A theory about digital ticks and pops and solution

[acorec]

We have been chasing the oft-referred to pops and clicks, ticks, etc., for the past few weeks. We have taken pretty much all so-far suggested steps to optimize our machine and operating system, and they all cumulatively have improved things, one little bit at at time. I've learned a lot about keeping most of my tracks archived and not using effects while tracking, as well, which helps a great deal in the quality of our tracks.

This forum has been a HUGE help in finding these steps and making it possible for us to make progress. Thank you so much.

But there seem to be other causes, as well, that remain constant, and I am starting to form a hypothesis, and would really appreciate feedback from some audio-techie expert(s) who might have time to consider and discuss this question.

Is it possible that to some extent, those ticks that seem to remain despite all our computing power and optimization steps, and occur with maddening consistency when recording high frequency sounds are due to the fact that there simply aren't any frequencies available in that input signal that are much above the human's "stardardized" audible hearing range (20kHz or so, right?), and I'm using the 96kHz setting to record my audio?

I know that parapgrah was long and confusing. Let me try restating and also giving examples. If the source of the audio is not delivering any significant amplitude (and even perhaps totally cut, no sound at all) in the frequencies above the presumed standard of 20 kHz (for which 44.1kHz sampling rate is fine), but we're sampling at 96 Khz, is it possible that there is "empty space" at the highest frequencies captured by the 96kHz sample rate? Could this also account for noise that intrudes when we use a guitar effects box where the effects might have heavily attenuated the highest frequencies or even might not have any in that range?

Maybe this is a stupid newbie question, or maybe I'm onto something.

Here's what got me thinking this. We had 5 songs done for us for free by a nice guy with a nice new studio and DAW who needed guinea pigs. He did a nice job, but when he gave us the mixdown, all we got were the .cda files. For various reasons, I decided to record the 5 .cda songs off the CD into Cakewalk Home Studio XL2004 via our Delta 1010's monitor mixer (so I could put them on the same CD as the new stuff we've been recording without getting clipping).

One song has some very high frequency chimes in the beginning. It's a very quiet place in the song. And I got this AWFUL zipperlike ticking at the high point of every beautiful chime note! It was really bad. What fixed it: recording at 44.1 and 16 bits.

That got me thinking: Was Cake Walk and/or the Delta 1010 AD converter (as would any other recording system capable of greater word length as well as higher sampling rate) filling in the missing digits with noise? Could I fix this problem by recording these 5 songs at 16 bits and 44.1kHz sampling rate? I tried it AND IT WORKED!

And what about analog input devices like clunky old microphones that just can't get the higher frequencies? Or even digital devices, (like a guitar effects box that's got the higher frequencies squished right down or out) that are in the input chain during recording? It's not even clunky, it just isn't "optimized" for high fequency sampling.

Is this possible that these each get more noise at higher sampling rates and even word lengths because there's just nothing there for the A/D converter to work on? It's not usually very noisy, actually, but when you're really listening, it's definitely there. And much worse on high frequency sounds. Plus it can "build up" if enough tracks each have even a little bit of it.

I'm plan to experiment with this hypothesis as I work to improve the quality of the tracks we're laying down. I didn't find a lot about this topic in any of the knowledge bases in mine or my partner's research. That doesn't mean it wasn't discussed, just not in conjunction with the "ticks and pops" problem (which we saw EVERYWHERE).

I thought I might throw this out, though, to see if others have more information on this possibility, and can give some pointers an how/when to use highest quality recording capabilities and when to pull them back a bit to get better results?

Thanks in advance, and maybe I will get a chance to run my question in person past some folks this next weekend, at Jam Fest. I'm looking forward to this, but nearly as much as vtgreen81 is.

