Popping, static during playback. Scoured the forums for a solution to this common and old problem.

[putyo$onme]

Problem in title. I can't get clean playback, no matter the length of the track, number of layers, etc.
I have searched the forums and have tried what few suggestions there have been: I have fiddled with recording preferences, I run it in exclusive mode, disabled all anti-virus software on my PC. I can't think of any other information at the moment but will provide whatever is needed.

Rig: i5 6500, 16 GB RAM, SSD.

Thanks.

[Acoustica Greg]

Hi,

What is your playback device in Sound Device preferences? Are you using Mixcraft version 8.1 build 408? Are you running 32- or 64-bit Mixcraft?

Greg

[mick]

I would be tempted to do a HDD de-frag .

[putyo$onme]

Hi,

What is your playback device in Sound Device preferences? Are you using Mixcraft version 8.1 build 408? Are you running 32- or 64-bit Mixcraft?

Greg

I am using 64-bit, version 8.1 build 408. My playback device is: "Speakers (Realtek High Definition Audio)".

Thanks!

[putyo$onme]

I would be tempted to do a HDD de-frag .

I don't have an HDD, I have an SSD. When I check Defragment, the status of my drive (which had windows installed only 23 days ago) is "OK".
If I have to defrag this frequently, I might be done with Mixcraft, as none of the other DAWs I've used in the past week have had this problem (Reaper, Ableton, Fruity Loops). At any rate, I've optimized the drive and it's still having the same problem.

I've managed to be able to play one simple layer with no popping, but the second I add a second layer, the digital popping begins again.

I don't know if I mentioned earlier, but none of my volume levels are in the red, they barely touch yellow.

Thanks!

[Acoustica Greg]

Hi,

Are you using Core Audio (WaveRT)? You can try choosing the "Use Exclusive Mode" option.

Greg

[mick]

Hi putyo$onme, I didn't want to give the impression Mixcraft needs regular defrag, on the contrary. I've had issues in the past and it was a defrag that sorted it and nothing to do with Mixcraft at all. I have had little to no problems with playback with onboard soundcard and lately an audio interface. Files becoming scattered all over the hdd means windows has to look in multiple places to put a file together so suggesting a defrag is part of the elimination process but its clear you already have this in hand. For playback try a high buffer size 2 or 3 times what you have at the moment, low buffers of 20ms or less are only needed for midi input from a keyboard.

[mick]

Just found this on Google.

Which buffer size setting should I use in my DAW?
Article #1723804 Updated on Jun 15, 2017 at 8:57 PM
Getting error messages caused by too many plug-ins in your session? Are you hearing clicks and pops because of a low buffer? Does the buffer size even matter? Which buffer size should you use anyway? Setting an appropriate buffer size will improve your DAW’s consistency and reduce your frustration with error messages!

Low Buffer

When you are recording, you should set the buffer size to a lower amount to reduce the amount of latency for more accurate monitoring. The downside to lowering the buffer size is that it puts more pressure on your computer’s processors and forces them to work harder. With that in mind, you should try to use as few plug-ins as possible during the tracking phase so that your computer’s processing bandwidth is uninhibited. If you start to choke your processors with other tasks, you will experience clicks and pops or errors which will make tracking your project a nightmare. It is usually OK to give your singer a little reverb or to use light plug-ins, but you should avoid using processor-intensive plug-ins with a lower buffer size.
Tip: You can usually raise the buffer size up to 256 samples without being able to detect much latency in the signal.

High Buffer

Once the tracking phase is done, you no longer need to worry about latency because you are no longer monitoring live signals. In the mixing phase, you will be monitoring playback only, so it is safe to raise the buffer all the way up to the highest setting. This will allow you to use more plug-ins as well as heavier plug-ins before encountering clicks and pops or errors, depending on your computer’s resources and limitations. You may notice a slight delay when you start playback in your DAW with the buffer turned all the way up, but this is normal and is not a sign that your DAW is choking.

[putyo$onme]

Hi,

Are you using Core Audio (WaveRT)? You can try choosing the "Use Exclusive Mode" option.

Greg

I have the popping issue while using both WaveRT and Use Exclusive mode.

[putyo$onme]

Just found this on Google.

Which buffer size setting should I use in my DAW?
Article #1723804 Updated on Jun 15, 2017 at 8:57 PM
Getting error messages caused by too many plug-ins in your session? Are you hearing clicks and pops because of a low buffer? Does the buffer size even matter? Which buffer size should you use anyway? Setting an appropriate buffer size will improve your DAW’s consistency and reduce your frustration with error messages!

Low Buffer

When you are recording, you should set the buffer size to a lower amount to reduce the amount of latency for more accurate monitoring. The downside to lowering the buffer size is that it puts more pressure on your computer’s processors and forces them to work harder. With that in mind, you should try to use as few plug-ins as possible during the tracking phase so that your computer’s processing bandwidth is uninhibited. If you start to choke your processors with other tasks, you will experience clicks and pops or errors which will make tracking your project a nightmare. It is usually OK to give your singer a little reverb or to use light plug-ins, but you should avoid using processor-intensive plug-ins with a lower buffer size.
Tip: You can usually raise the buffer size up to 256 samples without being able to detect much latency in the signal.

High Buffer

Once the tracking phase is done, you no longer need to worry about latency because you are no longer monitoring live signals. In the mixing phase, you will be monitoring playback only, so it is safe to raise the buffer all the way up to the highest setting. This will allow you to use more plug-ins as well as heavier plug-ins before encountering clicks and pops or errors, depending on your computer’s resources and limitations. You may notice a slight delay when you start playback in your DAW with the buffer turned all the way up, but this is normal and is not a sign that your DAW is choking.

