At Fixya.com, our trusted experts are meticulously vetted and possess extensive experience in their respective fields. Backed by a community of knowledgeable professionals, our platform ensures that the solutions provided are thoroughly researched and validated.
When I record with my m-audio mobile pre USB, there is no latency issues with recording but there is a delay in the headphones if I turn them up to hear myself while recording. How do I fix this? I'm
When I record with my m-audio mobile pre USB, there is no latency issues with recording but there is a delay in the headphones if I turn them up to hear myself while recording. How do I fix this? I'm using windows 8 and im recording with samplitude 11.5 producer version and for recording under the audio options i click the asio driver and It gives me the option to choose between magix low latency 2011 or m-audio mobile pre asio, and the only one that works is the m-audio mobile pre asio, which in the control panel I have the preferred buffer size set on 256 and I'm recording in 24 bit. I've tried increasing the buffer size but it slows down my computer and has no effect on the delay in the headphones. I've also checked and the m-audio drivers are up to date.
Well the mircophone isnt usb, but the digital interface is. I'm not using asio4all, i downloaded it before and tried to use it but it wouldnt work at all. As you can see in the picture those 2 are the only options i have. the m-audio mobile pre asio is the one i use because it came with the updated driver for my interface and works with my m-audio mobile pre interface, the other one came with my recording program because samplitude is made by magix but i tried it and it didnt work either. And i have already lowered my buffer settings way down to 256 because that was the only way that i could get rid of the delay while recording but i still have a delay in the headphones.Well the mircophone isnt usb, but the digital interface is. I'm not using asio4all, i downloaded it before and tried to use it but it wouldnt work at all. As you can see in the picture those 2 are the only options i have. the m-audio mobile pre asio is the one i use because it came with the updated driver for my interface and works with my m-audio mobile pre interface, the other one came with my recording program because samplitude is made by magix but i tried it and it didnt work either. And i have already lowered my buffer settings way down to 256 because that was the only way that i could get rid of the delay while recording but i still have a delay in the headphones.
specifically this part: Direct Monitoring When recording an audio signal into your music software, there is often a slight delay before it reaches the outputs of the software and MobilePre. This delay, called latency, is caused by the computer processing required to convert and record audio. Since this delay can be distracting, MobilePre provides a direct monitoring path from the inputs to the outputs, which is activated by pressing the Direct Monitor button. When the Direct Monitor button is pressed, the input signal is mixed with the output signal from your music software, and routed directly to the Line and Headphone Outputs. This allows you to hear the “live” inputs without latency. The Direct Monitor button has no effect on what is being recorded by your software. When using the Direct Monitor feature, make sure any software monitoring option for direct (or “low latency”) monitoring is disabled. This prevents the software from sending the recorded signal (post-disk) to the Line and Headphone outputs, which would cause “double-monitoring.”specifically this part: Direct Monitoring When recording an audio signal into your music software, there is often a slight delay before it reaches the outputs of the software and MobilePre. This delay, called latency, is caused by the computer processing required to convert and record audio. Since this delay can be distracting, MobilePre provides a direct monitoring path from the inputs to the outputs, which is activated by pressing the Direct Monitor button. When the Direct Monitor button is pressed, the input signal is mixed with the output signal from your music software, and routed directly to the Line and Headphone Outputs. This allows you to hear the “live” inputs without latency. The Direct Monitor button has no effect on what is being recorded by your software. When using the Direct Monitor feature, make sure any software monitoring option for direct (or “low latency”) monitoring is disabled. This prevents the software from sending the recorded signal (post-disk) to the Line and Headphone outputs, which would cause “double-monitoring.”
You can't post conmments that contain an email address.
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.
- If you need clarification, ask it in the comment box above.
- Better answers use proper spelling and grammar.
- Provide details, support with references or personal experience.
Tell us some more! Your answer needs to include more details to help people.You can't post answers that contain an email address.Please enter a valid email address.The email address entered is already associated to an account.Login to postPlease use English characters only.
Tip: The max point reward for answering a question is 15.
This is a common problem with recording multi-track sounds on a computer. It is caused by something that is known as 'latency' - basically it is down to the fact that the signal from your microphone takes a different route through the system to that taken by the audio playback.
The AT2020 is a USB microphone and the signal therefore has to pass through the USB bus circuit of your computer.