The pops and clicks almost always have to do with drivers/hardware setup. Windows PCs are the volkswagons of our time. You can assemble a host of literal junk inside a standard PC case and have a working PC. But, that is why most PCs suck for audio. I will venture to say that all the Dell and HP PCs are crap, bar none, for DAW work. A properly built PC for great reliable recordings cost ~$2000 in parts alone. The problem with building a PC is that people almost always tend to buy the cheapest components without thinking about it. The PC manufacturers are generally worse at this as they focus on cost/unit solely. I have worked on/with/built thousands of PCs in my time and it never ceases to amaze me, even today, at the cheap junk that Dell, HP and a number of other "PC Manufacturers" use to build their products. The Only exception to this is the laptops,

Now, to your problem. First I would load all the latest drivers for your video and soundcard (as well as the motherboard). Second, put your audio card as far away from your video card as possible. Now, if you have enough memory, and the soundcard is far away from your video card and all the drivers are updated, then the last thing you need to do is turn off ALL programs running in the background. You want your processor to focus on the recording software only. This should really stop the popping and clicking.

 

[vestast]

I will venture to say that all the Dell and HP PCs are crap, bar none, for DAW work. ....

My "crap" HP PC has been rock solid and NEVER crashes on me. I leave this thing running for weeks, and often only shut it down if I'm leaving for the weekend. And on top of that I'm using a firewire devices with 8 inputs again with no issues...

Most times just doing a few tweaks to the OS like disabling a few services works. Disconnecting from the internet and disabling any antivirus is important...

This website is great for tunning your PC

http://www.musicxp.net/tuning_tips.php

 

[boingoman]

Yup, sounds like a hard/soft ware issue. No matter the sample rate, the frequencies captured are limited to the piece of gear with the lowest frequency response. If you have a mic that is only sensitive to 24k, you could sample as much as you want, and you aren't gonna get any info over 24k. You are just gonna load your system with ever-increasing amounts of data to move.

Hubcap- thanks for the info. I don't really think cheaper came into play as far as the guard filtering at higher sampling rates, just the fact that a not-as steep filter acts more smoothly, as you pointed out.

 

[acorec]

My "crap" HP PC has been rock solid and NEVER crashes on me. I leave this thing running for weeks, and often only shut it down if I'm leaving for the weekend. And on top of that I'm using a firewire devices with 8 inputs again with no issues...

Most times just doing a few tweaks to the OS like disabling a few services works. Disconnecting from the internet and disabling any antivirus is important...

This website is great for tunning your PC

http://www.musicxp.net/tuning_tips.php

You are one of the lucky ones. I setup/repaired most of the PCs for the University of Massachusetts in Boston for 4 years. U-mass Boston buys all the computers for all of the state 2 year colleges as well as themselves. I have seen/worked on/repaired more computers than you would see if you walked through Comp-USA and looked for a month straight.

Fast Forward to now. I am an engineer at a small company and see the HPs and Dells that come through here. No offense to you, but they are still utter crap. Many work, many more have all kinds of problems. I built my own computer here as the IT guy offered me a nice Dell. I told him I wanted this, and that and he ordered it for me.

If yours works good, then fine. But someday you may suffer with a new Dell or HP. Maybe they will get better, but go look at the computer systems that Pro-Tools specs out. You will find a *slightly* different set of components.

 

[vestast]

You are one of the lucky ones....

I was thinking that after I posted that, that maybe I should have said I probably got lucky. I absolutly agree with you on the Dell's as they are our primary systems at work for desktops and some of the 1u rack servers.

I managed to get this PC at an extreme discount as it was a clearence. It was so cheap that there was no way I could build one cheaper with Canadian prices on stuff being a little more than $US..

I did a few upgrades on it but nothing major, and like you I would recomend going the custom built way if you can...

Anyway - Back to the thread !

