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  #1  
Old 09-07-2012, 09:42 AM
meambobbo meambobbo is offline
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help me understand audio timing in digital format

First let me state what I want to know, then I'll explain why I want to know it.

Cubase has a track delay in miliseconds you can use to hopefully sync up audio, or move it precisely in place. The smallest increment you can move a track is a hundredth of a millisecond.

Now, if my audio track is sampled at 96 kHz, that would mean there is one sample every .01041666... milliseconds - this means you cannot move a track by one sample length, but by a little less than that. The more hundredths of a ms you move a track by, the more out of sync its samples would be with respect to another track that was previously in sync with that track.

Similarly, a delay unit on a multi-fx processor lists delay times in milliseconds. These delay times are generally in milliseconds, but similar to above, the sample rate doesn't fit neatly into 1 ms intervals. With a multi-fx unit synchronized to a common clock for 2 outputs (like a stereo signal over SPDIF), how can that be?

Are the neat .01 ms or 1 ms increments just a guideline and the program or multi-fx unit just gets as close to that increment as possible while maintaining all samples in sync with the sample rate clock? Can Cubase actually move the audio by the increments it states, so that each sample for two different audio tracks can actually be slightly out of sync?

Ok...so the reason I'm asking is that I have a Pod HD and tend to find the onboard cabs a little lacking, especially for a thick yet crisp metal tone. I like to use dual amp tones, but use the same amp and relatively the same settings on each amp. I just use different cab/mic's for each of the two available channels. Thus, I can combine a cab with a good low-end response with one with a good high end response.

I noticed some combinations sounded dull and muffled. I then noticed adding an EQ with completely neutral settings to just one of the channels could improve the tone. My theory on why this worked was because the mic distances modeled in the cab/mic selections were slightly different, causing a comb filter effect. The EQ applies a very small delay to the signal, syncing up the two channels (maybe not completely, but closer to optimal). All of a sudden, the bright highs are back in the tone.

I want a more precise way of handling this. I don't want to have to stick neutral EQ's in my patches, eating up precious DSP. And it's difficult to determine if this process could work even better if I had a more precise means of time-shifting one of the channels.

I panned each channel hard left/right, and recorded each channel as separate tracks in Cubase, hoping to identify the exact amount of time-shifting necessary to achieve the best possible phase-correction. I tried to use Cubase's .01 ms precision increments but every change made the tone worse - nothing remotely resembling using an EQ in the Pod.

So I'm wondering why an EQ effect in the Pod works, but Cubase's time-shifting is unable to produce the same effect. Either my entire theory about the channels being out of sync is wrong, the delay from the EQ is < .01 ms, or there's some mismatch between the .01 ms increments and the actual sample length.

Any insight into this is much appreciated.
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  #2  
Old 09-07-2012, 09:46 AM
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Scott Peterson Scott Peterson is offline
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When you move tracks around, you create phase cancellation and that has the effect of an EQ in many cases - both in lessening and in enhancing different frequencies. It changes as you move tracks around.

You get the same thing when you dual mic an amp in the analog realm. Where those mic's are in proximity to each other can have a dramatic effect.
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  #3  
Old 09-07-2012, 09:55 AM
Will Chen Will Chen is offline
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Quote:
Originally Posted by meambobbo View Post
...I noticed some combinations sounded dull and muffled. I then noticed adding an EQ with completely neutral settings to just one of the channels could improve the tone. My theory on why this worked was because the mic distances modeled in the cab/mic selections were slightly different, causing a comb filter effect. The EQ applies a very small delay to the signal, syncing up the two channels (maybe not completely, but closer to optimal). All of a sudden, the bright highs are back in the tone...

So I'm wondering why an EQ effect in the Pod works, but Cubase's time-shifting is unable to produce the same effect. Either my entire theory about the channels being out of sync is wrong, the delay from the EQ is < .01 ms, or there's some mismatch between the .01 ms increments and the actual sample length.

