Splat 
Max Output Level: 0 dBFS
- Total Posts : 8672
- Joined: 2010/12/29 15:28:29
- Location: Mars.
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/02 22:26:37
(permalink)
I know it's already been said, but the Z3TA is definitely set at high resolution for the 96 and 48 (or 44.1) projects?
Cheers...
Sell by date at 9000 posts. Do not feed. @48/24 & 128 buffers latency is 367 with offset of 38. Sonar Platinum(64 bit),Win 8.1(64 bit),Saffire Pro 40(Firewire),Mix Control = 3.4,Firewire=VIA,Dell Studio XPS 8100(Intel Core i7 CPU 2.93 Ghz/16 Gb),4 x Seagate ST31500341AS (mirrored),GeForce GTX 460,Yamaha DGX-505 keyboard,Roland A-300PRO,Roland SPD-30 V2,FD-8,Triggera Krigg,Shure SM7B,Yamaha HS5.Maschine Studio+Komplete 9 Ultimate+Kontrol Z1.Addictive Keys,Izotope Nectar elements,Overloud Bundle,Geist.Acronis True Image 2014.
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/02 22:59:33
(permalink)
sharke
I was quite surprised at the difference between the two files. The 96kHz clip definitely has way more high frequency content, in fact the 44.1kHz sounded to me like it has some gentle low pass filtering in comparison.
As for Craig's two clips, I definitely prefer the sound of the 44.1kHz clip, it sure sounds a lot fatter and warmer, and the high frequencies in the 96kHz clip are way too harsh sounding.
But again, let me emphasize that comparison is NOT about what "sounds" best. It's about which was able to reproduce higher frequencies with less distortion. I deliberately doubled and transposed what was already a high keyboard part up even higher to make sure there would be plenty of highs. As I said earlier, if I was using that sound in a piece of music, I would have reached for the LPF immediately. The fact that you heard a major difference is the point. You can also trim highs that are there, but you can't create highs that weren't there in the first place. The other differences I heard with the amp sim and virtual drums were far more subtle, but there was a definite difference in the highs. So I did a caricature of the highs to hear what they would sound like. If you listen to them on quality headphones you'll hear the "gauze" in the background of the 44 file from the aliasing which is not present in the 96 one. The question about what's "accurate" for something generated in the box is valid. However, I spent many years designing devices with top octave divider organ chips. While they used digital technology to divide down a high frequency clock, they did not use digital audio technology in the sense of sampling, conversion, etc. and generated analog outputs, albeit via digital means. Because they generated square or pulse waves, I know what it sounds like to have lots of audible high-frequency content - they basically generated harmonics that just kept going and going. That sound is drilled into my brain, and that's the sound I heard from the 96kHz file. I probably should have mentioned I have a genetic predisposition toward good high frequency response (thanks, dad!) - I couldn't go into some stores as a kid because the "ultrasonic" burglar alarms were audible and very painful. Even though I'm 65, I had my hearing tested not too long ago and could still hear 13kHz well, which is unusual. I also didn't watch a lot of TV as a kid because the 15kHz oscillator from cathode ray tubes would drive me nuts. (An interesting aside of the top octave thing was when I was commissioned to create an instrument that could play music around 80-90kHz for dolphin research, involving transducers borrowed from the navy...but that's a whole other subject.)
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/02 23:29:34
(permalink)
CakeAlexS
I know it's already been said, but the Z3TA is definitely set at high resolution for the 96 and 48 (or 44.1) projects?