 

[VTmosaic]

The pops and clicks almost always have to do with drivers/hardware setup. Windows PCs are the volkswagons of our time. You can assemble a host of literal junk inside a standard PC case and have a working PC. But, that is why most PCs suck for audio. I will venture to say that all the Dell and HP PCs are crap, bar none, for DAW work. A properly built PC for great reliable recordings cost ~$2000 in parts alone. The problem with building a PC is that people almost always tend to buy the cheapest components without thinking about it. The PC manufacturers are generally worse at this as they focus on cost/unit solely. I have worked on/with/built thousands of PCs in my time and it never ceases to amaze me, even today, at the cheap junk that Dell, HP and a number of other "PC Manufacturers" use to build their products. The Only exception to this is the laptops,

Now, to your problem. First I would load all the latest drivers for your video and soundcard (as well as the motherboard). Second, put your audio card as far away from your video card as possible. Now, if you have enough memory, and the soundcard is far away from your video card and all the drivers are updated, then the last thing you need to do is turn off ALL programs running in the background. You want your processor to focus on the recording software only. This should really stop the popping and clicking.

We bought top components, did not skimp, the box was built for this purpose, but that doesn't mean we don't have a conflict somewhere. And PC's are a nightmare, I agree.

I am going to experiment some more this weekend, but I am beginning to doubt that it's the processor not keeping up. We have a huge amount and I have everything possible shut down. The box is dedicated.

I have applied all the latest drivers; this machine is current.

I'm wondering: do I have a bottleneck somewhere, a bus somewhere that is 16bits? Is that possible? I have to get specs and look at them and talk to my PC guy.

 

[chessrock]

. . . the last thing you need to do is turn off ALL programs running in the background. You want your processor to focus on the recording software only. This should really stop the popping and clicking.

I didn't think of that one, but yea, this one is very important. Take careful note of it. Unnecessary stuff running in the background can totally screw with the audio.

 

[VTmosaic]

Conversion between digital and analog domains require filters to remove out-of-band artifacts. The farther away from the highest audible frequency you can make the sampling frequency, the simpler (read cheaper) the filtering that is required and the less audible the side effects of the filtering are.

There are people who claim that higher sampling rates provide better sampling period but this is simply not borne out by the science behind digital signal processing that I studied in grad school.

Bit resolution (i.e.going to 24 bit from 16 bit) is WAY more important than sampling rate.

The clicking is due to the larger amount of data being moved...96/24 creates about 3.5 times as much data as 44.1/16, it's that simple. Pops and clicks are nothing more than missing samples in the stream. They are missing because the computer can't keep up.

Now, I am interested in learning more about "out of band artifacts," 'cause that sounds something like my original theory. The ticks and pops I'm getting are not the same as those I get when, for instance, I forget to increase my DMA buffer size between the time I was doing MIDI and starting to record audio.

I really do think I have enough processing and ram, though it's still possible I have a hardware bottleneck or conflict. Still, people with this same setup have no problems, I have heard. 'Course every single PC is unique, I know, even if they both came of the same assembly line right next to each other.

And when I record, I watch that processor indicator in my HSXL2004 display and it's sitting steady at only 3 %! Surely it would have to indicate at least some increased usage before it started dropping sample digits, wouldn't it? At least most of the time? I will keep experimenting, though.

Question: when you say, "The farther away from the highest audible frequency you can make the sampling frequency, the simpler (read cheaper) the filtering that is required and the less audible the side effects of the filtering are." Which direction do you mean? If the highest frequency I want to record is higher than humans are supposed to be able to hear, should I even bother? So, let's say I want to record highest frequency of 24,000hz. Are you saying go higher, way higher, like to 96hz? Or lower, like 44.1kHz? Which direction "farther?"

And, are you also suggesting that we apply a low pass or high cut filter or soemthing like that when sampling at a high frequency to get rid of these "artifacts?" Are you saying that we are picking up noise because we're picking up stuff that should be filtered out?

I know I sound dumb, but I'm not. Just new at this. Thanks for you help.

 

[VTmosaic]

I didn't think of that one, but yea, this one is very important. Take careful note of it. Unnecessary stuff running in the background can totally screw with the audio.

Gotcha. Our machine isn't connected to the internet. I uninstalled antivirus, disabled the printer ports, basically went through every checklist I found to optimize the machine for audio.

