on, and what is it set to? -
Logic 7 with UAD Card!!!
Hey... Is it possible or not to work with thw uad plugins while I'm still recording and workin on a song, is there a way to make it work without latency or do i have to wait until they re-write the program...???
Oh I get it. A wink helps.
Anyway...
Check your Plug-In Delay Compensation. It's under:
Logic Pro> Preferences> Audio> General
For tracking I have it set to "Audio tracks and instruments". (For mixing it's on "All").
Remove (bypass) all plugs on your Busses.
Bypass all plugs on your Outputs.
Lower you buffer settings to 128:
Logic Pro> Preferences> Audio> Drivers> I/O Buffer Size
Are you using more than one set of Outs?
Give that a try.
Check the UAD users group for info about which slot to use (I don't know, I have 3 slots and 3 cards).
http://www.chrismilne.com/uadforums/viewforum.php?f=3&sid=ca8010b83404f496dbe7df 177b61a4d9 -
Logic 7.01 - latency, to update or not
Hey everybody,
I have Logic Pro 7.01 It is working for me just fine. My question is this though. Does the versions or updates since 7.01 handle plugin latency better. For instance using an effect on UAD-1 card for tracking, like a compressor or amp simulation? Also, is there anything that I am really missing out on by going to the later updates? Thanks!"Also, is there anything that I am really missing out on by going to the later updates? Thanks!"
I meant, am I missing out on anything by not updating to later versions? sorry! -
Audio delay problem when using UAD plug ins on aux track
HI there,
I've been trying to route my drum tracks to bus 1, then send bus 1 to the aux 1 track also and apply UAD compression plug ins to it to mimick the NYC parallel compression technique.
But, when I do this, and add one of the UAD plug ins to the aux 1 track, the audio on that track becomes noticeably delayed by a few ms and you hear a kind of phasing/flamming sound as the bus 1 and aux 1 track combine slighlty out of sync. However, if i use logic's own compressors on the aux track there is no delay notivceable and it works fine.
frustrating as i'd like to be able to use the 1176 or lA2A on the drum track on the aux bus!!
any ideas?
J Ceetry this setting
I suppose that you need software monitoring for use Logic Compressor and reverbs in live monitoring mode
the setting is
Sample buffer 256 or 128
PDC set to Audio And Software Instrument...
this set if very good
I always use that with UAD plugins in Live mode on my professional studio
No latency at ALL
when Mixing set the buffer at 1024 and PDC to all...
this set is very important only in th Mixing Phase
feel free to ask for more advice
Cheers -
UAD-1e and Liquid Mix questions
Hey all,
I'm looking into 3rd party external DSP stuff, and the UAD-1e and Liquid Mix products sound compelling. I'm wondering if users of these have some advice as to what to watch out for, for instance, Liquid Mix's latency--does Logic adequately compensate for the delay? Any incompatibilities in 8/8.0.1/Leopard? Thanks.Looks like it was just a typo...
"Dear UAD Users,
We would like to clear up speculation and mis-information by making a formal statement from both UA & EL clearing up the following points.
1) UA & EL have signed a Trademark Licensing Agreement.
2) There will be plug-in partnership announcement at NAMM 2008 confirming that the EL Fatso will be a forthcoming plug-in for UAD-1.(ONE)
3) The expected ship date for the UAD EL Fatso plug-in will not however be until late 2008.
Apologies for any confusion and we wish you all a Happy 2008 in your music making!
Best,
Mike Barnes
VP Of Marketing
Recording Academy, ASCAP, NARAS Member
Analog Ears | Digital Minds"
Still - new cool plugins continue to appear - lots of people would like a Distressor plugin, and it seems that's a possibility now, as EL say there will be softwre versions of all their products...
...and it doesn't explicitly rule out the possibility of a UAD2 either... -
Low latency mode: what's the sense?
I've read it bypasses plug ins which causes latency. But if it is so what'is the sense of this function? If I understand well in this mode I don't know exactly which plug in will work and which no....
I cannot accept a situation like this.