No, it was set to 1.0 and high resolution, not 2.0 oversampling. I had already determined that I couldn't hear a difference between running instruments/processors when oversampled at 44 or run without oversampling at 96. Native Instruments confirmed that running GR at, say, 88.2 yields the same effective result as doing 2X oversampling. What I'm getting from those who don't like the idea of my running a project at 96kHz is that I wouldn't have to do it if all the elements involved in a 44.1 project included properly designed oversampling. You may find the graphs in this thread revealing. Apparently even many high-end plug-ins have significant, audible aliasing at 44.1/48 and in this thread, one of the commonly suggested remedies is running at a higher sample rate. They don't talk about amp sims or virtual instruments, just compressors, EQs, etc., but I would think the principles are the same. So if those graphs are to be believed, it's no wonder I think projects sound better run at a higher sample rate. Until everything is designed to be totally wonderful, I'm not going to avoid doing something that gives better sound quality. But frankly even if the sound thing wasn't an issue, I enjoy the lower latency. And with Apple insisting on getting 24/96 masters going forward, it looks like we don't have much choice anyway.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 00:40:07
(permalink)
AndertonWhat I'm getting from those who don't like the idea of my running a project at 96kHz is that I wouldn't have to do it if all the elements involved in a 44.1 project included properly designed oversampling.
Correct. If one designs a plugin that creates distortion, then they should account for that. But that doesn't mean that everything one might wish to use is properly designed, and if running at 96kHz makes them better then it is what it is.
You may find the graphs in this thread revealing. Apparently even many high-end plug-ins have significant, audible aliasing at 44.1/48 and in this thread, one of the commonly suggested remedies is running at a higher sample rate. They don't talk about amp sims or virtual instruments, just compressors, EQs, etc., but I would think the principles are the same.
What do you consider "significant, audible aliasing" regarding those charts? Any specific examples there? In many of the ones I looked at, the aliasing was often close to 100dB (or more) below the signal. Compressors, technically speaking, do distort the signal and one needs to be especially careful when aggressive settings with very fast attack/release times are used. Less aggressive settings should create no problems. EQ's should add no audible distortion unless it's put there by design in addition to the EQing itself, as in saturation in an analog emulation. You can get somewhat different EQ curves at high frequencies at different sampling rates (if it's not compensated for), but in that case which curve is "better" is generally a subjective assessment (as with any other EQ decision).
 In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
mudgel
Max Output Level: 0 dBFS
- Total Posts : 12010
- Joined: 2004/08/13 00:56:05
- Location: Linton Victoria (Near Ballarat)
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 02:13:30
(permalink)
I'm certainly no rocket scientist when it comes to this particular discussion nor am I just a rock so I understand a reasonable amount of the discussion. BUT I'd like to congratulate all involved for the manner in which they've presented their points. Even when there's been some points of contention in what in other forums would start a war all has remained civil.
It's another example of the quality of the people that are part of the Sonar forum family.
There was a time pre X3 where there was a tension evident in the forum but generally speaking the forum is pretty much a pleasure to be part of. I think it was Craig who mentioned (a while back) a new gestalt on this forum and this thread is a classic example as is the one about your favourite underrated feature. Group hug. :-)
Mike V. (MUDGEL) STUDIO: Win 10 Pro x64, SPlat & CbB x64, PC: ASUS Z370-A, INTEL i7 8700k, 32GIG DDR4 2400, OC 4.7Ghz. Storage: 7 TB SATA III, 750GiG SSD & Samsung 500 Gig 960 EVO NVMe M.2. Monitors: Adam A7X, JBL 10” Sub. Audio I/O & DSP Server: DIGIGRID IOS & IOX. Screen: Raven MTi + 43" HD 4K TV Monitor. Keyboard Controller: Native Instruments Komplete Kontrol S88.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 02:43:12
(permalink)
Some might find this useful, from Blue Cat's Dynamics user manual (regarding their built in oversampling): http://www.bluecataudio.com/Doc/Product_Dynamics/
Oversampling
You can use Oversampling to reduce the aliasing artifacts that can be produced by the non-linearities of the dynamics processor. It can be particularly useful for audio content with higher frequencies, or if you use large compression ratios with short attack and release times. It is applicable if you work with lower sample rates (such as 44.1 or 48 kHz). With higher sampling rates you usually do not need to work with an oversampled signal.