We are getting such nice clean recordings at 44.1/16 bits. But I sure do want to know if there's any point at trying to get them at 96/24. Surely there must be some advantage?

(I know of one: the local radio DJ I know say they have to have their files in 24 bit format... so if we ever get our little radio promo written and recorded for a friend's and bandmember's show, I'll be glad then I had 24 bit.)

 

[VTmosaic]

My "crap" HP PC has been rock solid and NEVER crashes on me. I leave this thing running for weeks, and often only shut it down if I'm leaving for the weekend. And on top of that I'm using a firewire devices with 8 inputs again with no issues...

Most times just doing a few tweaks to the OS like disabling a few services works. Disconnecting from the internet and disabling any antivirus is important...

This website is great for tunning your PC

http://www.musicxp.net/tuning_tips.php

Howdy, jf neighbor! How are you? Dry?

Thanks for that tip! I'll follow up on this. I figure even if it doesn't resolve my problem, it's still bound to be good!

And you give me hope that it isn't a hopeless cause because we are using a PC!

 

[vestast]

Howdy, jf neighbor! How are you? Dry?

Thanks for that tip! I'll follow up on this. I figure even if it doesn't resolve my problem, it's still bound to be good!

And you give me hope that it isn't a hopeless cause because we are using a PC!

Howdy right back at ya ! No rain since we got back... Figures eh ?

When I was running maudio hardware, I had the same pops and glitches that you are experiencing. Running those tweaks that I linked helped...

I don't know if this might help or not, but a few people I have talked to in the past have mentioned that if you have a hyperthreading PC, sometimes disabling hyperthreading improves performance issues.

I'm not sure how you do this (possibly through the device manager or the BIOS ?) as I'm not running HT, but maybe your PC guy could help you. I think it's something worth exploring. (although I'm sure a dozen peps will jump in here and argue on that one )

One other thing (I'm not sure if you already mentioned this). Running a seperate hard drive for your audio and a seperate hard drive for your apps really helps also..

Good luck and say hi to Mr. VT for me (eh)

 

[mshilarious]

Now, I am interested in learning more about "out of band artifacts," 'cause that sounds something like my original theory.

. . .

Question: when you say, "The farther away from the highest audible frequency you can make the sampling frequency, the simpler (read cheaper) the filtering that is required and the less audible the side effects of the filtering are." Which direction do you mean? If the highest frequency I want to record is higher than humans are supposed to be able to hear, should I even bother? So, let's say I want to record highest frequency of 24,000hz. Are you saying go higher, way higher, like to 96hz? Or lower, like 44.1kHz? Which direction "farther?"

And, are you also suggesting that we apply a low pass or high cut filter or soemthing like that when sampling at a high frequency to get rid of these "artifacts?" Are you saying that we are picking up noise because we're picking up stuff that should be filtered out?

The 'out-of-band' artifacts are called aliasing. When you have a frequency that exceeds 1/2 your sample rate, it can erroneously appear as a lower frequency--within the audible spectrum. Since a higher frequency is masquerading as a lower frequency, it's called aliasing.

To prevent that, A/D converters have anti-aliasing filters. These prevent ultrahigh frequencies from arriving at the converter.

When you have a 44.1kHz sample rate, you need a really steep filter, since it must not touch the audio at 20kHz, but have eliminated all frequencies above 22kHz.

With a 96kHz sample rate, since you have a theoretical 48kHz max frequency, that filter can be much more gentle, but it will typically still start attenuating at around 20kHz.

Either way, there is a filter that prevents aliasing. So unless your converters are defective, it shouldn't be an issue.

