Crackling and popping audio!

mcowing

New member
You read all the reviews, but, I never thought it'd happen to me. I just got a new interface (M-audio m-track 2x2M). I disabled the onboard mobo audio. I get fairly clear audio but I'm sure I hear some pops every now and then. Also, I can play itunes at the same time as playing audio in Ableton Live, but it pops and crackles and distorts like crazy, on monitors or headphones. This thing is supposed to be new and super low latency. Any suggestions?

core i7 4ghz
SSD
16gb RAM
gtx 760
m-audio m-track 2x2m
16bit 44.1khz
192-256+ms is what I tried.
m-audio latest drivers
Mackie CR4BT
Ableton Live
 
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Since another similar thread is fresh in my mind, you might try popping out the GPU and running off of on-board graphics and try again. A few of us have had issues with graphics cards causing some problems with audio. It'd be worth a try just to see.
 
Your CPU is pretty good so it's unlikely that it can't keep up with a buffer size even lower than that.
I'd check the CPU load in the task manager, anyway.

Try and use the High Performance power plan when you need low latency.
C-States and power saving settings are also bad for low latency.

I'd also try and use LatencyMon (it's free). It will check if your system is suitable for low latency applications and if there's an issue with a specific driver.

When recording or if you need low latency, disable any network interface controller or wi-fi adapter.
Win10 usually keeps every driver updated, but I'd check that anyway.
 
This thing is supposed to be new and super low latency.

16bit 44.1khz

If i may ask in between?
Why so low quality set? And i see that more than once on this forum. My project i usually work in 24/36bit (=standard) and 96khz. Never ever 16/44.1.
 
I am NO PC guru but I think there is some confusing information here?

AFAIK the INTERNAL audio 'engine' in a DAW works at 32bit (or higher) float and thus make an internal overload nigh on impossible. An interface however works its inputs and outputs at either 16 or 24 bits and at whatever sample rate takes your fancy.

The OP might HAVE to run at 16bits because of the Youtube content? I do know however I CAN record Radio 3 internally at 24 bits 44'1kHz (Samplitude) although of course the source sound is nowhere near that standard (so I don't!) .

I no longer have a W10 PC but there MIGHT be a funny there? My first recourse would be to try a Win 7 machine*. Also, look at Services and kill any Wireless adaptors.

*You don't need an i7 CPU and 16G of ram to run that wee AI, or anything near it!

Dave.
 
Running at 44.1/16 and a buffer size of 256, there should be no reason for a machine with those specs to struggle. It should be able to handle a buffer of half that size without breaking a sweat.

At this point, make sure that your BIOS and chipset drivers are up to date, and seek out any firmware updates that may have been released for your various devices, including the audio interface. And I'd again suggest trying to record/play audio after removing the GPU and running off of on-board graphics, just to see if the GPU is causing the troubles.
 
Have you made sure that your using the most upto-date drivers for that M-Audio card?
If you are - maybe uninstall and then reinstall that driver.
 
The OP might HAVE to run at 16bits because of the Youtube content?

Isn't it possible then to load 44.1/16bit youtube content in a 24bit-32float(indeed!)/96khz project? I've done that many times.

Running at 44.1/16 and a buffer size of 256, there should be no reason for a machine with those specs to struggle.

That's a bit my point to. 44.1/16 is quite low if 24-32/96 is possible too, so must be possible easy. If not something else must be wrong. Like the already mentioned driver or whatever. Or perhaps a bad connection somewhere in the system?

And my question isn't anwered. Why do i read more than once here that many of you work in 44.1/16? I only export to mp3 end result in that, and mostly even higher. Or is it some kind of misinformation that's given?
 
And my question isn't anwered. Why do i read more than once here that many of you work in 44.1/16? I only export to mp3 end result in that, and mostly even higher. Or is it some kind of misinformation that's given?

If people set their interfaces to 44.1/16, there will be a variety of reasons: it's a default setting that hasn't been changed; there has been no perceived need to change it; it does a pretty good job, and any quality handicaps that setting may have are usually trivial compared to the quality impacts of other parts of the system.

I don't know how many HR members record at 44.1/16. Maybe a poll might be informative. In the late nineties, I did because there was no other option on my interface.

I now record 44.1/24, or 48/24 if I'm recording for video.
 
I record at 48k/24 for the most part, and that's just because I record videos for YouTube. I'd just record at 44.1/24 otherwise. I used to record at 96k/24, but it didn't buy me anything but larger files.
 