Beware that each oversampling stage consumes a lot of CPU (more than double CPU usage). You should use 4x oversampling with the stereo version for mastering purposes only, or with very low attack/release times in Peak mode. This is typically a feature you don't want to activate on every single track of a project.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
BJN
Max Output Level: -86 dBFS
- Total Posts : 222
- Joined: 2013/10/09 07:52:48
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 05:41:09
(permalink)
Wow what a discussion. Toing and froing.
I have seen discussions and even arguments on this subject before.
I too am glad to see a civil discourse about it.
I have come to three possibilities why higher frequencies sound better.
The Science says we should not hear an improvement at higher sampling rates.
Yet we do.
One theory is that broader harmonic content at higher rates enhance the fundamental
information and thus discernable to the ear.
An other is that it proves the existence of the personality as the soul or spirit inhabiting the body and it is the spirit perceiving the higher frequency content. In other words there is more to the perception of the sound than just the mechanical ear and nerves processes of the body.
Lastly, it is a well kept secret by the high priests of the zen of audio engineering where science is used to disabuse anyone from even trying the idea for themselves. Yet even MAstering Engineers up sample and process from there.
There are three persuasions for you,
The first might be the acceptable one. LOL
------------------------------------------------------- Magic: when you feel inspired to create which in turn inspires more creation. And the corollary: if magic happens inspiration might flog it to death with numerous retakes. Bart Nettle
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 10:10:07
(permalink)
drewfx1
The sample time does not remotely equal the time resolution. This is a common misunderstanding of how sampling works.
It depends on bit depth as well as sample rate, but the short answer is 48kHz has a timing resolution of FAR greater than 1/48,000.
Can you explain this? When I zoom in to the sample level, I see a straight line that last x number of microseconds. I understand that gets smoothed when it's reconstructed, but how can data shorter than one sample be encoded into a straight line? What Moorer is saying is that if you have two events 10 microseconds apart, those events cannot be encoded in something that cannot resolve fewer than 20 microseconds. For a film analogy, if the frame rate is 30 frames per second and you have two different, sequential visual events occurring during the time that one frame occurs, how can those two events play back? I don't see how they could be encoded in a single frame as two separate events.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 12:46:11
(permalink)
Anderton
drewfx1
The sample time does not remotely equal the time resolution. This is a common misunderstanding of how sampling works.
It depends on bit depth as well as sample rate, but the short answer is 48kHz has a timing resolution of FAR greater than 1/48,000.
Can you explain this? When I zoom in to the sample level, I see a straight line that last x number of microseconds. I understand that gets smoothed when it's reconstructed, but how can data shorter than one sample be encoded into a straight line? What Moorer is saying is that if you have two events 10 microseconds apart, those events cannot be encoded in something that cannot resolve fewer than 20 microseconds.
For a film analogy, if the frame rate is 30 frames per second and you have two different, sequential visual events occurring during the time that one frame occurs, how can those two events play back? I don't see how they could be encoded in a single frame as two separate events.
