Hello, I am new to digital music and I am unable to fix a crackling in my DAW. I would sincerely appreciate any help anyone could give diagnosing this problem. I'm using LMMS 1.0 and a Realtek built in soundcard on a 64 bit Windows 7 laptop. Here is what I have tried so far: 1.) First, I turned off system sounds, turned power usage to high performance, turned HQ mode in the DAW on and off. None of that seems to have any effect. 2.) I don't believe it is a single instrument or plug-in because it happens with the Vestige plug-in playing a "thunder" instrument, another Vestige running "rain", an AudioFile plugin playing an on board "explosion.ds" sample and another 2 Audiofiles with high and low pitch .wavs of bullfrogs. I have tried each of these tracks singly (with "solo" selected so the others are muted, although still being processed) and of course everything together. It has happened in all of these circumstances, although it was easier to provoke with Vestige than AudioFile. That could bebecause of the static-y nature of thunder and rain. Although it has happened in all of these circumstances, it does not happen every time. 3.) There is a difference in the distortion depending on whether I set the samplerate in Windows Sounds/Playback/Speakers/Properties/Advanced to 24 bit 48000 Hz or 24 bit 44100 Hz. When it is set to 48000 Hz the crackling is inconsistent, seeming to be better at first, but building up with more playbacks, the more frequent the playback requests the worse. It is never in the same place, but always in the same general areas. Occasionally, however it will happen immediately. Sometimes it is quiet and brief, sometimes loud and extended. However, when the samplerate is set to 44100 Hz the crackle is from the beginning of the song everytime, in quiet, brief, and regular intervals about 1/3 of a second apart (just counting in my head). At this samplerate increasing demand for playback does not shake it out of its pattern. 4.) I've seen alot of advice online for Asio4all as a method for giving the DAW exclusive and direct access to the soundcard. It does not work with LMMS, however, that should be moot in terms of bypassing windows kmixer because Windows apparently has its own built in way to do that - WASAPI. I have this selected in LMMS for my backend. It can operate in exclusive mode or shared mode - in exclusive it should give direct and exclusive soundcard access. The problem is that the only way to toggle between modes seems to be through the interface of your DAW and LMMS does not appear to offer any option for that. I assume since it is not providing a toggle that it would use the exclusive mode as default, but I can't confirm that yet since LMMS doesn't have great documentation. So I believe that LMMS has direct soundcard access but can not be 100% sure at this time. 5.) That brings us to buffer underruns. LMMS doesn't seem to have a way to count underruns. The entire clip has exported cleanly 6 times although the distortion continues to happen during playback, which would seem to implicate them, but I can raise the buffer size to 16,384 frames (LMMS max) and it has no effect. You would think that at that size latency would be intolerable, but underruns virtually eliminated. 6.) I have tried to check the signal size or for "hidden" signal by counting up my dBV's, but I may have fundamentally misunderstood this concept. Following the advice of several articles I started my levels out at -12dBV, some then went up to -8dBV. I set the master volume on the FXMixer to -6 dBV (started at 0 dBV, but -6 sounded better). There are 10 tracks. I had some tracks that I wanted to minimize the contribution of during the test, stuff I laid down that might be in the song latter, but aren't part of this section that I've been trying to correct the sound for. I dropped them to -33dBV. I added +6dBV for every effect such as volume automation, and even for a track having been sent to the mixer, as though that were an "effect" too since it is something the program has to keep track of. I should state that all mixing is part of an onboard function of LMMS, I have no external mixer. Adding all these together I got -183.6, which I believe means I have plenty of "headroom"? Whether or not that's the right term, I believe it still means that excessive signal is not the issue. Please feel free to inform any glaring ignorance I have displayed during this description. 7.) Finally, I don't believe the issue is simple lack of processor power. I have a quad core 1.9 GHz processor with 5.4 GB of usable RAM. I'm at the end of my knowledge and my wits here. I'd be so grateful for some help. Please not too technical in the response if you can. What's in this post represents the total of my digital music engineering knowledge, all of which I have acquired in the last 2weeks. I have never used any other DAW or musical hardware. Thanks!