Apart from anything else I did use low latency mode (because I listened some latency after having inserted a plug) and I checked all the plugs during playback and they all seem to work. So?
I don't understand...Usually I use Low Latency mode in when mixing-mastering.
I use UAD DSP plugins on the master... this is cause of many issue... the MAIN problem is that Logic is able to compensate Plugin Audio Latency (PDC)... but not Graphics .. (GPDC is not available... instead Logic 7... Logic 7 have GPDC!!! ... in Logic 8 graphical plugin delay compensation is a missing features)
the resulting behavior is that All meters show anticipate by the sound...
but this is not fixable.. or better.. Apple developers must included the Graphic delay compensation in order to get metering in sync with Audio... (I hope in the 8.1 version)
Anyway.. i love UAD sound and I use them in any case..
return back to your question, I use Low Latency mode when i need to record some missing details .. without get DSP latency!
It is a helpful features (not so basical but helpfull).
Logic Pro 7 is also a great DAW!
G -
Hi all,
I've been having a problem for a while, now. It affects both external midi instruments and also automation.
It is although there is reverse latency. The ext. midi instruments are playing early and the automation is heard earlier than it looks on the grid - i.e. if I draw in a plugin to be bypassed on beat 1 of a bar, it will actually turn off nearer beat 4 of the previous bar. Unfortunately it is not dead on a beat, however - so it takes a lot of trial and error to get the timing bang on.
In terms of the automation - I know that there are the settings in Preferences>Audio>General. I have Sample Accurate Automation set to Volume, Pan, Sends, Plug-in Parameters and I have Plug-in Latency>Compensation set toAll : However no adjustment of these settings makes a difference. Could it be something to do with the UAD-2 satellite card?
With the external midi instruments - I understand that Logic does not delay the midi being sent out for some strange reason. Is there any way to manually adjust for the latency? I use external midi instruments all the time in my projects, often using them well into the later stages of the projects when I have a lot of plugins in use and so a lot of latency is induced. Turning all the plugins off to reduce latency isn't ideal either because the sound of the synth I want to record in will depend on the mix, obviously, which will depend on the plugins. I often resort to bouncing the mix, and recording in a new project - which again, is not ideal.
Your help will be much appreciated!!
Alex
Mac Pro; OS X 10.6.7; Logic 9.8.1; Apogee Ensemble; UAD-2 Satellite Quad; Universal Audio LA-610; E-Magic MT-4Wasn't really suggesting you switch....
http://www.uaudio.com/blog/optimizing-uad-logic-pro/
http://www.logicprohelp.com/forum/viewtopic.php?t=79212
One of the replies in LogicProHelp suggests monitoring only through Aux channels. -
I am sure there is a simple solution to this ... I haven't found it on my searches yet.
In Logic 7.2 I am routing BFD auxiliaries to one bus and running some overall compression and eq. Now I have latency on the bus and it is lagging behind the rest of the tracks. What can I do to compensate and pull it back up to speed with the other tracks?I am sorry no one has seen fit to help you with this problem however help is at hand please take time to read this very carefully. I believe this is your answer you seek, The reason I have gone into such depth is simple so that you will have a grounded understanding of how logic address the problem and also what plugs in Logic have Latency within them by default. Well here goes
Logic Pro 7.1: Full Plug-in Delay Compensation can cause late recordings and increased latency
In Logic Pro 7.1, if you set Plug-in Delay Compensation to All, any new tracks you record will be late and out of sync with previously recorded tracks. You may also experience increased latency when playing Audio Instruments live. Here are some concepts behind the issue and tips for resolving it.
About full delay compensation
One of the new features in Logic Pro 7.1 is full delay compensation across busses, aux channels, and outputs. This means you can place plug-ins that normally induce latency on busses, auxes, or outputs, and the audio routed through these plug-ins will be in sync with everything. Logic achieves this by calculating the amount of delay caused by plug-ins assigned to a bus, aux, or output, and delaying the streams sent to other busses, auxes, or outputs by an appropriate amount to line everything up. This ensures that all tracks will be synced to sample-accuracy.