Rule #1: Never use analogies when trying to understand sampling - they're almost always wrong (in whole or in part) because sampling just isn't intuitive and it doesn't really work the same way other stuff does. Rule #2: Never argue the analogy, as it inevitably just takes things OT.  Be careful when zooming in - many (most?) DAWs just show a picture that "connects the dots", which is extremely misleading as this has little to do with what a reconstructed signal looks like (or how a sampled signal is reconstructed). It gives you a reasonable picture at low frequencies, but a high frequency sine wave looks nothing like a sine wave - even though your DAC outputs a nice looking sine wave. The short answer is what happens is if you move your signal a fraction of a sample in time (at a reasonable bit depth), then the sample values will change. Consider a 12kHz sine wave sampled at 48kHz: 1. Since 12 kHz is exactly 1/4th the sample rate, you get exactly 4 samples per cycle. 2. This means that successive samples are exactly 90° apart. 3. Let's say we take our first sample s1 at 0°. That means s2 is at 90°, s3 is at 180°, s4 is at 270° and so on. 4. Now let's move our signal back .5 samples in time. 5. Now s1 is at 45°, s2 at 135°, s3 at 225°, s4 at 315°, and so on. Hopefully it's obvious that the sample values are not going to be the same when we are sampling the sine wave at different phases in its cycle. So the question becomes, how far can you move the waveform on the x-axis (time) without having (almost) any of the y-axis (sample amplitude) values change? Again, hopefully it's obvious that with higher resolution on the y-axis (increased bit depth), the amount you can move your signal in time without changing the sample values changes gets smaller. An experiment to try: 1. Make a stereo wave form at 44.1/48kHz where every sample in L vs. R are absolutely identical (i.e. L=R). 2. Upsample by 2x (or higher). 3. Shift L by 1 sample in time at the higher rate. 4. Downsample back to the original SR. 5. Zoom all the way in and compare L and R. You will find that not every sample in L and R are the same anymore - because you time shifted L by a fraction of a sample.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
bitflipper
01100010 01101001 01110100 01100110 01101100 01101
- Total Posts : 26036
- Joined: 2006/09/17 11:23:23
- Location: Everett, WA USA
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 13:10:44
(permalink)
...if you have two events 10 microseconds apart, those events cannot be encoded in something that cannot resolve fewer than 20 microseconds. Keep in mind that anything that happens entirely inside a 10-microsecond timeframe is much too fast to worry about. You only worry about those frequencies after you've bought your $2,000 oxygen-free polarized cables.
 All else is in doubt, so this is the truth I cling to. My Stuff
|
abb
Max Output Level: -88 dBFS
- Total Posts : 118
- Joined: 2004/01/19 02:04:35
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 13:53:47
(permalink)
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 14:39:48
(permalink)
drewfx1
An experiment to try:
1. Make a stereo wave form at 44.1/48kHz where every sample in L vs. R are absolutely identical (i.e. L=R).
2. Upsample by 2x (or higher).
3. Shift L by 1 sample in time at the higher rate.
4. Downsample back to the original SR.
5. Zoom all the way in and compare L and R.
I understand how reconstruction and smoothing works. The question is about capture. How can something with a duration of 20 microseconds encode two events that are 10 microseconds apart?
|
John
Forum Host
- Total Posts : 30467
- Joined: 2003/11/06 11:53:17
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 14:47:59
(permalink)
If I have this right the above from Craig is the argument about resolution or how many slices are being made at any given time.
From what I understand is it has no impact.
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 15:00:14
(permalink)
bitflipper
Keep in mind that anything that happens entirely inside a 10-microsecond timeframe is much too fast to worry about. You only worry about those frequencies after you've bought your $2,000 oxygen-free polarized cables.
This has nothing to do with frequency. Moorer's paper was about binaural hearing and the perception of delays between events, not listening to continuous frequencies. I would guess this is a refinement of the precedence effect "the first wavefront law"). He maintains people with average acuity can recognize a time differential between impulses hitting each ear of as little as 15 microseconds, and some could discriminate down to 5-8 microseconds. I don't find that hard to believe. If I nudge a waveform one sample at a time compared to same waveform out of phase, it's clear the jump between samples is quite large. If you then switch back out of phase, the earlier one does "weight" toward one side of your hearing, as predicted by the precedence effect. Certainly 15 microseconds meets the requirement of being below the listener's echo threshold. I didn't do the research, I'm just referencing his. I certainly don't believe we know everything there is to know about hearing and the subsequent processing of that information by the brain. Just remember how freaked out people were when they realized we see things upside down, and the brain does the needed corrections so we see images right side up.
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 15:26:56
(permalink)
John
If I have this right the above from Craig is the argument about resolution or how many slices are being made at any given time.
From what I understand is it has no impact.
