Voxendo Latency Delay plug-in

Hi Guys,
Would anybody be able to help me install this plugin on my G5 please?
I'm running Logic 8 and after I clicked on the icon was asked what application to open the plug-in with. I tried Logic but it didn't work.
Any assistance greatly appreciated. Thanks in advance.
Cheers,

Is it an AU plugin? That is when it has the suffix .component. You should place it in the Library/Audio/PlugIns/Components folder first, then start Logic and it should appear in the Plug In list.
Never doubleclick a plugin, as it is not autonomous software, it only works inside a host (Logic, in this case).

Similar Messages

Mapping a Delay plug-in?

Hi guys,
I am trying to assign a button in mainstage to turn my delay plug-in on and off. But i can't get it to do that. I can get it to bypass the delay, but i want it to start off and then when i hit the button, turn on.
This is probably the stupidest question with the simplest answer, but i have just spend about 2 hours trying to figure it out...and i just can't.
Cheers!

I found out how to do it.
I couldn't figure out how to start off with it bypassed and then hit the button to un-bypass it.
The solution was to change the outcome of the button. So assign the button to bypass it in the edit screen. And then off to the right hand side there should be a panel of options. In here i set the button when off to make the delay plug-in OFF. And then when i click the button, it would turn the "un-bypass" on, thus making the delay work! Yay!
Thanks for the comment though!
I had to run it

Avoiding phase cancellation with sample delay plug in

Hi
I had soem horns in a mix sounded flat so i used the sample delay plug in to slightly delay the right side then they sounded great , really big and sat in the mix but when witching to mono they nearly dissapear.
Is there any trix to avoid this phase cancellation when swithching to mono?

Data Stream Studio wrote:
1.-I think if you reverse polarity on 1 of the channels of your mix instead of just reversing polarity of one side of the problematic horns, you may well solve one phase issue and create a whole lot of new ones.
2.-This is a good idea, maybe there's a setting that sound as fat as the one you're using but will not phase cancel in mono.
3.-I'm not sure what audio file you'd move... could you explain?
If it's a Stereo file, split it into two mono tracks, and move on of the two up/down the timeline by a few samples, until you get the least phase combing effect.
Maybe the OP needs to go back and re-mix that song, and find the culprit horn out of the whole section... if he can...

Delay plug-in usage

I am recording live through 2 mics. I need to introduce a delay on one mic of 25 ms (millisecond). Do I use the stereo delay plug-in or the sample delay? and how do I set it to 25 ms?
I am new to this and using Logic Express
Thanx in advance.

Either would work fine, I'd go for the sample delay. To work out how many samples per millisecond is pretty easy. If you're working at 44.1khz, the thats 44.1 samples per millisecond, 96 per ms for 96khz and so on.
1102.5 samples = 25ms at 44.1khz
Hope this helps.

Latency/Delay is KILLING ME.

So for some time I've been having some serious problems with latency/delay issues and it has finally left my computer basically worthless for recording music. There has always been some delay, but it was never too difficult to just line up the regions where they needed to be in relation to the metronome or other instruments, but Garageband has now taken to trying to speed up in the middle of recording. So sometimes the beginning of an audio clip will be at the right speed but it will speed up a little into it and cause it to change pitch, which is how I know it's not just a mistake by the musician. This renders the clip worthless. And it's beginning to do this every time. I need help. Any ideas? Thank you so much.

It is near to impossible to record into one session and paste pieces into a new one. Even a small discrepancy could mess off your timing.
It is better to record one large session with multiple tracks. Even on my G4 iBook I have done up to 20 tracks without difficulty. Some pointers:
1. If timing is exact through the song use a drum loop as a click track. You can use a metronome but a drum loop helps you 'get in the groove.' When all track are done you can delete or mute the click track.
2. Record dry (no effects) so your timing can be exact. Delays and reverb can throw off your timing as well as be a strain on your processor if They are activated for many tracks. Some effects are heavy processor hitters
3. Watch your playhead (the moving triangle) as you record. If it is always turning orange or red you are pushing your processor. It should remain white.
MacBooks and iBooks have special energy saver settings that drastically effect Garageband performance. Make sure your energy saver is not set to the 'best battery life' setting. You need the high performance setting to get the best from Garageband which is a processor hogging app.
Other ideas to maximize performance:
http://docs.info.apple.com/article.html?path=GarageBand/4.0/en/6567.html
Kurt

Latencies / delay in live streams

I'm doing some tests with FMS and live streaming, but I observe quite long delays.
When the stream is first accessed, the latency is around 3-5 seconds, which is acceptable.
But as we watch the stream for about 30 minutes or more, the latency grows to around 20 seconds, which is just too much.
What settings should I look at to keep the latency / delay minimal?