Despite claims of multi-tasking your computer cannot do everything at once so it naturally creates delays to make itself work properly. This will be more of a problem with a computer which has a 'cut-down' processor such as a Celeron (Intel) or Duron/Sempron (AMD). These processors are really only intended for basic computing (office type) use and are not really up to the high demands of video or audio recordings.
You may get some improvement by tweaking some of the audio settings for your sound-card to change the hardware acceleration.
This article may help you http://www.pcmus.com/latency.htm
If you are using a dedicated midi /music recording program such as Cubase then you may be able to correct the time lag using the 'quantise' function after recording your vocals.
However the best solution is likely to be a hardware one
Either buy a proper multi-track mixer/recorder designed for the purpose or get a second USB microphone - play your backing track on an external device - using one microphone to capture your voice and one to re-record the backing track (the only downside for this second option will be a gradual degrading of the sound quality each time it is re-cycled)
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.
This is a common problem with digital recording. Simply put, it takes some time to convert analog (audio) to digital and then more time to convert it back. So, your playback track goes through conversion and introduces some delay. You don't hear this because you have no zero reference. Your recorded sound goes through the A/D converter and also incurs some delay. When you play them back together, you hear the difference in time from one track to another. Most pro recording programs have some method to minimize or cancel the delay like slipping the tracks so they all start together. Pro Tools just introduced a version that automatically does this. They have always had what it called "low latency" monitoring which attempts to keep the "shortest" digital/analog path so overdubs are close enough that we don't hear any issues.
So, since you talk about the mixer but not the recording program, I can't tell you specifics but I'd look into the recording program for time alignment settings and options. If you have a manual for the recording program, try to look up latency.
This is normal for recording. Even professionals do the same. What you need is to set your input to mono(either hardware or software), or explore your software look for effects like STEREO to MONO with some delay which I prefer for my recording. I did have a good result. Good Luck
adjust your audio buffers, if your recording software allows it. however, there will always be some latency in usb recording devices. i recommend firewire or pci internal cards for recording.
Im having the same probz, got a Hp Pavilion Vista using mobile pre amp usb when i record using a condenser mic XLR nothing is picked up from the channels in Cubase or Nuendo. Im trying to record vocals but the Mic is not picked up and i cannot hear myself in the headphones and it wont transfere to cubase.
It has worked before when i had a desktop, but this only worked with a separate m-audio sound card inside the desktop and i plug in the output jax leads from the desktop (soundcard) to the m-audio usb pre amp
Is this why it is not working for the laptop do i need a Jax connection and where is this audacity
Hi micoach The condenser mic you are using requires phantom power for it to operate. Pay to check that that is turned on m-audio. Just to be sure.you say you can get sound off the m-audio headphone socket. Just need to check that. You will need to check that you have selected the device for recording from the preferences--audio I/O options menu within audacity. Perhaps check these things and get back to me here if you still have problems.I can go into more detail with you.
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.
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.
Well the mircophone isnt usb, but the digital interface is. I'm not using asio4all, i downloaded it before and tried to use it but it wouldnt work at all. As you can see in the picture those 2 are the only options i have. the m-audio mobile pre asio is the one i use because it came with the updated driver for my interface and works with my m-audio mobile pre interface, the other one came with my recording program because samplitude is made by magix but i tried it and it didnt work either. And i have already lowered my buffer settings way down to 256 because that was the only way that i could get rid of the delay while recording but i still have a delay in the headphones.
did you check your device's direct monitoring?
http://akmedia.digidesign.com/support/co...
specifically this part: Direct Monitoring When recording an audio signal into your music software, there is often a slight delay before it reaches the outputs of the software and MobilePre. This delay, called latency, is caused by the computer processing required to convert and record audio. Since this delay can be distracting, MobilePre provides a direct monitoring path from the inputs to the outputs, which is activated by pressing the Direct Monitor button. When the Direct Monitor button is pressed, the input signal is mixed with the output signal from your music software, and routed directly to the Line and Headphone Outputs. This allows you to hear the “live” inputs without latency. The Direct Monitor button has no effect on what is being recorded by your software. When using the Direct Monitor feature, make sure any software monitoring option for direct (or “low latency”) monitoring is disabled. This prevents the software from sending the recorded signal (post-disk) to the Line and Headphone outputs, which would cause “double-monitoring.”
×