If you want to record a 24kHz frequency, you will have a few barriers. You need a mic with ultrahigh response. Most mics start diving at 16kHz or so. You have to use a 96kHz converter that passed that frequency. Just because they theoretically can doesn't mean that they do. All the other gear in the chain has to pass 24kHz too. Your tweeters have to be capable of 24kHz--some are, most aren't. Finally, your ears have to be capable of hearing it

 

[VTmosaic]

The 'out-of-band' artifacts are called aliasing. When you have a frequency that exceeds 1/2 your sample rate, it can erroneously appear as a lower frequency--within the audible spectrum. Since a higher frequency is masquerading as a lower frequency, it's called aliasing.

To prevent that, A/D converters have anti-aliasing filters. These prevent ultrahigh frequencies from arriving at the converter.

When you have a 44.1kHz sample rate, you need a really steep filter, since it must not touch the audio at 20kHz, but have eliminated all frequencies above 22kHz.

With a 96kHz sample rate, since you have a theoretical 48kHz max frequency, that filter can be much more gentle, but it will typically still start attenuating at around 20kHz.

Either way, there is a filter that prevents aliasing. So unless your converters are defective, it shouldn't be an issue.

If you want to record a 24kHz frequency, you will have a few barriers. You need a mic with ultrahigh response. Most mics start diving at 16kHz or so. You have to use a 96kHz converter that passed that frequency. Just because they theoretically can doesn't mean that they do. All the other gear in the chain has to pass 24kHz too. Your tweeters have to be capable of 24kHz--some are, most aren't. Finally, your ears have to be capable of hearing it

But if this were aliasing, then reducing the sampling would NOT reduce the noise, it would increase it, right? My problem occurs anywhere above 44.1/16. I can't even go to 44.1/24 bits without increasing the regularity and presence of this noise that only seems to appear on high frequency sounds.

This noise of mine happens on female vocals, chimes, crickets and wailing guitars. These aren't in the 24kHz range, are they? Blue Bear mentioned transients... but I have tried the recording levels so low you couldn't even see the bumps on the wave form and still got it.

If I record at 96kHz/24bits, my cricket has a click on the highest notes. It's very regular. Same thing with the chimes. I have to reduce to 44.1kHz to get an acceptable recording. So, that's the opposite of the aliasing issue, right?

Also, don't particularly want to record high frequencies like 24kHz. I don't sing that high.

 

[VTmosaic]

Howdy right back at ya ! No rain since we got back... Figures eh ?

When I was running maudio hardware, I had the same pops and glitches that you are experiencing. Running those tweaks that I linked helped...

I don't know if this might help or not, but a few people I have talked to in the past have mentioned that if you have a hyperthreading PC, sometimes disabling hyperthreading improves performance issues.

I'm not sure how you do this (possibly through the device manager or the BIOS ?) as I'm not running HT, but maybe your PC guy could help you. I think it's something worth exploring. (although I'm sure a dozen peps will jump in here and argue on that one )

One other thing (I'm not sure if you already mentioned this). Running a seperate hard drive for your audio and a seperate hard drive for your apps really helps also..

Good luck and say hi to Mr. VT for me (eh)

I haven't tried messing with hyperthreading. I'll ask my pc guy (he's a musician, too, so we're tempting him out with a recorded jam session).

We do have the separate hard drives. But you do remind me that there is one thing that might be related to this: Cakewalk HSXL2004 was intially installed with the the "audio files" folder set up on the C: drive, instead of the dataspace drive (D. I tried to move them by changing the settings, but the software freaked out so I put them back.

Maybe, just maybe, the problem is that my audio files are on the C: drive and when I'm recording at the higher sampling rates, it's causing a problem!

I'm gonna see about fixing that.

I have run down the complete list of tips to optmize XP Pro, and done almost all of them. No impact.

I've even being doing things like bouncing down to a single stereo track of all previous tracks and archiving the individual tracks so I reduce load on the processor while laying more tracks in the project.

Also these resource-saving tips are great even if they don't fix this problem. It seems to make no difference.

I"m gonna check out that hard drive thing. I did know we want separate hard drive which is why we had it built this way. But having those audio files on the C: drive may just be thwarting those good intentions!