Example scenario
To give you a better understanding of what's going on here, let's say you have a simple song with a few drum, bass, guitar, and vocal tracks. All the drum tracks are routed to a bus with several plug-ins inserted that together induce 500 ms of latency. The vocals are routed through another bus that has a set of plug-ins that induce 250 ms of latency, and the bass and guitar tracks are routed straight to the main outputs without going through any latency-creating plug-ins. If latency were not corrected, then the bass and guitar tracks would play 500 ms in front of the drums. The vocals would be 250 ms in front of the drums but 250 ms behind the guitar and bass.
With full delay compensation engaged, Logic Pro will delay the streams sent to the outputs for guitar and bass playback by 500 ms to line them up with the drums, and will also delay the bus the vocals are streamed to by 250 ms to line them up with the drums and guitars. The precise calculations are handled for each stream automatically. During playback and mixing, the delay created to compensate for plug-in latency is imperceptible, but it can cause problems if you do any tracking after turning on full plug-in delay compensation. These recordings will be late because of the delay that Logic imposes to compensate for latency. The degree of lateness will vary, depending on which plug-ins are in the signal chain.
Eliminating latency
There are several ways to deal with these latency issues. The best way is to simply avoid setting up any signal routing that involves using plug-ins that create latency on busses, auxes, or outputs until you are done tracking. Once you start mixing, go ahead and set up any routing you like, including setting Plug-in Delay Compensation to "All."
If you can't avoid recording tracks after setting up such a routing scheme, then bypass any plug-ins you have on busses, auxes, or outputs that create latency, and change your delay compensation setting from "All" to "Audio tracks and instruments" while you are recording. You can quickly bypass plug-ins by Option-clicking on their inserts in the Track Mixer. You can toggle the Plug-in Delay Compensation setting by choosing Logic Pro > Preferences > Audio > General, or by setting up a key command for "Plug-In Delay Compensation: All." This key command will toggle the delay compensation setting between "All" and "Audio tracks and instruments." When you finish recording, re-enable the plug-ins on the busses and toggle the delay compensation setting back to "All."
Note: Bypassing plug-ins on Outputs will not eliminate the latency that they create. To eliminate it, you must remove these plug-ins entirely from the Output's insert slots.
If you are recording audio, another strategy would be to choose Audio > Audio Hardware & Drivers, then disable Software Monitoring. You would then need to monitor your recording via an external hardware mixer, or third party software mixing application that may be provided with your audio interface. When Logic is not providing software monitoring of incoming audio, then it can correctly position audio recordings even when full delay compensation is enabled. It is important to note that this technique will not help with recording Audio Instruments.
For the same reasons as described above, you may get increased latency when playing Audio Instrument tracks live if the delay compensation is set to All. As with recording audio tracks, try to complete any Audio Instrument recording for a song before turning on the full delay compensation. If you find it necessary to add a new audio instrument recording after setting up a signal routing that involves latency-causing plug-ins on busses, auxes, or outputs, then follow the same advice stated above for adding recorded audio. You may also notice that the meters on busses, auxes, or outputs that have delay-compensated plug-ins will not be in sync with the tracks that are routed through them—however, the audio will be sample-accurate. Additionally, the Song Position Line's visible position will not be compensated. Unless you are using an exceptionally large set of delay-compensated plug-ins, this should not be particularly noticeable.
Another effect with delay compensation set to All is that MIDI tracks triggering external sound modules will be out of sync. This is because Logic has no direct control over the audio output of external devices. A possible solution for this would be to route the audio outputs from the external MIDI devices to inputs on your audio hardware and monitor them through Logic. This way, the audio streams from the MIDI devices can be compensated during playback. Using Logic's External instrument to route MIDI to your external devices is an ideal way to work in this situation.
Here is a list of Logic plug-ins that cause latency:
Adaptive Limiter
Limiter
Multipressor
Noise Gate
Silver Gate
Linear Phase EQ
Match EQ
EnVerb
Space Designer
Denoiser
Pitch Correction
Pitch Shifter II
Vocal Transformer
Some third-party Audio Unit plug-ins may also cause latency—refer to the user guide for the respective products to see about this. All plug-ins that run from DSP cards such as the PowerCore or UAD card have inherent latency. Logic Pro compensates for the latency of any of these third party plug-ins, but as with Logic's own plug-ins, avoid assigning those that exhibit latency to busses, auxes or outputs until you are finished with tracking.
