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 plugin because it happens with the Vestige plugin 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 a lot 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!
Comments
A couple of common culprits are the buffer size or not having th
A couple of common culprits are the buffer size or not having the latest soundcard drivers (or you could even try rolling back to a previous version of that driver to see if it makes a difference). Since you're on Windows 7, you could try downloading a trial version of a different DAW to see if LMMS is the source of the problem. Actually, Presonus gives away a free version of their Studio One software. But Chris is right, if you want quality recordings, onboard soundcards like the Realtek are not a good way to go. Even a cheap USB/firewire interface will probably perform better.
Your Realtek card wasn't designed for the rigors of multitrack r
Your Realtek card wasn't designed for the rigors of multitrack recording and playback... it was designed for gaming, playing CD's and audio playback on the web.
The crackling you are hearing is the card reacting to buffer issues, and the fact that it's being asked to do far too much by the DAW.
Consider something like a two channel Presonus USB audio I/O.
I agree with Donny And also you talk about (Vestige plug-in play
I agree with Donny
And also you talk about (Vestige plug-in playing a "thunder" instrument, another Vestige running "rain")
If those are Virtual instruments, computing the sound in realtime, it could be your source of problems. Render them in an audio track and deactivate the VSTI
Let us know how it comes up !
Ahhh... VSTi's... yeah, I didn't even factor those in.... seriou
Ahhh... VSTi's... yeah, I didn't even factor those in.... serious memory and processor hogs. And it doesn't matter how loaded you are in your CPU and Ram.
In short, you are asking/expecting that built-in audio card to do something that it was never designed to do in the first place.
That Realtek card simply cannot handle the power, speed and latency needed to efficiently run production audio. You could spend the next year trying to tweak that Realtek to work with your DAW program, and it ain't gonna happen... you'll find that it's a futile effort.
(Basically, you're trying to tow a 100 ft yacht with a '74 Pinto). ;)
Drop a few bucks on a real audio I/O... you can get into a very nice USB audio I/O with good mic pres and converters for around $150.
I promise that if you do, you'll never even think about that Realtek card ever again.
FWIW
d/
Hi everyone, Thank you all for your answers. Several of you ment
Hi everyone,
Thank you all for your answers. Several of you mention buffer size. I thought this potential source of crackling was ruled out by the fact that the problem still occurs even when I raise the buffer size to 16,384 frames. Is that not the case?
Also, is it even possible to change sound cards in a laptop? I have no idea how to do that.
DonnyThompson You mention about the VSTi's being "serious memory and processor hogs. And it doesn't matter how loaded you are in your CPU and Ram." Why wouldn't it matter how loaded you were in CPU and RAM?
pcrecord You said "If those are Virtual instruments, computing the sound in realtime, it could be your source of problems. Render them in an audio track and deactivate the VSTI". I actually tried that already and no luck. However, I think I'll try it again and this time remove the original VST track completely instead of just muting it. I'll let you know if that works.
@Chris Perra Thank you for mentioning that not only the sound card but the compatibility between the sound card and the DAW might be an issue. I don't know what "limiter" or "stereo master out" means?
apstrong good idea on the free demo download I will try that. What does a USB/firewire interface do? I thought this was for external instruments? I am working completely in the box.
Thank you all so much for your time!
"...You mention about the VSTi's being "serious memory and proce
"...You mention about the VSTi's being "serious memory and processor hogs. And it doesn't matter how loaded you are in your CPU and Ram." Why wouldn't it matter how loaded you were in CPU and RAM?..."
what I was saying - or at least trying to say - was that it wouldn't make any difference with the RealTek audio card you currently have.
Of course you need sufficient amounts of processing power and memory to run these apps. But all the power in the world won't matter with that Realtek card. You're trying to push Niagara Falls through a kitchen faucet.
I'll say it one more time...The Realtek card was not designed to do what you want it to do.
Forget trying to tweak buffers, forget trying to tweak your PC to get it to be more efficient... forget trying to make that card work for audio production.
It was not designed to handle the rigors of digital audio multi track production. The converters are cheap, the latency is ridiculous, and the drivers used in the implementation of the card are nowhere near stable or efficient or powerful enough to handle DAW production.
If you'd hooked up an MBox, or a Presonus, or even a low budget Berhringer USB audio I/O, you'd be recording by now, instead of asking what workarounds are possible with that "toy" audio card.
I don't know how else to say it. I've told you what your best option is... but do what you wanna do.
@DonnyThompson Thanks for getting back to me. Oh, I'm completel
DonnyThompson Thanks for getting back to me. Oh, I'm completely convinced I need to move up from Realtek. But I am trying to gather information, and especially check my comprehension because I am self taught and so new. For instance, do you think you could specifically address my question about buffer underruns? Am I correct that at a buffer size of 16,384 frames having a buffer issue would be nearly impossible given my specs? Or is it not as simple as that?