You're going to want to look at the bandwidth that you're pushing into your FMS system and the bandwidth you're getting out. Usually when you see evolving live latency it's because there's a bandwidth bottleneck somewhere and you're dropping frames and falling behind as FMS attempts to deliver a stream that's too big for the destination. If not then perhaps it's something else, but you'll want to eliminate that possibility first as it's the most common.
Asa

Latency Delay X

I have a SERIOUS problem. I run a recording studio, a small one. I have an artist who has recorded 2 out of 8 songs for an album that I have been paid to produce. I use Sonar 4 (cakewalk) for recording. I now have out of nowhere a delay in my sound that is about .5 to .0 second behind the actual sound input. This is not limited to the microphone, as this delay is also in my midi keyboard as well. I had an Audigy 5. sound card, after replacing my soundsblaster li've 5. card. This did not solve the problem. I re-installed windows and my applications nothing changed. I replaced my audigy card with a $200 X-fi sound card in hopes that it would improve .... nothing changed. I upgraded from gb to 2 gb ram (ddr) and still no luck. **side note** with my X-fi card now my mic plays out loud all the time. I had never before been able to hear the voice going into the mic, now i can, and i dont like it like that.**? I finally again re-installed windows and tested the mic with only the audio drivers and none of the software installed, not even the extra creative sofware... nothing improved. I then installed the creative software thinking perhaps it was needed, ... still bad. I AGAIN re-installed windows and installed EVERYTHING other than windows on another dri've not the primary OS dri've, and not a partition but a separate physical dri've. still no change. ASIO drivers have not helped at all!!! If there is something I havent done please let me know, because I have work to do, and bills to pay. So i need to get back to work, and this problem is giving me more days off than I desire. HELP!?Pentium 4 - 3.2 Ghz w/ Hyper Threading2-GB RAM (DDR)250 GB serial ata60 GB IDE20 GB?USB external320 GB firewire externalWindows XP Media CenterX-fi Fatality SoundcardGeForce 7300 GS PCIexp Videocard350W PowerSupply

Oh hey, since I'm here maybe you guys can help me. I also have a small home studio and I'm having some trouble with the sound quality. My specs are below
Pentium 4 3.4Ghz
2Gb DDR2 Ram
Sound Blaster X-Fi Platinum
Now, heres my studio setup. I have a mixer. I run my guitar, keyboard and mic through that and then into the X-Fi. But my problem is whenever I get finished recording with distortion the quality is really chunky, like its really heavy on the bass and loses some clarity. I have the mixer and sound card adjusted properly so I'm thinking its just how its being transferred. I was thinking of buying and Analog/Digital converter. have any ideas? I would really appreciate it.

Garageband, Edirol, Sound Latency / Delay Problem

Anyone have experience with Edirol UA-25 connected to a Macbook? Problem I'm having is with sound delay.

i'm having the same issues. on 2 separate machines. imac with OSX 10.6.8 and MBP with 10.8.3 it happens maybe 3 mins into playing or if i swap out amps. i usually have to reboot the whole machine to get it working.

Jabber 10.5.2 Chat Interface Latency Delay Poor Performance High CPU

Several users within my organization reported slow response when sending instant messages in Jabber. The interface response was very sluggish when opening chat windows, sending, typing messages etc. Response was almost as if the typing was buffered. Pressing enter after the message s could take 2 seconds before the message was delivered. We noticed high CPU when using the chat interface.
Restarting the Jabber client did not resolve the issue.
I closed Jabber and renamed the history files on the affected clients then re-opened. This restored the application / interface performance in every case. One of the chat history files was only 1MB so I am confused as to why a chat history db that small would adversely affect the interface so much.
=================================
4/23/15 Additional Information:
Installed Jabber 10.6.2 59142 on a client experiencing the issue in an attempt to resolve. The issue still persists until the history file is renamed/recreated evern after the upgrade.

