Latency and reverb

Have a read at this article. I assume you are using ASIO4ALL if not then you should.

http://www.image-line.com/support/FLHelp/html/recording_audio_USBmic.htm

Specifically this part:

Note about latency: USB microphones generally only allow you to hear the sound being recorded after it has passed through FL Studio. As the latency of USB microphones depends on the ASIO4ALL latency settings, there may be a troublesome delay between the sound made and it being reproduced from your speakers or headphones. For example, low latency monitoring is particularly important to vocalists as high latencies cause a distracting echo effect. The only solution to this problem is to lower your buffer length settings (there are limits) or obtain a new soundcard with direct-monitoring.

http://www.mmo-champion.com/threads/677453-windows-7-Vista-latency-fix
FIND THIS .MAY WORK.

You'll never get rid of the latency problem if you record from a speaker. Speakers are analog and when you mix digital with analog you get latency.

Try increasing the latency. Depending on your computer, it may not be able to handle low latency levels. Latency is a measure of the delay between the time you make a sound and the time the computer records that sound. With the Mbox mini, the unit for latency is the number of samples the driver collects before sending them to an application as input. Lower latency can make recording easier and more accurate, but not all computer systems can operate at the lowest latencies without introducing noise. Open the Mbox mini dialog and select the latency tab. experiment with higher latency levels and see if it goes away. This is what the dialogue might look like. another solution would be to try a different DAW. I had the same problem with my Fast Track Pro in ProToolsand it worked fin on Reaper, Garageband and Logic.

Sounds like what is called a latency problem. You need to go into the settings in Sonar and under audio, adjust the Latency to a higher number...I don't know sonar home studio, or I would tell you the exact area to go in the menu's to adjust latency..Hope this helps!

To see if it's your hardware, up your latency settings to "highest" via the tascam control panel. Also, be sure you are familiar with the various ways to up latency in your recording package: in cubase it's under device settings, vst bay. If this works, you know it's some setting somewhere. Try disabling your internet connections and setting pc performance to "background services" Also, read a pc optimization guide for daws - they are everywhere. Tascam will email one to you if you ask.

Lastly - try the asio4all driver. This fixed my pops, but when I understood all the various latency settings and set the tascam to highest latency, I could use the tascam driver. Highest latency on my p4 3.0 2gb ram pc means 12ms input and my output is 61ms.

If the asio4all driver works - it's a likely latency issue.

Sounds like you have a latency problem. This can be for a variety of reasons, something Cakewalk admits. The reason for that could be everything from processor limitations, not enough RAM, your soundcard, or simply that you need to change the latency configuration in the audio settings menu of the program. Considering that that is the cheapest fix, try changing settings first. Be sure to follor the help information in Cakewalk - they can be a little confusing for Cakewalk (I have no idea why they can't seem to explain their programs after all these years) but you will be able to figure out what settings are right for you.

My guess is that when you record, you are listening to the previously recorded track, and playing to it. Then when you play it all back as a mix, they are not in sync as when you were recording them.

If that is not the problem, plesae explain further. You are able to monitor your previously recorded tracks, right? Not just playing each track "blind" and trying to have them match? Hope not. That is rarely possible - if at all.

If you are monitoring your prerecorded tracks while recording on top of them and it sounds different when you play it back, it's definitely latency.

Mark

It's crucial that you understand the concept of latency.

Latency it the time delay between the trigger and the operation.
In terms of computer sound recording and generation:
1) The time between the midi trigger and the sound generated by a software synth to come out of the speakers.
2) The time between the analog audio signal input to be converted to digital and stored on the hard drive.
3) The time between the digital audio track to be retrieved from the hard drive and converted to an analog signal that comes out the speakers or headphones.
4) All of the above at the same time.

The factors that contribute to this latency form a pipeline that include the A/D (analog to digital) device (UA-25), the software drivers for that device, the operating system, the recording application being used, the operating system again, your internal computer speed and your hard drive speed. Remember that the sound needs to go both in and out, often simultaneously (thats called full duplex).

A latency of up to 50ms is probably tolerable, but for optimum recording, it should be closer to 5ms. If you were to go over 50ms, you would need some sort of time compensation, whether automatic or manual, because nothing would synchronize properly, be it the metronome or previously recorded tracks.

"What's this mean, and how does it relate to my situation?" you ask.

Lemme tell you. The recording program and drivers will incorporate software buffers to adjust for slower components in your sound recording pipeline. Without buffers, or with too few buffers, the pipeline can drop information (sound) if it is not be able to process it quickly enough. This will often result in stuttering and skips. Sound familiar?

So why not just put LOADs and LOADs of buffers in the pipeline? You most certainly can, but for every buffer you add, you gain a bit of latency. And remember that too much latency is a bad thing.

"Ouch! I still don't get it. What am I supposed to do?" you ask.

I reply, "Good question, Grasshopper." You want to stop your stuttering. You need to increase the efficiency of your sound pipeline. For regular people this means the following, in order:

1. Make sure you purchase an A/D-D/A device that has low-latency drivers utilizing one or more of the following standards: ASIO, WDM, MME or CoreAudio. Selecting the one that your RECORDING APPLICATION utilizes. ASIO is widely supported and good. WDM is the worst because it is really just regular Windows drivers, but they can sometimes be optimized for low latency.
2. Make sure you install your drivers properly, selecting the one that your RECORDING PROGRAM uses.
3. Make sure that your recording application has the hardware selected for your A/D-D/A device and the drivers that you previously installed.
4. Make sure that your recording application is using ONLY the hardware specifically designed for low latency, which is your A/D device that you previously purchased and installed.
5. Make sure you include as many buffers as you can while maintaining a low latency. Often, this is dependent on your recording application.
6. Set the UA-25 to 44.1KHz or 48KHz, not 96KHz. 96Khz does not allow for full duplex.
   

Do these things and attempt to record something. If it stutters right away or you get "underruns" you need to increase your buffers. If everything is peachy for a while, but soon you start getting skips, try increasing the buffers, or get a faster hard drive or turn off some digital effects.

In summation: A Celeron 1.6GHz should have sufficient power to record, play and add a few VST plug-ins, maybe even a dozen or more. Chances are that your hard drive is fast enough to retrieve and record 16-32 tracks at a time. Turn off all the stupid, little programs that make icons near your clock, like the DELL support and Weatherbug and AIM and AOL and HP Monitor and anything that takes processing power and hogs memory.

Hi emmastevens

what you are hearing is latency(delay) in the electronics, software, and maybe your computer is producing some problems also. Have you tried to monitor yourself direct from a mixer before you feed the signal into the MBox.

regards
Graeme

The main cause of latency is your hardware. What kind of soundcard do you have? If you are using the one that came with your computer then that is your problem, no matter how good or new your PC is, manufacturers like to save costs by using cheap motherboard mounted soundcards, I had exactly the same problem as you when I started using a MIDI keyboard with cubase.

I'd suggest buying a soundcard designed for audio applications, I use the M-Audio Fasttrack which you can pick up very cheaply nowadays. I have my latency down to around 1/20 second and as an added bonus I get a 1/4 input and an XLR input as well for recording.