Any insight into this is much appreciated.
I discovered this behavior a while back and posted about it here including clips: http://www.thegearpage.net/board/sho...d.php?t=964530 users anot and mpr even scoped out the frequencies attenuated. The POD takes .0625 ms to process EQ according to the data in that post.

It works in the POD because the unit does not appear to precalculate latency. In most multi-fx processors and all DAWs I've ever worked on, the latency is calculated and the all audio is offset to ensure everything aligns correctly. The POD does not do this. As such, if a signal in either of the parallel chains incurs more latency than the other, you will experience a comb filtering effect.
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  #4  
Old 09-07-2012, 09:59 AM
meambobbo meambobbo is offline
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Right - the comb filter effect. That is what I believe is happening in my Pod between each channel due to different cab/mic selections. When I use an EQ effect to fix this, I am not using the EQ to filter/boost any frequencies. The settings are completely neutral. If I toggle it on/off behind an amp on a single-amp patch, it is impossible to determine if the EQ is on or off - it has no effect on the tone. When I use it behind only one of the amps in a dual amp patch, the change in tone is VERY dramatic. I reason that the EQ is not boosting/filtering any frequencies on that channel, only time-shifting the signal very slightly to achieve phase correction with the other channel.

But I want a more precise method to do this. The .01 ms time-shifting increment in Cubase was no help. So I'm wondering if this has something to do with sample rates and what is actually going on with how the signal is being time-shifted. What I'm particularly confused about is how the .01 increments in Cubase don't match up to the length of an individual digital sample. Cubase has to eventually send a 96 kHz digital signal to my interface which converts the digital back to audio. So I assume all the tracks have to fit in the same buckets. So what is a .01 ms shift actually doing?
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  #5  
Old 09-07-2012, 10:08 AM
Will Chen Will Chen is offline
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Quote:
Originally Posted by meambobbo View Post
Right - the comb filter effect. That is what I believe is happening in my Pod between each channel due to different cab/mic selections. When I use an EQ effect to fix this, I am not using the EQ to filter/boost any frequencies. The settings are completely neutral. If I toggle it on/off behind an amp on a single-amp patch, it is impossible to determine if the EQ is on or off - it has no effect on the tone. When I use it behind only one of the amps in a dual amp patch, the change in tone is VERY dramatic. I reason that the EQ is not boosting/filtering any frequencies on that channel, only time-shifting the signal very slightly to achieve phase correction with the other channel.

But I want a more precise method to do this. The .01 ms time-shifting increment in Cubase was no help. So I'm wondering if this has something to do with sample rates and what is actually going on with how the signal is being time-shifted. What I'm particularly confused about is how the .01 increments in Cubase don't match up to the length of an individual digital sample. Cubase has to eventually send a 96 kHz digital signal to my interface which converts the digital back to audio. So I assume all the tracks have to fit in the same buckets. So what is a .01 ms shift actually doing?
Read the thread I quoted, there's some good data in there. I'm not sure of what exactly your trying to do, but a comb filter is most audible when 2 exact signals are played back one of which has been delayed and the signal is summed to mono.
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  #6  
Old 09-07-2012, 10:09 AM
meambobbo meambobbo is offline
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Hey Will - you are helpful as usual. I wasn't even aware of that thread - I thought I was the only one who noticed that!

Now keep in mind, I'm GLAD the Pod HD doesn't offset latency (at least sometimes). If it did, I couldn't achieve the phase correction between different cab/mics I'm getting. I guess if they had ensured all the cab/mics were as in-phase as possible from the get-go, this wouldn't be an issue though.

I guess that additional 0.0025 ms is what prevented me from hearing in Cubase what I can in the Pod.

Now, I kind of question that all the EQ's introduce .0625 ms of delay. I have noticed different EQ's contributing different tonal effects, even with completely neutral settings. I will read the entire thread you posted before jumping to conclusions though.