You may also find that simply bypassing plug-ins that run from DSP cards will not be sufficient to correct the latency they induce. If you are using these plug-ins, there are a couple of ways to deal with this. One would be to remove the plug-in entirely while you are recording tracks, and then re-insert them when you're done recording. Be sure to save your settings for each of these as a preset first, so that you can recover the appropriate settings when you re-insert them. Taking this idea a bit further, you could save the entire channel strip setting for each bus, aux or output that uses plug-ins that cause latency and then easily reassign everything for each channel in one go.
I hope you will get something useful from this good luck -
Plugin Delay Compensation, Stylus RMX & the UAD-1
Forgive me if this was answered previously, but I didn't really find anything when I searched the forum...
Anyways, Stylus is all in sync when it plays, but if I add a UAD-1 plugin to it it gets delayed and doesn't sync up anymore. I have the pda setting set to all and I tried 256 and 128 as my buffer. anyone have any ideas?Use the delay compensation plugin on the UAD. AFAIR UAD doesn't handle latency compensation in the traditional means. (Report to host from Plugin)
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Looking for help getting EZdrummer to work in Logic Pro 9
First off let me say I have looked everywhere and even talked to toontack about the following problem. Toontrack as said that the plug-in is working properly and the problem is with Logic.
Ok so I purchased toontrack ezdrummer and drumkit from ****. After installing them and updating to ezrumming 1.2 I am unable to load ezdrummer in logic pro 9. The really strange thing is that it will launch when using garageband and fully function.
In logic it is not in AU manager and not there for selection in the channel strip, so I thought at first that it was only installing as a VST. However, when looking in Library>Audio>Plug-Ins>Components, EZdrummer.component is there. I thought this was where Logic looked for AU plug-ins but I'm now thinking thats wrong but I've not been able to find anything saying different.
Please guys any help you can offer would be outstanding. I'm so fed up I'm about to get a refund but I would rather use this software.
Thanks.EZDrummer works fine here.
You could try trashing this file:
Macintosh HD/Users/YourUserName/Library/Caches/com.apple.audiounits.cache
On the next launch, Logic will rescan all your plugins.
You could also try a new OS X user account.
And what exact version of Logic are you using? -
Latency issue in Desktop Sharing
We are planning to develop a conferencing solution using LCCS. I am trying to evaluate the screen/desktop-sharing application. I am experiencing 5-10 seconds latency during the transfer of the screen data to the other end.
I am using the demo application(ScreenShareSubscriber and ScreenSharePublisher), provided in the SDK.
Some more details:
- Current OS is Windows 7 (32 bit).
- I am behind a proxy.
- I am running the applications in India.
- Using Flex builder 4.6 with Flash player 11.1.
- Using the developer account to test the application.
Questions:
- Can the delay be reduced programatically? If yes, then how?
- The final solution may be used by people distributed across the globe. Is there a possibility that, the latency is affected by your location?
- If the above is true, does Adobe provide cloud services (for commercial applications) that are distributed in different location, to reduce the latency?
- Can proxy server be an issue? We have port 443 open on the proxy server for TSL connections.
- If the above is true, then how can we avoid the issue? The final application may be used in a corporate network and we cannot ask everyone to change their network settings to connect LCCS services.
I have checked some posts on the forum, which say that, the performance is faster on Macs. We are currently not targetting the Mac platform.
Thanks,
SubratHi,
Please double check the following firewall port between two subnets.
Front End Servers-Lync Server Application Sharing service 5065 TCP used for incoming SIP listening requests for application sharing.
Front End Servers-Lync Server Application Sharing service 49152-65535 TCP Media port range used for application sharing.
Clients 1024-65535* TCP Application sharing.
If the issue persists, you can use the Lync server logging tool on FE server to test the process of desktop sharing.