Also, @everyone I forgot this in my last post, but 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.
Let's consider what VSTI does. Virtual instruments are using sam
Let's consider what VSTI does. Virtual instruments are using samples most of the time. Samples are recordings of the instrument notes and assembled in such a way that you can play them on a midi controler (keyboard). The computer/DAW, processes the vsti like multi-channels recording because with most VSTi you can play more than one note at the time. So if I play a standard chord on the piano, It may have 3 sounds to play and also some effects, some filters and reverb working at the same time. All this adds up and must be processed by your computer and that then encodes it through the driver of the soundcard.
An onboard soundcard doesn't have the circuitry to react fast enough to follow what the DAW is sending. Usually we can tweak the buffer but it will take you so far. BTW a too high buffer setting maybe worst because the buffer of a sound card usually use the computer's memory and not an onboard chip (if it has a an onboard buffer, it is still limited) Usualy we start the buffer at 256 and then try to go up or down and check the results.
What does a USB/firewire interface do?
It is an audio interface that is independant from the motherboard of your computer and it is design to compile and convert digital information from your DAW to analog outputs for your speaker and the inverse, converting analog to digital. But it does it with better driver and construction to help lower the latency and risk of dropping the audio signal. Also, a lot of audio interface come with preamps, that let you plug a mic or an instrument to record them. The preamp amplified the very faint signal to a record level and can come in many quality that will affect the sound in good or bad ways..
What does it mean that the crackling is distinctly different when I change the samplerate in Speaker Properties from 48000 Hz to 44100 Hz?
Others can explain it better but the hz (hertz) are the number of time the sound sampled before it is sent to disk as data. Analog audio is recorded by sampling it 44,100 times per second, and then these samples are used to reconstruct the audio signal when playing it back. The cracking is different because your computer/soundcard work a little bit harded to process 48khz than 44.1khz
If that Realtek has a "sealed" sampling rate, and you are trying
If that Realtek has a "sealed" sampling rate, and you are trying to play or record using another sampling other than what the card is able to reproduce, you'll run into all kinds of problems.
You have a lot to learn here as a beginner. And I mean a lot. If you really want to do this, I would suggest an online course in beginner level digital audio production to get you started.
Last time I checked, Berklee (yes, that Berklee, in Massachusetts) was offering a beginner's course on introduction to digital audio, and it was free.
Yup... still there:
The sample rate change in the clicking is due to you playing bac
The sample rate change in the clicking is due to you playing back something that was recorded at a different sample rate. If it was recorded at 44.1 and played back at 48 or visa versa the pitch will change.
Are you willing to spend some money? If so...
Get a usb interface. It does the job of your realtek soundcard. Replacing it when you select it. It plugs into you computer via usb. You can plug Midi or Mic/1/4 inch cords into it.
The time and mental energy that you will save is well worth the money. With things like a realtek, if it doesn't work there's not too much you can do. You can try and find a third party asio driver that can work with it but because it's made for many different soundcards it won't necessarily work. Looking at my realtek inside cubase in my laptop it has a buffer of 2048.
It isn't adjustable.. So I can listen back or mix ok, but can't track or do midi without 48 mill of latency.
A stand alone usb interface with bundled software will be the fastest way to get you on your way.
Also if you are committed to LMMS make sure what you get is compatible with it.
Chris Perra, post: 417927, member: 48232 wrote: The sample rate
Chris Perra, post: 417927, member: 48232 wrote: The sample rate change in the clicking is due to you playing back something that was recorded at a different sample rate. If it was recorded at 44.1 and played back at 48 or visa versa the pitch will change.
Good point Chris, If there is a track in the wrong format they won't play right. But I think the crackling is happening in both format and the user wanted to know why it was different. ;)
Elia : I think you have a concensus : BY AN NEW AUDIO INTERFACE !! (even the sound quality and noise ratio will be better)
I've never used LMMS... Or a Realtek sound card to do recording.
I've never used LMMS... Or a Realtek sound card to do recording.. If it were me I'd try and get a basic interface That has asio drivers and bundled Daw software if you aren't married to LMMS. That way you know The soundcard/interface is designed to work with the Daw. Or maybe see if there is someone out there that has a working setup sound card wise that works well with LMMS and get that soundcard.
You can do recording with a built in P.C. sound card but it can be infuriating to get things to work properly as there are so many variables. Removing the potential incompatibility problems of the Daw and soundcard is a good way to start.
Volume levels and track count shouldn't matter. If your summed output volume/stereo master out is under 0db and your soundcard buffer is high enough and you have enough processing power and ram.. Which it seems that you do.
Are you using a limiter on your stereo master out?