Wouldn't it be cool if that were it! Treeline said he thought it might be just some little setting among the 100's of such between the OS, soundcard and Cakewalk!

I'll indeed say hey to mr VT, eh?

 

[VTmosaic]

We have a workaround!

We ripped the PCI card out of our fancy new machine and plugged it into the slot of one of our business machines and it works so beautifully I almost experienced ... nirvana shall we say, just listening to vtgreen's chimes FINALLY sounding like the real things!

We are gonna rework the features and hardware config of the new machines, get rid of both RAID and super duper display board or reconfigure them to work well with the Delta1010, leave fast processor and mega ram and big hard drive.

Meanwhile, we STILL can keep on working on our music/recording project!

But, I did learn one important lesson. PC's are powerful enough to do a decent job of recording live audio at 96kHz/24 bits as long as everything is working properly together, no bottlenecks. They don't even have to be all that suped up to do a good job! Also, we were able to reduce the DMA buffer setting to a mere 512 on the old machine and still record smoothly with no dropouts! So, it was a resource problem on the new computer, and it just could not handle those really high frequencies? (As you wise old hands predicted, sure enough.) The new machine's DMA buffer size was maxed, 2048, and still we had problems.

Oh, we might drop back to Windows2000, as well, which is another thing that's different about our old machine. Sweet, stable Windows2000.

Maybe our PC guy will become a member of this forum and become a wizard at setting up these DAW's (I can hope)... I did send him a link to this forum. He is a musician (keyboards), and DJ (I think), as well.

Thanks a LOT for all your suggestions. They cumulatively helped us to follow a path to a solution! Yippee. Feel like we've passed another milestone in our homerecording/studio odyssy.

 

[acorec]

We have a workaround!

We ripped the PCI card out of our fancy new machine and plugged it into the slot of one of our business machines and it works so beautifully I almost experienced ... nirvana shall we say, just listening to vtgreen's chimes FINALLY sounding like the real things!

We are gonna rework the features and hardware config of the new machines, get rid of both RAID and super duper display board or reconfigure them to work well with the Delta1010, leave fast processor and mega ram and big hard drive.

Meanwhile, we STILL can keep on working on our music/recording project!

But, I did learn one important lesson. PC's are powerful enough to do a decent job of recording live audio at 96kHz/24 bits as long as everything is working properly together, no bottlenecks. They don't even have to be all that suped up to do a good job! Also, we were able to reduce the DMA buffer setting to a mere 512 on the old machine and still record smoothly with no dropouts! So, it was a resource problem on the new computer, and it just could not handle those really high frequencies? (As you wise old hands predicted, sure enough.) The new machine's DMA buffer size was maxed, 2048, and still we had problems.

Oh, we might drop back to Windows2000, as well, which is another thing that's different about our old machine. Sweet, stable Windows2000.

Maybe our PC guy will become a member of this forum and become a wizard at setting up these DAW's (I can hope)... I did send him a link to this forum. He is a musician (keyboards), and DJ (I think), as well.

Thanks a LOT for all your suggestions. They cumulatively helped us to follow a path to a solution! Yippee. Feel like we've passed another milestone in our homerecording/studio odyssy.

I can tell you that Win 2000 can be a problem with some hardware drivers. Stsrt by looking at ALL your drivers on the companies websites. Make SURE that they say that their drivers are WIN2000 compatable. Some ARE NOT.

I ran into this sometime ago. This could be a thing of the past, but, it is worth checking out anyway. WIN 98 (I assume that is what you are using on your buisiness computer?) is much more compatable with old hardware/drivers even as new as the year 2002. Also, if your motherboard has built-in audio hardware on the mother board itself, I have had incredible problems with PCI audio cards. It is just impossible to bypass the internal audio hardware and have a smooth functioning machine. Some Video cards have problems with interrupts that produce popping and clicking. One last thing, RAID will KILL your processing time and slow the machine waaaay down. Get rid of it.

These are some areas to be carefull with.

Good Luck sorting it out, I have been there/done that so many times.