Here is the link of using the Lync logging tool:
http://blog.schertz.name/2011/06/using-the-lync-logging-tool/
Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there. Please make
sure that you completely understand the risk before retrieving any suggestions from the above link.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
A quick primer on audio drivers, devices, and latency
This information has come from Durin, Adobe staffer:
Hi everyone,
A common question that comes up in these forums over and over has to do with recording latency, audio drivers, and device formats. I'm going to provide a brief overview of the different types of devices, how they interface with the computer and Audition, and steps to maximize performance and minimize the latency inherent in computer audio.
First, a few definitions:
Monitoring: listening to existing audio while simultaneously recording new audio.
Sample: The value of each individual bit of audio digitized by the audio device. Typically, the audio device measures the incoming signal 44,100 or 48,000 times every second.
Buffer Size: The "bucket" where samples are placed before being passed to the destination. An audio application will collect a buffers-worth of samples before feeding it to the audio device for playback. An audio device will collect a buffers-worth of samples before feeding it to the audio device when recording. Buffers are typically measured in Samples (command values being 64, 128, 512, 1024, 2048...) or milliseconds which is simply a calculation based on the device sample rate and buffer size.
Latency: The time span that occurs between providing an input signal into an audio device (through a microphone, keyboard, guitar input, etc) and when each buffers-worth of that signal is provided to the audio application. It also refers to the other direction, where the output audio signal is sent from the audio application to the audio device for playback. When recording while monitoring, the overall perceived latency can often be double the device buffer size.
ASIO, MME, CoreAudio: These are audio driver models, which simply specify the manner in which an audio application and audio device communicate. Apple Mac systems use CoreAudio almost exclusively which provides for low buffer sizes and the ability to mix and match different devices (called an Aggregate Device.) MME and ASIO are mostly Windows-exclusive driver models, and provide different methods of communicating between application and device. MME drivers allow the operating system itself to act as a go-between and are generally slower as they rely upon higher buffer sizes and have to pass through multiple processes on the computer before being sent to the audio device. ASIO drivers provide an audio application direct communication with the hardware, bypassing the operating system. This allows for much lower latency while being limited in an applications ability to access multiple devices simultaneously, or share a device channel with another application.
Dropouts: Missing audio data as a result of being unable to process an audio stream fast enough to keep up with the buffer size. Generally, dropouts occur when an audio application cannot process effects and mix tracks together quickly enough to fill the device buffer, or when the audio device is trying to send audio data to the application more quickly than it can handle it. (Remember when Lucy and Ethel were working at the chocolate factory and the machine sped up to the point where they were dropping chocolates all over the place? Pretend the chocolates were samples, Lucy and Ethel were the audio application, and the chocolate machine is the audio device/driver, and you'll have a pretty good visualization of how this works.)
Typically, latency is not a problem if you're simply playing back existing audio (you might experience a very slight delay between pressing PLAY and when audio is heard through your speakers) or recording to disk without monitoring existing audio tracks since precise timing is not crucial in these conditions. However, when trying to play along with a drum track, or sing a harmony to an existing track, or overdub narration to a video, latency becomes a factor since our ears are far more sensitive to timing issues than our other senses. If a bass guitar track is not precisely aligned with the drums, it quickly sounds sloppy. Therefore, we need to attempt to reduce latency as much as possible for these situations. If we simply set our Buffer Size parameter as low as it will go, we're likely to experience dropouts - especially if we have some tracks configured with audio effects which require additional processing and contribute their own latency to the chain. Dropouts are annoying but not destructive during playback, but if dropouts occur on the recording stream, it means you're losing data and your recording will never sound right - the data is simply lost. Obviously, this is not good.
Latency under 40ms is generally considered within the range of reasonable for recording. Some folks can hear even this and it affects their ability to play, but most people find this unnoticeable or tolerable. We can calculate our approximate desired buffer size with this formula:
(Sample per second / 1000) * Desired Latency
So, if we are recording at 44,100 Hz and we are aiming for 20ms latency: 44100 / 1000 * 20 = 882 samples. Most audio devices do not allow arbitrary buffer sizes but offer an array of choices, so we would select the closest option. The device I'm using right now offers 512 and 1024 samples as the closest available buffer sizes, so I would select 512 first and see how this performs. If my session has a lot of tracks and/or several effects, I might need to bump this up to 1024 if I experience dropouts.