I noticed in your post you excluded the Mid-Focus EQ. You can make it neutral by setting HP to 0% and LP to 100% and Gain to 0%. You may need to boost Q's to about 57% instead of 50% but that's nitpicking.

Also, the Graphic EQ is not neutral at default settings. It slightly boosts highs.

Anyway, thanks for the feedback. Any idea about how delays can operate outside the sample rate? For instance, at 96 kHZ, .0625 ms is 6 sample sizes. Cubase is also operating at 96 kHZ, but it let's me delay the audio by .0600 ms. How does that work?
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Old 09-07-2012, 10:43 AM
meambobbo meambobbo is offline
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Quote:
Originally Posted by Will Chen View Post
Read the thread I quoted, there's some good data in there. I'm not sure of what exactly your trying to do, but a comb filter is most audible when 2 exact signals are played back one of which has been delayed and the signal is summed to mono.
Let me try to explain a little better...

I find the Pod HD's onboard cab/mics lacking when used individually, but they sound good when I use two different ones that complement each other. I prefer this over using 3rd party IR's or having to introduce additional hardware into my rig. However, some combinations can sound muffled, as some of their frequencies, particularly the high-end, cancel out.

I discovered the EQ thing same as you noticing that even neutral settings make a tonal difference. I concluded this was because a delay from an EQ effect basically synced up the different cab/mics, so there was LESS frequency cancellation.

I theorized that the EQ's are an imprecise method to sync up the signals. I wanted to submit a feature request to Line 6 to more precisely control the delay of any signal, hoping that it would allow me to achieve even better syncing of the two signals and even less frequency cancellation, or to be able to sync up cab/mic combinations that the EQ "trick" was not helpful with.

I thought I could use Cubase's track time-shifting to demonstrate how a variable very precise delay could achieve this. I sought to create a demo clip to add weight to my feature request. While I usually pan both my channels to center in the Pod to get a mono tone, I panned them full left/right and recorded them into separate mono tracks in Cubase, so I could time-shift in Cubase and mix them down to mono there. I could never achieve the success in Cubase I had with using an EQ in the Pod.

So I'm basically trying to figure out if I'm misunderstanding the nature of sample rates and timing in digital signals, or some other reason why I can time-shift with improved results on the Pod but not in Cubase.

Do you by chance know of a plugin or program that would allow me to time-shift a track in Cubase in terms of samples rather than milliseconds? I think that's where the disconnect is.

From my understanding of digital signals, you CAN time-shift a signal by amounts differing from the sample rate, but doing so would necessarily change the values of all your samples. I assume this is what Cubase is doing, and I don't have the precision to time-shift the tracks by 6 sample lengths. Whereas the Pod isn't altering all the sample values - it's just delaying the signal by 6 sample lengths.

And yes, I do understand that a comb filter is most pronounced using identical audio with a slight delay between copies of that, or by recording a singular audio source using 2 different mics at different distances. I understand that since I'm using two different signals (different cab/mics), comb filter may not be the most appropriate term. I also understand that guitar speakers and possibly mics have differing phase inaccuracy and that it would be impossible to prevent some frequency cancellation throughout the entire frequency spectrum. I was simply hoping to achieve less frequency cancellation than I currently get, or what I perceive as a better tone. I need more precise time-shifting tools to do so it seems.
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  #8  
Old 09-07-2012, 11:07 AM
meambobbo meambobbo is offline
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Hey Will, just read through that whole thread - that was smart of anot to use a noise signal and a frequency analyzer to find the cancelled frequencies and compute the precise delay from that. Seems obvious now 8-/

Now that I have a methodology, I am going to use it to compute the exact times of the individual EQ's. I think the assumption that they are all 6 samples, or .0625 ms is incorrect - I get differing tones by trying different ones, even after making them neutral. See my patch on this thread ( http://line6.com/support/thread/87588 ). There is a noticeable difference between using the Mid-Focus vs. Parametric. I'm fairly sure I set up the Mid-Focus to be tone neutral.