That's where the controversy lies. Of course reconstruction will reconstruct a waveform; no question about that, otherwise the outputs of digital audio systems would be stair-stepped instead of continuous. This isn't about reconstructing a waveform, but about reconstructing a characteristic of the binaural listening experience which is, after all, how we hear sound. The question is whether reconstruction is sufficiently precise to reconstruct the timing difference between two signals that are, say, 8 microseconds apart. I don't see how that's possible if the capture medium can't resolve differential timings under 21 microseconds. Let me explain what I think is going on. A 48kHz sample clock samples an incoming voltage, which is at "x" volts. So far, so good. 5 microseconds later, "y" volts is present at the input. 8 microseconds after that, "z" volts is present at the input. 5 microseconds later, "w" voltage is present at the input and that voltage lasts for 10 microseconds. When the next sample occurs 21 microseconds after the first one, it will read the "w" voltage, but it will ignore the "y" and "z" values because they occurred between samples. I don't see any way the "y' and "z" values could factor into the encoding process because the system never sees them. So then the question becomes does reconstruction reproduce those "ignored" variations successfully, and if not, does it matter? The argument that says it doesn't matter maintains that smoothing will accurately fill in the values between the "x" and "w" voltages, and will therefore reconstruct the frequency that was present at those times. However, my understanding of Moorer's argument is that if there were spatial cues in between "x" and "w," they will be lost. Whether that matters or not depends on whether you accept Moorer's contention that people can discriminate between extremely short time delays when signals hit both ears. As I doubt anyone in this thread has verified or disproven these experiments, I don't think it's possible to accept or dismiss them out of hand. However, if (I emphasize "if," although I don't know what his motivation would be for making things up) he is correct, then given the sampling scenario presented above, I simply don't see any way that a 21-microsecond window can encode incoming delay-based events whose duration is significantly less than that.
|
Anderton
Max Output Level: 0 dBFS
- Total Posts : 14070
- Joined: 2003/11/06 14:02:03
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 15:33:37
(permalink)
abb
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
I don't think he's talking about time resolution, I think he's talking about amplitude resolution after 24-bit D/A conversion. Whether he's considering 24 bits as the actual value or taking into account circuit board layout issues, noise, laser trimming tolerances, etc. I don't know but I don't see how it relates to the question I'm asking.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 16:31:01
(permalink)
Anderton
drewfx1
An experiment to try:
1. Make a stereo wave form at 44.1/48kHz where every sample in L vs. R are absolutely identical (i.e. L=R).
2. Upsample by 2x (or higher).
3. Shift L by 1 sample in time at the higher rate.
4. Downsample back to the original SR.
5. Zoom all the way in and compare L and R.
I understand how reconstruction and smoothing works. The question is about capture. How can something with a duration of 20 microseconds encode two events that are 10 microseconds apart?
Consider again the sampling of a pure sine wave example - not just the frequency and amplitude of the sine wave is captured by sampling, but the phase as well. Your 10 microseconds in the time domain equates to phase shift in the frequency domain. So take the 12kHz sine wave example again: ~10 microseconds = 45 degrees So in this case it just means you are sampling a sine wave with 45 degrees of phase shift (~10 microseconds) and the sampled values will be different because the samples are taken at different points in the cycle. So if you sample two sine waves 45 degrees out of phase with each other they will be reconstructed as 45 degrees out of phase with each other. And you know that Fourier says that any complex waveform is just a combination of sine waves at various frequencies, amplitudes and phases.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
abb
Max Output Level: -88 dBFS
- Total Posts : 118
- Joined: 2004/01/19 02:04:35
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 16:34:26
(permalink)
Anderton
abb
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
I don't think he's talking about time resolution, I think he's talking about amplitude resolution after 24-bit D/A conversion. Whether he's considering 24 bits as the actual value or taking into account circuit board layout issues, noise, laser trimming tolerances, etc. I don't know but I don't see how it relates to the question I'm asking.
I'm similarly confused as to the relevance of the number; hence my question.
|
John
Forum Host
- Total Posts : 30467
- Joined: 2003/11/06 11:53:17
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 16:39:22
(permalink)
Anderton
John
If I have this right the above from Craig is the argument about resolution or how many slices are being made at any given time.
From what I understand is it has no impact.