Now that we hopefully have a pretty firm understanding of what constitutes latency and under what circumstances it is undesirable, let's take a look at how we can reduce it for our needs. You may find that you continue to experience dropouts at a buffer size of 1024 but that raising it to larger options introduces too much latency for your needs. So we need to determine what we can do to reduce our overhead in order to have quality playback and recording at this buffer size.
Effects: A common cause of playback latency is the use of effects. As your audio stream passes through an effect, it takes time for the computer to perform the calculations to modify that signal. Each effect in a chain introduces its own amount of latency before the chunk of audio even reaches the point where the audio application passes it to the audio device and starts to fill up the buffer. Audition and other DAWs attempt to address this through "latency compensation" routines which introduce a bit more latency when you first press play as they process several seconds of audio ahead of time before beginning to stream those chunks to the audio driver. In some cases, however, the effects may be so intensive that the CPU simply isn't processing the math fast enough. With Audition, you can "freeze" or pre-render these tracks by clicking the small lightning bolt button visible in the Effects Rack with that track selected. This performs a background render of that track, which automatically updates if you make any changes to the track or effect parameters, so that instead of calculating all those changes on-the-fly, it simply needs to stream back a plain old audio file which requires much fewer system resources. You may also choose to disable certain effects, or temporarily replace them with alternatives which may not sound exactly like what you want for your final mix, but which adequately simulate the desired effect for the purpose of recording. (You might replace the CPU-intensive Full Reverb effect with the lightweight Studio Reverb effect, for example. Full Reverb effect is mathematically far more accurate and realistic, but Studio Reverb can provide that quick "body" you might want when monitoring vocals, for example.) You can also just disable the effects for a track or clip while recording, and turn them on later.
Device and Driver Options: Different devices may have wildly different performance at the same buffer size and with the same session. Audio devices designed primarily for gaming are less likely to perform well at low buffer sizes as those designed for music production, for example. Even if the hardware performs the same, the driver mode may be a source of latency. ASIO is almost always faster than MME, though many device manufacturers do not supply an ASIO driver. The use of third-party, device-agnostic drivers, such as ASIO4ALL (www.asio4all.com) allow you to wrap an MME-only device inside a faux-ASIO shell. The audio application believes it's speaking to an ASIO driver, and ASIO4ALL has been streamlined to work more quickly with the MME device, or even to allow you to use different inputs and outputs on separate devices which ASIO would otherwise prevent.
We also now see more USB microphone devices which are input-only audio devices that generally use a generic Windows driver and, with a few exceptions, rarely offer native ASIO support. USB microphones generally require a higher buffer size as they are primarily designed for recording in cases where monitoring is unimportant. When attempting to record via a USB microphone and monitor via a separate audio device, you're more likely to run into issues where the two devices are not synchronized or drift apart after some time. (The ugly secret of many device manufacturers is that they rarely operate at EXACTLY the sample rate specified. The difference between 44,100 and 44,118 Hz is negligible when listening to audio, but when trying to precisely synchronize to a track recorded AT 44,100, the difference adds up over time and what sounded in sync for the first minute will be wildly off-beat several minutes later.) You are almost always going to have better sync and performance with a standard microphone connected to the same device you're using for playback, and for serious recording, this is the best practice. If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device. Attempt to match the buffer sizes and sample rates as closely as possible, and consider using a higher buffer size and correcting the latency post-recording. (One method of doing this is to have a click or clap at the beginning of your session and make sure this is recorded by your USB microphone. After you finish your recording, you can visually line up the click in the recorded track with the click in the original track by moving your clip backwards in the timeline. This is not the most efficient method, but this alignment is the reason you see the clapboards in behind-the-scenes filmmaking footage.)
