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Please Advise on Persistent Crackling Sound Condenser Microphones
Old 31st July 2014
  #1
Here for the gear
 

Please Advise on Persistent Crackling Sound

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 on board 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!
Old 31st July 2014
  #2
Gear Addict
 
Yumid's Avatar
 

Im not reading that whole thing..just turn your Buffer size up.
Old 31st July 2014
  #3
Here for the gear
 

@Yumid- I'm sorry reading is tiring to you. Thank you for taking the time to reply, although I wonder why you bothered to trouble yourself to tell me you weren't going to trouble yourself. As I explained in the post you didn't read I tried that already. My post is long because I did a lot of work researching and diagnosing my own problem before bothering the forum, as posters are asked to do. Perhaps respondents should be asked in return to read what they respond to.
Old 31st July 2014
  #4
Gear Guru
 
Muser's Avatar
these kinds of things are often due to the soundcard drivers.
if you can re-install those I'd try it. sometimes even more than once.

maybe also check for updates and what those have been found to remedy.
e.g. posts mentioning re installs and fixed crackles pops etc.
Old 31st July 2014
  #5
Lives for gear
use an external soundcard.
Old 31st July 2014
  #6
Whenever I have crackling issues, its usually a direct result of my buffer being too low for the CPU usage.
Old 31st July 2014
  #7
Gear Addict
 

Quote:
Originally Posted by inversound View Post
use an external soundcard.
I totally agree with inversound, i would personally never try and use a factory built in laptop(or desktop) soundcard for a DAW, these soundcards are made for listening to mp3´s and such.

Most importantly i think your main problem is likely to be computer drivers that behave badly which take over priority and cause drop outs, this is a very common problem especially with laptops where the Wifi often is one of the main culprits.

I would suggest that you first test with the excellent freeware program DPC latency checker, here is a link:DPC Latency Checker

Its not strange at all that exporting works since its just calculations, low latency audio on the other hand is real-time and if the computer has a higher priority on other things than audio you might experience a dropout/click and such. You often have to do some tuning of the computer to get it to work.
Old 31st July 2014
  #8
Here for the gear
 

Hi everyone,
Thank you for your responses!

@Neonknight My laptop is used with a wired connection, not WIFi. Thank you for the advise on DPC latency checker. What exactly would I be testing for? Please correct me if I'm wrong, but I thought latency was only an issue if you were connecting external instruments, or performing in real time. I'm working completely in the box, and only currently worried about recording.
On another note, how do I stop my computer from putting a "higher priority on other things than audio"? You also say "i think your main problem is likely to be computer drivers that behave badly". What does that mean specifically and how do I get them to behave well?

@joe_04_04 I didn't think it was possible for it to be a buffer problem with my buffer size cranked up to 16,384 frames? Is it somehow still possible?

@Muser Thank you for suggesting re-installing the drivers to the soundcard. I haven't tried that yet.

@inversound @Neonknight What kind of external soundcard do you recommend? Does it connect via USB or some other way?

By the way - what does it mean that the crackling is distinctly different when I change the samplerate in Speaker Properties from 48000 Hz to 44100 Hz? #3 in my original post for more info. I thought that was weird and probably meant something specific.

Thank you all for your time and help.
Old 31st July 2014
  #9
Gear Addict
 

Quote:
Originally Posted by Ankhet View Post
Hi everyone,
Thank you for your responses!

@Neonknight My laptop is used with a wired connection, not WIFi. Thank you for the advise on DPC latency checker. What exactly would I be testing for? .
Even wired LAN can be as problematic as WIFI and there are also other hardware peices and their drivers that can be problematic and programs running as well.

I would like you to first test with the DPC latency checker so that we can see if it is indeed problems with driver priority etc. When you have testet we can go from there.

When you run the program, run it for several minutes, it should look like in this pic below if its ok= all green bars.



If you have red bars like in the pic below you have problems and its never gonna work with low latency real time audio. As you can see the DPC checker give you some hints on what could be causing the problems.



Quote:
Originally Posted by Ankhet View Post
@Neonknight Please correct me if I'm wrong, but I thought latency was only an issue if you were connecting external instruments, or performing in real time. I'm working completely in the box, and only currently worried about recording.
.
I got the impression that you are running synth plugins and such, am i wrong ? Even synth plugins that react instantly to you pressing a key on a keyboard runs in real time and need low latency audio otherwise the sound wont come instantly. So its pretty much the same as monitoring a mic on a recording interface only the sound is just a one way conversion from digital to analog, the mic is converted to digital and then to analog again so that you can hear it, this is usually called DAW monitoring.
When you play an mp3 from an album for example you dont need real time audio, it doesnt even matter if the song takes a second or so before it starts, but with a synth that would be a catastroph, it would be unplayable.

Quote:
Originally Posted by Ankhet View Post
On another note, how do I stop my computer from putting a "higher priority on other things than audio"? You also say "i think your main problem is likely to be computer drivers that behave badly". What does that mean specifically and how do I get them to behave well?
There are tweaks that can be done to tune things up, but you may also need to deactivate WIFI and LAN before you record. I wont go into detail for now before the results of the DPC checker test


Quote:
Originally Posted by Ankhet View Post
@inversound @Neonknight What kind of external soundcard do you recommend? Does it connect via USB or some other way?
USB is common for especially simpler interfaces, firewire and sometimes USB for bigger more advanced interfaces. Lets test first and see, maybe you can get your internal to work for your purpose.
Old 14th November 2015
  #10
Here for the gear
 

I had a similar problem with my focusrite interface. Never had an issue with my Presonus, but that could be merely a driver kinda thing. What I ended up having to do, was go into my windows control panel and change all of my settings (for my sound card and the interface) over to 24/48000. I tried this same thing with 24/96000... but to tell the truth, I've heard no audible difference between 48000 and 96000, but that's another topic that could be debated. 24 is the key.

Then what I did, was to set my windows sound card to default (for both playback and record) and disabled my interface for both playback and record. Your DAW will know which to use for recording and playback if you set it up correctly. If not... it can easily be fixed within the DAW.

The only problem with this is now you're not able to listen to Youtube or any other audio inputs through using your interface as a master controller. You'll only be able to hear sound through your DAW when recording and mixing. This really isn't a huge deal, because... it fixed it.

The issue (as I'm told by Sweetwater and Focusrite technicians) is... Windows sound cards can often be a pain. If you don't separate which function is using which sound card... there can be cross-talk and it just screws things up. It's not always the case, but it was in my instance. Like I said above... I still can't get 96khz to work without some random audible clicking, but 24/48000 works fine.

Another note... I'm also using a laptop with an i7 2.4ghz processor and 8G RAM. I don't think it's 100% your computer's capability to process things.

Another note... your buffer size does play a big role. I know that you said it before, but make sure that while listening... you have it set really high. Meaning... your DAW will say something like 960 samples. If this sample size is too low... your system can't process all of the plugs fast enough and lead to crackling. If you're recording... this isn't where you want to set your buffer. You may have to turn plug-ins off on other channels and record clean, but then you'll want to set your buffer about mid way or slightly closer to 1. Just experiment with these two things. You'll figure it out eventually. It can be a real pain until you find the right recipe. Good luck!

Last edited by DWdrummer67; 14th November 2015 at 02:27 PM..
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