I've had the same problem in the past and I defragmented and cleaned up the drive and it stopped the noise.
 
Thanks for the advice guys. I haven't had a ton of time but I did spend some time tweaking things and I got the noise down. I reinstalled all my usb drivers. For those that don't know, you go into device manager (windows) and find your list of USB devices. Right click on each generic driver and choose uninstall. When you restart your computer windows will automatically reinstall what is needed. It's a good technique if you're having any USB related issues on your computer. I also lowered the output volume from within windows (easily located on your Start Bar, on the right hand corner) There are many volume adjustments available to me lol, 1. within windows 2. the m-audio knob 3. the knob on my monitors (Mackie CR4)!!

I haven't played around with the bit/sample yet. The firmware is up to date as is the driver for the M-audio M-track. I haven't checked my BIOS but updated it several months ago. Same with chipsets. Will check soon.
 
Thanks for the advice guys. I haven't had a ton of time but I did spend some time tweaking things and I got the noise down. I reinstalled all my usb drivers. For those that don't know, you go into device manager (windows) and find your list of USB devices. Right click on each generic driver and choose uninstall. When you restart your computer windows will automatically reinstall what is needed. It's a good technique if you're having any USB related issues on your computer. I also lowered the output volume from within windows (easily located on your Start Bar, on the right hand corner) There are many volume adjustments available to me lol, 1. within windows 2. the m-audio knob 3. the knob on my monitors (Mackie CR4)!!

I haven't played around with the bit/sample yet. The firmware is up to date as is the driver for the M-audio M-track. I haven't checked my BIOS but updated it several months ago. Same with chipsets. Will check soon.

Windows will not necessarily update with correct drivers man. Mobo, bios, graphics card, interface, etc. drivers should be downloaded directly from the manufacturer.

Windows will usually grab things like basic printer and flash drive drivers, but not always. Best to make sure you have the correct proprietary ones.
 
Windows will not necessarily update with correct drivers man. Mobo, bios, graphics card, interface, etc. drivers should be downloaded directly from the manufacturer.

Windows will usually grab things like basic printer and flash drive drivers, but not always. Best to make sure you have the correct proprietary ones.

Of course. You'll notice I said only the generic USB drivers. Specific devices should always have the correct driver from the manufacturers.
 
This topic is very confused - the initial question was with pops and clicks, so what we should have done is asked to listen - this gives the steer towards electronics and driver issues, and away from preamp, mic and input problems.

Then we came to all the stuff on sample rates and bit depths. Clearly we have a number of members interested in old analogue formats, so it's a bit laughable that we insist that we should use the highest settings on every project. Much of my work is with an archive of recordings from around 94, with a few that were originally analogue in the 80s. If I open an old project then they're going to be 16 bit 44.1KHz and frankly it really doesn't matter. Many of my projects feature old sound sources too, and there is no need to mess around with sample rate conversion when changing a sax part, or adding some vocals - especially when the end result is going to be on a CD. What's the point converting things then back again, because that's not good for purity.

Any new projects will be 32/96 which gives me the best balance between processing power, files sizes and system capacity. I've got a collection of old VST instruments too - I have yet to hear the difference between 48K Vs 96K, let alone 192K. What exactly is the point of recording information that is in most cases, absent - Nyquist was a clever bloke, and loads of my synths and other source sounds are frankly a but harsh at HF and eq usually tames that nicely - so why would I wish to record it?

I'm perfectly happy that individuals can record in whatever format they like - but with more and more distribution actually on mp3s, the actual point eludes me. Quarter century old DAT recordings at 48K still sound good in my collection. Adding an extra two octaves of emptiness doesn't convince me that advice to record at 192 makes any sense whatsoever. Sorry. I think personally, it's PT Barnum all over again.
 
This topic is very confused - the initial question was with pops and clicks, so what we should have done is asked to listen - this gives the steer towards electronics and driver issues, and away from preamp, mic and input problems.

That tends to be the nature of forum posts . . . they can meander along different paths like a grand old river.

Clearly we have a number of members interested in old analogue formats, so it's a bit laughable that we insist that we should use the highest settings on every project.

It's one of the paradoxes that exist in recording, i.e. the wish to record lo-fi at the highest quality.

. I think personally, it's PT Barnum all over again.

Humans are attracted by bright shiney things, and marketers know how to make the most of this.
 
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