Other Hardware: Other hardware in your computer plays a role in the ability to feed or store audio data quickly. CPUs are so fast, and with multiple cores, capable of spreading the load so often the bottleneck for good performance - especially at high sample rates - tends to be your hard drive or storage media. It is highly recommended that you configure your temporary files location, and session/recording location, to a physical drive that is NOT the same as you have your operating system installed. Audition and other DAWs have absolutely no control over what Windows or OS X may decide to do at any given time and if your antivirus software or system file indexer decides it's time to start churning away at your hard drive at the same time that you're recording your magnum opus, you raise the likelihood of losing some of that performance. (In fact, it's a good idea to disable all non-essential applications and internet connections while recording to reduce the likelihood of external interference.) If you're going to be recording multiple tracks at once, it's a good idea to purchase the fastest hard drive your budget allows. Most cheap drives spin around 5400 rpm, which is fine for general use cases but does not allow for the fast read, write, and seek operations the drive needs to do when recording and playing back from multiple files simultaneously. 7200 RPM drives perform much better, and even faster options are available. While fragmentation is less of a problem on OS X systems, you'll want to frequently defragment your drive on Windows frequently - this process realigns all the blocks of your files so they're grouped together. As you write and delete files, pieces of each tend to get placed in the first location that has room. This ends up creating lots of gaps or splitting files up all over the disk. The act of reading or writing to these spread out areas cause the operation to take significantly longer than it needs to and can contribute to glitches in playback or loss of data when recording.There is one point in the above that needed a little clarification, relating to USB mics:
_durin_ wrote:
If USB microphones are your only option, then I would recommend making certain you purchase a high-quality one and have an equally high-quality playback device.
If you are going to spend that much, then you'd be better off putting a little more money into an external device with a proper mic pre, and a little less money by not bothering with a USB mic at all, and just getting a 'normal' condensor mic. It's true to say that over the years, the USB mic class of recording device has caused more trouble than any other, regardless.
You should also be aware that if you find a USB mic offering ASIO support, then unless it's got a headphone socket on it as well then you aren't going to be able to monitor what you record if you use it in its native ASIO mode. This is because your computer can only cope with one ASIO device in the system - that's all the spec allows. What you can do with most ASIO hardware though is share multiple streams (if the device has multiple inputs and outputs) between different software.
Seriously, USB mics are more trouble than they're worth. -
Windows Server 2012 Hyper-V network latencies
Hi All,
I have an issue with our Windows Server 2012 Hyper-V hosts that I can't seem to figure out. Situation:
2 x Dell PowerEdge R815 servers with AMD opteron 6376 16 core CPU's and 128 GB RAM running Windows Server 2012 with Hyper-V.
2 virtual machines running on the same physical host and connected to the same virtual switch show high TCP connection latencies. One virtual machines runs a SQL Server 2012 database instance and a Dynamics AX 2012 R2 instance. The other machine a
SharePoint 2013 instance and the AX client. We see latencies of 20ms and higher on most of the TCP connections that are made from the sharepoint machine to the sql server machine.
At first I thought it might have something to do with the physical NIC's. It turned out that VMQ wasn't correctly supported by the firmware of the Broadcom BCM5709c cards. By default this setting is enabled. Turning off the VMQ setting somewhat improved
the situation but the latencies are still at 8ms and higher.
What I don't understand is what influence enabling/disabling VMQ should have on network performance. As I understand it now virtual machines connected to the same virtual switch bypass the physical altogether. Another point is that VMQ should actually improve
performance, not decrease it.
Another question I have is about the various tcp offloading settings on the physical NIC's. After installing the new firmware and drivers from Dell most of these settings are set to disabled. The documents I have been able to find talk about Windows Server
2008, any thought how these settings relate to Windows Server 2012 and whether they should be enabled?
Thanks in advance for your time and thoughts
Kind regards,
Dennes SchuitemaHi Denes,
Please try to update your BroadCom NIC driver version ,the newest version should be 7.8.51
For details please refer to following link :
http://www.broadcom.com/support/ethernet_nic/netxtremeii.php
Best Regards
Elton Ji
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