We had the same issue here and did find a fix. It seems that the new IE11 cumulative patch KB3038314 was the root cause for us. We always QA Microsoft patches before release and all QA personal reported the issue. After removal of that patch the issue was resolved! A reboot was required but not needed to see the change in performance. Can one of you guys give it a shot and see if you have the same result. Right now I am just attributing it to our environment but it would be great to get other feedback from others.
wusa.exe /uninstall /kb:3038314 /quiet /norestart

Plug-Ins in a delay feedback-loop

I would like to set plug-Ins in a feedback loop from a delay plug-In
maybe in a simalar way like this:
from a audio track "send", i route the send signal in a delay. the delay output goes after in other plugins (filter, reverb, pitch shifter) and comes back in the delay input, but only maybe maybe 20%-30% of the output volume.
Does anybody know how to do this routing?

Admittedly, I had not tried the routing I had posted. I was just "thinking it through"... evidentially, not very well...
I knew I'd read multiple posts over at Sonikmatter about this sort of thing. Here's the first one I came across doing a quick search:
http://community.sonikmatter.com/forums/index.php?showtopic=7282
See if that gets you closer to what you had in mind...

Is there any way to decrease latency other than changing the I/O buffer

Hello,
I'm recording with two presonus firepods linked together. which allow for me to record up to 16 track at a time. I'm noticing some latency that makes it somewhat hard to record with. I do not want to lower the buffer for fear of intruducing distortion. is there any other way to lower latency levels in logic? If not would switching to a different interface help? if so, which one can you recamend?
thanks

After you've determined the smallest possible buffer setting for your system to run properly, the easiest way to achieve a minimal latency system from that point on is to have an external mixer for monitoring the audio you're recording. This way you're don't have to monitor audio thru your DAW software.
Apparently, software monitoring latencies are getting better (see the extensive thread/debate about the Symphony system on this forum -- search for "Symphony"). The only reason you'd need to use software monitoring is if you use plug-ins on the input channel(s) of what you're recording. Otherwise, software monitoring of audio thru your DAW does nothing but add latency (delay) to your monitored signal, as well as cause you to go prematurely gray in trying to deal with it.
I believe the Firepods are FW devices, and if so, there could be not only an increased amount of latency over, say, PCI-based systems, but the amount of latency might be variable every time you boot your system (there are extensive threads about this phenomenon on this forum -- do a search and you'll find them). If you're experiencing this, it is probably due, in short, to problems with the way the FW drivers are written.
I have a PCI-based system (MOTU, PCI-e 424) and putting arguments over whether MOTU stuff sounds the same, better, or worse than any other audio interface, this kind of system never experiences random changes in latency.
The other thing you must do is calibrate your recording delay. Again, with certain FW systems you might have to do this each time you boot your system. But I can tell you that with the card-based system I'm using, my latency and hence my recording delay settings never change, and I can be assured that what I record plays back in exactly the right spot every time.
BTW, the recording delay can be adjusted by using a simple looback test (again, there are numerous threads on this forum which explain how this is accomplished).

Latency Tests Report (based on iSchwartz posts)