That's where the controversy lies. Of course reconstruction will reconstruct a waveform; no question about that, otherwise the outputs of digital audio systems would be stair-stepped instead of continuous. This isn't about reconstructing a waveform, but about reconstructing a characteristic of the binaural listening experience which is, after all, how we hear sound.
The question is whether reconstruction is sufficiently precise to reconstruct the timing difference between two signals that are, say, 8 microseconds apart. I don't see how that's possible if the capture medium can't resolve differential timings under 21 microseconds.
Let me explain what I think is going on.
A 48kHz sample clock samples an incoming voltage, which is at "x" volts. So far, so good. 5 microseconds later, "y" volts is present at the input. 8 microseconds after that, "z" volts is present at the input. 5 microseconds later, "w" voltage is present at the input and that voltage lasts for 10 microseconds.
When the next sample occurs 21 microseconds after the first one, it will read the "w" voltage, but it will ignore the "y" and "z" values because they occurred between samples. I don't see any way the "y' and "z" values could factor into the encoding process because the system never sees them.
So then the question becomes does reconstruction reproduce those "ignored" variations successfully, and if not, does it matter? The argument that says it doesn't matter maintains that smoothing will accurately fill in the values between the "x" and "w" voltages, and will therefore reconstruct the frequency that was present at those times.
However, my understanding of Moorer's argument is that if there were spatial cues in between "x" and "w," they will be lost. Whether that matters or not depends on whether you accept Moorer's contention that people can discriminate between extremely short time delays when signals hit both ears. As I doubt anyone in this thread has verified or disproven these experiments, I don't think it's possible to accept or dismiss them out of hand. However, if (I emphasize "if," although I don't know what his motivation would be for making things up) he is correct, then given the sampling scenario presented above, I simply don't see any way that a 21-microsecond window can encode incoming delay-based events whose duration is significantly less than that.
I see what you are getting at. The problem is when you introduce binaural into it. Spacial data is not held in one mono audio stream but with stereo pairs. This is the way I understand it. You have two streams of data that interact with one another to give a sense of space. Mono doesn't do this. There is nothing in the mono signal to give a space sense. BTW Binaural is not the same as stereo. From my experimenting many years ago it is a technique for creating the same aural experience as being there. You use two mics place about as far apart as ones head with a baffle between representing the ears. You need headphones to listen to the resulting recording. On loud speakers it just sounds like mono. You may not have meant that and only were referring to our two ears.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 16:39:54
(permalink)
AndertonLet me explain what I think is going on.
A 48kHz sample clock samples an incoming voltage, which is at "x" volts. So far, so good. 5 microseconds later, "y" volts is present at the input. 8 microseconds after that, "z" volts is present at the input. 5 microseconds later, "w" voltage is present at the input and that voltage lasts for 10 microseconds.
When the next sample occurs 21 microseconds after the first one, it will read the "w" voltage, but it will ignore the "y" and "z" values because they occurred between samples. I don't see any way the "y' and "z" values could factor into the encoding process because the system never sees them.
So then the question becomes does reconstruction reproduce those "ignored" variations successfully, and if not, does it matter? The argument that says it doesn't matter maintains that smoothing will accurately fill in the values between the "x" and "w" voltages, and will therefore reconstruct the frequency that was present at those times.
What the sampling theorem says is that for a signal band limited to one half the sampling frequency, theoretically* everything is captured. This includes everything at any point in time between the samples for a properly band limited signal. So "y" and "z" are indeed stored in the sampled signal - that's how come it can be reconstructed. *In practice it's isn't perfect, but neither is human hearing.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 16:46:18
(permalink)
abb
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
1/(2 * pi * quantization levels * sampling frequency) where quantization levels = 2^(bit depth) This is a theoretical limit. In reality, it's not going to be that good.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 16:48:52
(permalink)
Anderton I don't know but I don't see how it relates to the question I'm asking.