And again, keep in mind that the Graphic EQ is not tone neutral at default settings.
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Old 09-07-2012, 12:17 PM
gtrnstuff gtrnstuff is offline
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DP lets you shift a track by as low as 1 sample. I know, not very helpful to a Cubase user. Unless it's there but not documented very well. I thought most DAW's chased each other's feature sets pretty closely.
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Old 09-07-2012, 12:30 PM
meambobbo meambobbo is offline
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thanks for the info - anything is helpful at this point. ultimately what I'm looking for is not to perform this every time by DAW. I want to simply record the potential tonal improvements that would be made available by being able to time-shift as such to pitch to Line 6 my feature request of adding such functionality into the Pod itself.

What's my Vegas odds that I'm wasting my time?
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  #11  
Old 09-07-2012, 01:05 PM
Will Chen Will Chen is offline
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Quote:
Originally Posted by meambobbo View Post
thanks for the info - anything is helpful at this point. ultimately what I'm looking for is not to perform this every time by DAW. I want to simply record the potential tonal improvements that would be made available by being able to time-shift as such to pitch to Line 6 my feature request of adding such functionality into the Pod itself.

What's my Vegas odds that I'm wasting my time?
How do I express infinity as a fraction?

To be brutally honest, if you're having to jump through those kinds of hoops to get what you consider a good tone I think the POD HD is not for you.
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Old 09-07-2012, 01:19 PM
meambobbo meambobbo is offline
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I'm very picky. Many people have said my patches and advice prevented them from returning their units. I think they're good enough, but I always seek to improve.

And although I'm sure I'd prefer any of the higher-end units, I'm on a small budget. I'll be keeping a keen eye open for Digitech's next move though. The GSP1101 is enticing, but I haven't heard anything from it that completely wow'ed me to the point to abandon the HD500, especially after putting so much time into it.
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Old 09-07-2012, 01:29 PM
Will Chen Will Chen is offline
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Originally Posted by meambobbo View Post
I'm very picky. Many people have said my patches and advice prevented them from returning their units. I think they're good enough, but I always seek to improve.

And although I'm sure I'd prefer any of the higher-end units, I'm on a small budget. I'll be keeping a keen eye open for Digitech's next move though. The GSP1101 is enticing, but I haven't heard anything from it that completely wow'ed me to the point to abandon the HD500, especially after putting so much time into it.
I'm not meaning to belittle the tips you've offered up. Certainly they've helped many. And certainly I'm far less picky than you are, given such little time in the day to play, I prefer to simply play...
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  #14  
Old 09-07-2012, 01:41 PM
meambobbo meambobbo is offline
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I definitely didn't take insult, and to be honest you're right. I'm about ready to finalize all my patches, re-record my patch demo to hopefully help people out, and move on. It's funny - every time I go to record clips I realize my chops are crap. So I know I need to stop tweaking and start playing.

But then again, the SLO models are right around the corner, and who knows what else. God help me.
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Old 09-11-2012, 03:05 PM
meambobbo meambobbo is offline
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a slight update - Voxengo Sound Delay is a cubase plugin that allows sample-based time-shifting. If the EQ's on the Pod delay the signal by about 6 samples or .0625 ms, then this provides a more precise method of time-shifting, but is not limited to the .01 ms increments Cubase allows. So between the .01 ms units and the sample-based shifts, I should be able to dial in the ideal tones I want to demonstrate why such an effect would be very valuable for the Pod.

again, i'm probably shouting in the wind, but I find using two cabs breaths so much life into the cab/mic modeling that I can rally strong support behind the effort, especially if given a more intuitive and precise method of syncing the two channels to reduce frequency cancellation.

of course, if L6 were to introduce user-uploadable IR's, this will all become a moot point...

I also intend to test white noise against any effect that can be tone-transparent to determine if it delays the signal and by how much - maybe there's some effects in the unit that take different amount of time to process, and could thereby be used towards better results.
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