Hi all,
With an hour to kill at home while my family gets here,I set to discover what some latency figures are on my system.Please look at my signature,the equipment list is current.
Specifically I wanted to see if the latency figures were consistent,in order to insert a record Delay best suited to each IO buffer size for my particular equipment.also,I wanted to explore the possible default latencies of some of my favorite Logic plugins,plus one Nomad Factory EQ.
This is what I found:
Latency Test:
Digi002Rack,
Logic Pro 7.2.3
CoreAudio Driver 7.3cs1
44.1kHz 24bit Logic session
Physical outputs 1+2 and back into channel inputs 1+2,
with 6" balanced TRS audio cables.
Plugin Delay Compensation set to =ALL
Sample Accurate Automation set to=volume,Pan,Sends only.
Pass Keyboard Events to Plugins set to=all.
Audio Merge Defaults: crossfade time 20mS curve 0
Core Audio: Enabled
System Memory requirements:114.0 MB
Driver:Digidesign HW( 002)
I/O buffer Size = (see chart below)
Recording Delay = (see chart below)
Max Number of Audio Tracks: 80
64 Busses option set to: OFF (unchecked)
Universal Track Mode: ON (checked)
Larger Disk buffer: ON
24bit Recording:ON
Software Monitoring: ON
Process Buffer Range:LARGE
ReWire Behavior:Playback Mode (Less CPU Load)
Maximum scrub Speed:Normal
Scrub response:Normal
WITH NO PLUGINS,and 0 as the Recording Delay:
IO buffer size = 32 274 samples
IO buffer size = 64 274 samples
IO buffer size = 128 350 samples
IO buffer size = 256 606 samples
IO buffer size = 512 1117 samples
IO buffer size = 1024 2140 samples
By setting equal but negative values for the sample delay,I compensated the throughput of the Digi002R.
WITH NO PLUGINS,
Recording Delay set negative to sample delay above:
IO buffer size = 32 0 samples
IO buffer size = 64 0 samples
IO buffer size = 128 0 samples
IO buffer size = 256 0 samples
IO buffer size = 512 0 samples
IO buffer size = 1024 0 samples
With 1 Channel EQ on click track:
IO buffer size = 32 0 samples
IO buffer size = 64 0 samples
IO buffer size = 128 0 samples
IO buffer size = 256 0 samples
IO buffer size = 512 0 samples
IO buffer size = 1024 0 samples
With 1 GuitarAmpPro on click track[default setting]:
IO buffer size = 32 14 samples
IO buffer size = 64 14 samples
IO buffer size = 128 14 samples
IO buffer size = 256 14 samples
IO buffer size = 512 14 samples
IO buffer size = 1024 14 samples
With 1 SpaceDesigner on click track[default setting]:
IO buffer size = 32 -128 samples
IO buffer size = 64 -128 samples
IO buffer size = 128 -128 samples
IO buffer size = 256 -128 samples
IO buffer size = 512 -128 samples
IO buffer size = 1024 -128 samples
With 1 LinearPhaseEQ on click track[default setting]:
IO buffer size = 32 0 samples
IO buffer size = 64 0 samples
IO buffer size = 128 0 samples
IO buffer size = 256 0 samples
IO buffer size = 512 0 samples
IO buffer size = 1024 0 samples
With 1 Nomad Factory PEQ5B EQ on click track[default setting]:
IO buffer size = 32 0 samples
IO buffer size = 64 0 samples
IO buffer size = 128 0 samples
IO buffer size = 256 0 samples
IO buffer size = 512 0 samples
IO buffer size = 1024 0 samples
I hope this is informative to you out there wondering all about latency,I/O buffer settings,et al.
Cheers

I posted this to the logic pro forum, but in case it helps anyone else here it is. Since it is slightly different from iS's method, whereby he drags the file around in the sample editor, I use the sample delay plug, I figured someone might benefit from this, and this thread has the right title for it.
Well I don't know about firewire latency issue specifically, but here is the hardware I tested and results:
Apogee Symphony
Apogee Rosetta 800
MacPro
Logic 7.2.3
OS X 10.4.9
Ran the test which went like this for anyone who wants to try it.
1- At your audio interface, take output 8 and connect it to input 8 (or which ever in out combo you want, create a loop back connection)
2- In logic, setup the input of Master Test Track to the output of your click track so you can record your click track to an audio signal. This will give you a nice transient to work with and a simple wave form for you to look at.
3- After you record Master Test Track setup the I/O out of the track to send to track 7-8 (if you used out 8 on your interface) and pan it hard left so it will send on out 8.
4- Create new audio track on track 2 (or available open track) and set its input to input 8.
5- Record the signal that you just sent from Master Test Track to track 2
6- Before you play back, insert in Master Test Track and add sample delay, which is in the logic>delay section of plugins. Be sure it's set to 0.
7- Insert on Track 2 the gain plugin and reverse the phase by clicking the reverse phase button.
8- Be sure that both tracks I/O out at this point go to your monitors. Both tracks dead center.
9- Start playback. I setup a cycle loop over a certain section of the recording so it will loop as I look for the sample diff.
10- Go to Master Test Track and open the sample delay plug and if you click and hold where the number is, you will see a little - show up. This will let you increase the number slowly.
11- As it's playing and you increase the number, thereby delaying the master track's playback, you will hear the sound change and start to fade. At one point it will either become inaudible or barely audible. if you go one extra sample after that, it will start getting louder. The number before it gets louder is your record latency. So if 83 is softer than 84 or 82 than this is your number.
12- To be absolutely sure this is right. Reset your test by deleting the recorded audio on Track 2. Bypass the Master Test Track's sample delay. Now go to the Audio and Hardware Drivers settings and set your Recording delay to the NEGATIVE value of your result UNLESS your sample delay is negative. So in my case it was 83, I set it to -83. Repeat the steps to record the audio through your interface. What should happen this time is that when you play back with the Master Test Track sample delay BYPASSED you should either not hear the click or it should be incredibly faint and sound exactly like your test. This means your test worked, your numbers are right, and your tracks will line up nicely from now on.
What my tests showed...
It seems that using the above setup I have a sample shift of 83 samples. That number is consistent through reboots and hardware shut down. I tried it 3 times and each time the right number was 83.
I don't know if this behavior will be different with firewire audio interfaces since I use an audio card. However, for the hardware I use, it is absolutely consistent.
I will keep my eye on this number for a while to make sure it doesn't shift.
If I misunderstood something about the results, fire off.
R