It means that for a properly band limited signal, sampling preserves timing information to a tiny fraction of the sampling period.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
Splat
Max Output Level: 0 dBFS
- Total Posts : 8672
- Joined: 2010/12/29 15:28:29
- Location: Mars.
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 17:01:40
(permalink)
Bring on the quantum sampler...
Sell by date at 9000 posts. Do not feed. @48/24 & 128 buffers latency is 367 with offset of 38. Sonar Platinum(64 bit),Win 8.1(64 bit),Saffire Pro 40(Firewire),Mix Control = 3.4,Firewire=VIA,Dell Studio XPS 8100(Intel Core i7 CPU 2.93 Ghz/16 Gb),4 x Seagate ST31500341AS (mirrored),GeForce GTX 460,Yamaha DGX-505 keyboard,Roland A-300PRO,Roland SPD-30 V2,FD-8,Triggera Krigg,Shure SM7B,Yamaha HS5.Maschine Studio+Komplete 9 Ultimate+Kontrol Z1.Addictive Keys,Izotope Nectar elements,Overloud Bundle,Geist.Acronis True Image 2014.
|
abb
Max Output Level: -88 dBFS
- Total Posts : 118
- Joined: 2004/01/19 02:04:35
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 17:01:48
(permalink)
Anderton
John
If I have this right the above from Craig is the argument about resolution or how many slices are being made at any given time.
From what I understand is it has no impact.
That's where the controversy lies. Of course reconstruction will reconstruct a waveform; no question about that, otherwise the outputs of digital audio systems would be stair-stepped instead of continuous. This isn't about reconstructing a waveform, but about reconstructing a characteristic of the binaural listening experience which is, after all, how we hear sound.
The question is whether reconstruction is sufficiently precise to reconstruct the timing difference between two signals that are, say, 8 microseconds apart. I don't see how that's possible if the capture medium can't resolve differential timings under 21 microseconds.
Let me explain what I think is going on.
A 48kHz sample clock samples an incoming voltage, which is at "x" volts. So far, so good. 5 microseconds later, "y" volts is present at the input. 8 microseconds after that, "z" volts is present at the input. 5 microseconds later, "w" voltage is present at the input and that voltage lasts for 10 microseconds.
When the next sample occurs 21 microseconds after the first one, it will read the "w" voltage, but it will ignore the "y" and "z" values because they occurred between samples. I don't see any way the "y' and "z" values could factor into the encoding process because the system never sees them.
Hi Craig, During the digitization process, signal voltage fluctuations on timescales shorter than the sample period can influence the encoded waveform by virtue of techniques like delta-sigma modulation. This involves an initial, coarse sampling at extremely high sample rates (relative to the range of human hearing) followed by integration (which smears the sampled voltages over time). Fig. 4 on this webpage http://www.maximintegrated.com/en/app-notes/index.mvp/id/1870 shows a block diagram of a typical sigma-delta modulator circuit. Apparently delta-sigma modulation is quite prevalent these days, so perhaps that explains how "...the "y" and "z" values could factor into the encoding process..." Cheers...
|
abb
Max Output Level: -88 dBFS
- Total Posts : 118
- Joined: 2004/01/19 02:04:35
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 17:09:44
(permalink)
drewfx1
abb
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
1/(2 * pi * quantization levels * sampling frequency)
where quantization levels = 2^(bit depth)
This is a theoretical limit. In reality, it's not going to be that good.
Hmm, it seems that you're conflating amplitude resolution (bit depth) and temporal resolution (sampling rate). I don't see how the two interact to increase temporal resolution.
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 17:24:18
(permalink)
abb
drewfx1
abb
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
1/(2 * pi * quantization levels * sampling frequency)
where quantization levels = 2^(bit depth)
This is a theoretical limit. In reality, it's not going to be that good.
Hmm, it seems that you're conflating amplitude resolution (bit depth) and temporal resolution (sampling rate). I don't see how the two interact to increase temporal resolution.