This solved it for me, though it only really deepens my dissatisfaction with Mixcraft. I am having problems only during playback on WaveRT (what I use to record, especially audio tracks, but not necessarily midi tracks). When I change to ASIO which has a buffer size of 512 right now, I do not experience any popping at all. However, latency is ~11ms. Completely unusuable for recording.

So the problem now is that I have to switch back and forth between the two drivers constantly during the recording process. Am I just being a spoiled baby or does this seem unreasonably inefficient and time consuming?

Warning: Incoming Rant:
First, let me say the best thing about Mixcraft so far is the community on the forums and I really appreciate the timely responses by everyone. I've made several posts on here and the replies are always swift, kind, and typically well-researched.

I feel a lot of buyer's remorse with Mixcraft and I'm trying not to give up, but damn man, a trial of Ableton does exactly what I want without having to constantly fiddle with drivers and buffer sizes in the middle of recording a very basic track. I just don't understand how anyone can happily use this DAW in such a haphazard, clunky fashion. Do working producers really switch audio drivers multiple times just to make a simple, unmastered, plug-inless scratch track?

Here's where I humbly ask your honest opinion: Is this just a matter of me buying the wrong DAW for my needs or do I need to adjust my expectations as a music maker? I mean, Ableton is $450 and Mixcraft was only $90, so I understand certain disparities in features, but my issue is with very basic usability. Does anybody have any advice or suggestions?

[mick]

I'm out of (basic) suggestions but there are guys on this forum that are considerably more knowledgeable than I am. All I can add is that even when I used the onboard sound card I got no popping etc even with 20 or more tracks playing and using 11.5 ms latency buffer, all the demo tunes play perfectly no matter what sound device I use. Oddly, there has been a subject where a better video card improves audio performance but I can't remember too much about the details. I'm now using a M-Audio card with latency down to 2ms and still get no popping, my last throw of the dice is you could try the free asio4all audio driver and see if it helps.

[Mark Bliss]

I have a laptop with a Realtek HD card that I use often for portable studio uses.
My answers to your questions may not be what you want to hear, use the info as you wish. This has been my experience:

First off-

"Do working producers really switch audio drivers multiple times just to make a simple, unmastered, plug-in less scratch track?"

I would suspect the answer here is largely no.
One of the first things I learned when I bought a DAW program (apx. 10 yrs ago) was that to get reliable results for recording, one needed to ditch the on-board sound card and run a dedicated audio interface and its proprietary audio driver for acceptable results.
Some reasons- the mic preamps tend to not be very good, the audio drivers aren't really designed for that purpose, etc.

And for what its worth, I tried ASIO4all at one point and it caused far more problems than it solved.

That said, I do use my Realtek HD card in my current laptop quite frequently for playback and mixing existing recordings, mostly for simplicity and mobility.
Things I have found-
The Realtek ASIO driver is just about useless. It doesn't work right. As far as the OP's statement about 11 ms latency, if it were really 11 ms, it shouldn't be unusable. My experience has been that the latency reporting must not be correct at times. But even that aside, it just doesn't work well.
I generally ditched trying to record via the Realtek, and primarily use CoreAudio mode fpr playback only. (So exclusive mode isn't even necessary since I am not trying to achieve low latency.)

One thing that hasn't been brought up-
If you are using a laptop as I often am, I have discovered that by default, the computers performance is hampered by power saving mode settings. As delivered it never went into performance mode and would therefore result in the described skipping and stuttering when attempting to play DAW projects. Changing those settings so that it is in performance mode at all times (when power cord is plugged in) fixed that, and I can now run upwards of 40 tracks and generous numbers of plug ins without much (if any) issue.

Back to recording-
One thing I sometimes do with limited success (again, for portability) - is to use a USB mic with its own monitoring output. Thus used, it effectively becomes your soundcard for recording. Its not a great solution, but works for me in a pinch.

Ultimately however, for best results, you will likely find you need the proper tool for the job. A purpose built interface and its proprietary ASIO driver for recording.

You never mentioned what you are using for a mic BTW.

[JonInc]

Mark is spot on.

Your issues are due to bad sound card drivers. I'm surprised that the Realtek card offers you ASIO drivers... Or are you using the ASIO4all generic driver?

Most of us use Mixcraft without the problems you mentioned -- but we have also invested in a proper audio interface designed for multitrack recording.

I promise you that if you go to the forums of the other DAWs you mention that work "just fine", you'll see that the vast majority of users there also have proper audio interfaces (not built-in sound cards). That's because built-in sound cards are very cheap in quality and are not designed to stream audio efficiently.

Can they work? Sure, sometimes. But you'll never get the best performance or lowest latency you'll get from a proper interface. Actually, you got lucky. Realtek (and similar) are probably the number one cause of issues with any DAW, not just Mixcraft.

If ASIO4all driver doesn't help you, then your only other option is to buy a decent audio interface. They start around $100 USD.

[chibear]

A couple of more things to look at:

1. Do the clicks and pops still exist in the mixdown when you render the project to wave?

2. Have you tried either freezing or mixing all tracks but the one you are working on to audio (especially MIDI)?

3. How many tracks are you talking about in the project? Are they audio or MIDI? What fx is used on each track.

You are at a definite disadvantage working from a single drive (even a SSD) that must concurrently service Windows, Mixcraft, plus any streaming from any VSTi libraries and audio streaming.

Also in Kontakt and a few others there is an intolarance to streaming too much data over 1 or 2 MIDI channels, so if that is your case, I would advise spreading your MIDI tracks over the available 16 channels.

Hope some of it might be some help.

[Ianpb]

I have an SSD.

Solid state drives should not be defragmented. It's not even necessary to do so.

https://helpdeskgeek.com/featured-posts ... ag-an-ssd/