Why does recording delay change?

so yesterday i recorded vox and acoustic gtr totally fine. then i went to add bass today and i get a huge recording lag in placement of recorded audio. what changed this? everythings been stable for months. do i really have to go through all this recording delay calculation trouble just to have it work right again? seems totally ridiculous.
cory

Did anything at all change between yesterday and today? Did you add any latency inducing plug-ins (or are you using any currently)?
As well, updating your profile, as far as equipment goes, might help zero in on your problem.
jord

Still Latency after Freezing tracks... confused.

Up till now, I haven't really had an issue with Latency until I really started running a good amount of tracks. Right now I have 14 tracks with 1 or 2 plug-ins on each channel. Now I'm having some Latency when using the EXS24 from my controller keyboard (or any instrument for that matter)
When opening a fresh NEW project, there is hardly any noticeable Latency when using a stand alone Instrument.
I thought Freezing tracks is supposed to free up CPU, but before and after freezing has hardly any effect for Triggering and Audio input Latency. I'm a former Cubase 4 user and it's never been this bad. I have the same hardware setup. See my sig.
What can we do to optimize for Latency? Is the Apogee Symphony the only answer?

clank72 wrote:
Don't you hate it when it's always something really simple and stupid? After changing the Recording Delay to 0 and then fixing the Attack on my Logic Instrument most of the Latency went away. The Recording Delay setting must of been "Bumped" when I was adjusting another setting in the preferences.
Recording Delay Compensation is related to fine adjusting sample accuracy setting
with some device/system I need to set it as +/- 50 or similar value..
this is only for be sure about perfect syncronization when an incoming audio signial has been recorded in Logic.
Recording Delay must be change when the audio drivers are direct connected with the A(D converter...
the RME driver does not require adjustment...
0 set should be o.k.
We ALL have Latency. Wether it's a problem has to do with the person and how many live tracks simotaniously. I notice Latency at even 3ms. Some Vocalists will notice too.
Obvious we all have latency... but I mean... I don't have problem to play a faster part of piano solo with buffer setted as 256 on 48 Khz Sample rate (also 44.1 is good in my sustem
If anyone wants to hear their Latency, try plugging headphones into an outboard/external synth that is plugged into your audio device. Arm a track with the correct input from the synth, then turn up your speakers, put the headphones on only ONE ear and the other ear pointed towards your monitors. Load up a snare or sound with a fast attack on it. Hit the keys and you will notice Latency. It might have a chorus effect on it usually.
Usually I route the same MAIN output mirrored to headphones when recording a single instrument or a Vocal tracks.
I prefer to listen exactly the same audio program during recording phase..
I know this is not a "sound engeener" way...
But I prefer to listem everything during recording ... including FX, reverbs and Delay...
I always use SW set to ON!
I think everything is working good for now. It just bothers me when it's somthing simple and I feel like I wiasted someones time on the forum that has put aside time to help me out...err!
I'm glad to know that
you are welcome for any additional question
Thanks!
Thanks
G

Recorded track delayed on playback

Just starting out with Logic. I'm using an Aurora 16 with firewire and a Macbook Pro. Tried to record some bass yesterday, and it seemed fine as we were recording, but the recorded track is delayed on playback. What's up with that?

What were your Buffer settings during recording? I assume you didn't monitor through Logic (Software monitoring) but directly through your Interface otherwise you would have noticed the latency while playing. So set the Buffer settings in the Audio prefs to the lowest possible (try 32 or 64) and see if that improves the issue.
Did you record with your Plug In delay compensation to "All" or "tracks+Busses" or off? Try "tracks" or "off" next time and also make sure you don't have a latency inducing plug on your Stereo outs like Adaptive Limiter or alike.

Voxendo Latency Delay plug-in

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