No. It's as I said - if you shift your waveform to be sampled in time by a fraction of a sample period before sampling, you will get different sample values than you otherwise would. Is it not clear that "different sample values" implies "different signal"? What's the difference between the signals? One has been shifted in time and nothing else. At higher bit depths there are more possible sample values, meaning the amount you can shift your signal in time without getting different sample values (different sampled signal) is smaller. Hence greater timing resolution at greater bit depths.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|
The Maillard Reaction
Max Output Level: 0 dBFS
- Total Posts : 31918
- Joined: 2004/07/09 20:02:20
- Status: offline
.
post edited by Bash von Gitfiddle - 2018/10/04 22:44:02
|
abb
Max Output Level: -88 dBFS
- Total Posts : 118
- Joined: 2004/01/19 02:04:35
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 17:57:58
(permalink)
drewfx1
abb
drewfx1
abb
drewfx1
gswitzDrewFX, what would the time resolution of 24 bit 48kHz be? Can you figure it out? I'm curious.
Something on the order of 0.0000000000002 seconds.
How did you compute this value?
1/(2 * pi * quantization levels * sampling frequency)
where quantization levels = 2^(bit depth)
This is a theoretical limit. In reality, it's not going to be that good.
Hmm, it seems that you're conflating amplitude resolution (bit depth) and temporal resolution (sampling rate). I don't see how the two interact to increase temporal resolution.
No.
It's as I said - if you shift your waveform to be sampled in time by a fraction of a sample period before sampling, you will get different sample values than you otherwise would.
Is it not clear that "different sample values" implies "different signal"?
What's the difference between the signals? One has been shifted in time and nothing else.
At higher bit depths there are more possible sample values, meaning the amount you can shift your signal in time without getting different sample values (different sampled signal) is smaller. Hence greater timing resolution at greater bit depths.
It's easy to visualize what you're depicting. I also fully understand what you're saying. However what's not clear is how your argument relates to increased temporal resolution from a mechanistic perspective. Are you saying that part of the digitization process involves shifting signals around in time? If not I just don't understand how having more amplitude resolution will increase your temporal resolution. BTW, did you see my post above (#84)? That's what I mean by 'mechanistic.' Cheers...
|
sven450
Max Output Level: -72 dBFS
- Total Posts : 945
- Joined: 2004/03/16 08:11:49
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 18:48:21
(permalink)
Damn. This s*** is for real. I feel smarter having just having come across this! Thx!
|
drewfx1
Max Output Level: -9.5 dBFS
- Total Posts : 6585
- Joined: 2008/08/04 16:19:11
- Status: offline
Re: Remember that 96K TH2 thread? I Just had my mind blown, big-time
2014/06/03 18:54:37
(permalink)
abbIt's easy to visualize what you're depicting. I also fully understand what you're saying. However what's not clear is how your argument relates to increased temporal resolution from a mechanistic perspective. Are you saying that part of the digitization process involves shifting signals around in time? If not I just don't understand how having more amplitude resolution will increase your temporal resolution.
Maybe this will help. Let's express a few samples of a 10kHz sine wave sampled at 48kHz at different bit depths (they start at 35 degrees): 24 16 8 4 bits4811507 18794 73 47882713 30791 120 7-731116 -2856 -12 -1-8261167 -32271 -127 -8-3545179 -13849 -55 -4 Now let's sample it .001 samples later in time (which is ~.02 microseconds):24 16 8 4 bits
4820498 18830 73 4
7878950 30777 120 7
-742054 -2899 -12 -1
-8263066 -32278 -127 -8
-3535225 -13810 -54 -4 In this short sequence at 4 bits the samples are all identical, at 8 bits only the last is different, whereas at 16 and 24 every sample is quite different. So the lower bit resolution limits our ability to capture a small time shift. Note that if I bothered to do a much longer string of samples we would eventually see some changes even at 4 bit with this amount of time shift.
In order, then, to discover the limit of deepest tones, it is necessary not only to produce very violent agitations in the air but to give these the form of simple pendular vibrations. - Hermann von Helmholtz, predicting the role of the electric bassist in 1877.
|