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  • xnor
  • [*][*][*][*][*]
  • Developer
Audibility of phase shifts and time delays
Reply #25
Haven't we also not seen this Yamaha page before: temporal_resolution?

They say not 10us but 6us.
  • Last Edit: 22 April, 2015, 11:20:18 AM by xnor
"I hear it when I see it."

Audibility of phase shifts and time delays
Reply #26
Haven't we also not seen this Yamaha page before: temporal_resolution?

They say not 10us but 6us.



I wouldn't haggle over 10 uSec as opposed to 5 uSec as being the threshold. However, its impact seems to have been considerably inflated:

"To also accurately reproduce changes in a signal’s frequency spectrum with a temporal resolution down to 6 microseconds, the sampling rate of a digital audio system must operate at a minimum of the reciprocal of 6 microseconds = 166 kHz. Figure 515 presents the sampling of an audio signal that starts at t = 0, and reaches a detectable level at t = 6 microseconds. To capture the onset of the waveform, the sample time must be at least 6 microseconds."

So they are basically saying that 44 KHz can't accurately reproduce audible changes in a signal, and 96 Khz can't accurately reproduce audibly changes in a signal, you have to go to almost twice that, or 166 KHz!  They are basically on the Meridian bandwagon of 192 KHz being the historically-ordained sample rate that is required for sonically accurate reproduction.

Going back in history, Clark's paper that put audibility @ 40 uSec alluded to the fact that he didn't think it was the time delays as such that caused the audible differences, but the changes in frequency response that they necessarily caused. He said this to me in so many words at the time. Later on I did some ABX tests that were positive for audible differences due to the 1/2 sample inter-channel delay in the converters in the CDP 101 combined with a system that had a center channel speaker that received L+R out of an ordinary analog summer.  Same basic explanation.

I think that we are getting treated to the typical placebophile moving goal posts fueled by conflating experimental results with actual operational circumstances that are different.


  • 2Bdecided
  • [*][*][*][*][*]
  • Developer
Audibility of phase shifts and time delays
Reply #27
Later on I did some ABX tests that were positive for audible differences due to the 1/2 sample inter-channel delay in the converters in the CDP 101 combined with a system that had a center channel speaker that received L+R out of an ordinary analog summer.  Same basic explanation.
I make that 1dB down at 13.1kHz, 2dB down at 18.4kHz, etc. I'd struggle to detect that these days. With a whole sample delay, you can halve those frequencies (making it easily audible), and get a full 6dB attenuation by 14.7kHz.

But as everyone says, in a proper 44.1kHz 16-bit system there are no such problems.

Cheers,
David.

Audibility of phase shifts and time delays
Reply #28
Later on I did some ABX tests that were positive for audible differences due to the 1/2 sample inter-channel delay in the converters in the CDP 101 combined with a system that had a center channel speaker that received L+R out of an ordinary analog summer.  Same basic explanation.


I make that 1dB down at 13.1kHz, 2dB down at 18.4kHz, etc. I'd struggle to detect that these days.


I would too. However this was in the late 1980s. The kid was on his game!

Quote
With a whole sample delay, you can halve those frequencies (making it easily audible), and get a full 6dB attenuation by 14.7kHz.


Flip the argument around.  1 sample is 22 uSec,  1/2 sample is 11 uSec, and  1/4 sample is 5.5 uSec.  And there we are in the 5 uSec range.

I don't know if I could have heard the 1/4 sample delay back in the day. I never tried, and I didn't  know how to try as hard then as I learned later.

This summation of delayed and non-delayed signals can happen in wire or it can happen, perhaps with less coherence in the air.

Quote
But as everyone says, in a proper 44.1kHz 16-bit system there are no such problems.


Yes, we have to say that for the placebophiles in attendance. ;-)

Fact of the matter is that even with the 1/2 sample interchannel delay uncorrected, the CDP 101 was challenging to ABX with positive outcome for differences with music and stereo speakers.  Those who ranted and raved about its horrid sound quality were always placebophiles, well-primed by articles from the Golden Eared press.  That hasn't changed.

The point is that the potential audible effects related to 5 uSec delays can be completely explained by their frequency response effects. Then we don't have to throw away what the classic books about psychoacoustics say about the audibility of phase and delay.
  • Last Edit: 23 April, 2015, 09:59:19 AM by Arnold B. Krueger

  • mzil
  • [*][*][*][*][*]
Audibility of phase shifts and time delays
Reply #29
I agree "temporal resolution/smearing" is mumbo jumbo and Bob Stuart woo. What's possibly audible is simply the frequency response alteration due to constructive/destructive interference of the two waveforms. I notice exactly that here in a minor change to the sound between the .5 millisecond spaced clicks and the 1 millisecond spaced clicks:

https://www.youtube.com/watch?v=3zZRy-UArXM

Granted this Youtube test is in milliseconds, not microseconds, but the principal still holds true: the audible detection at this 500 microseconds difference level is due to frequency response, not "timing". I'm new to Audacity, so it would take me forever to construct, but if one of you Audacity veterans would like to make a click track with much shorter than 500 microsecond spacings, those of us with [slightly] younger ears would be more than glad to take the test.
  • Last Edit: 23 April, 2015, 02:30:50 PM by mzil

Audibility of phase shifts and time delays
Reply #30
I agree "temporal resolution/smearing" is mumbo jumbo and Bob Stuart woo. What's possibly audible is simply the frequency response alteration due to constructive/destructive interference of the two waveforms. I notice exactly that here in a minor change to the sound between the .5 millisecond spaced clicks and the 1 millisecond spaced clicks:

https://www.youtube.com/watch?v=3zZRy-UArXM

Granted this Youtube test is in milliseconds, not microseconds, but the principal still holds true: the audible detection at this 500 microseconds difference level is due to frequency response, not "timing". I'm new to Audacity, so it would take me forever to construct, but if one of you Audacity veterans would like to make a click track with much shorter than 500 microsecond spacings, those of us with [slightly] younger ears would be more than glad to take the test.


Here's what Kunchur says about past work trying to determine this experimentally:\

"
In experiments probing temporal resolution, a pair of
stimuli are presented that differ in their temporal structure.
As the temporal difference is progressively reduced,
one finds the threshold for barely being able to discern
a difference. In one experiment by Leshowitz (1971), listeners
were presented with a single pulse or two narrower
pulses (with the same total energy) separated by an interval
?t. The click and click-pair could be distinguished
down to ?t ? 10 ?s.

In this case, the two stimuli have differences
in their amplitude spectra and their discernment
was explained on this basis. Isospectral variants of this experiment
were carried out by Ronken (1970) and later by
Henning and Gaskell (1981) where one stimulus consisted
of a short pulse followed by a taller one separated by an
interval ?t. The second stimulus was a similar pair with
the time order reversed and hence had the same amplitude
spectrum. The shortest ?t for which these stimuli could be
distinguished was about 200 ?s.

Another type of constantamplitude-
spectrum experiment involves the detection of
gaps in noise (Plomp, 1964; Penner, 1977; Eddins et al.,
1992). In these the threshold for gap detection was of the
order of 2 ms. The issue of determining temporal resolution
while avoiding spectral cues was recently tackled (Yost
et al., 1996; Patterson and Datta, 1996; Krumbholz et al.,
2003) through the use of iterated rippled noise.

The experiment of Krumbholz et al. (2003) showed that differences in
delay between a masker and signal could be discerned down
to 12.5 ?s. In their work a masking paradigm was used to
argue that spectral cues did not play a role in the discernment.
Note that in all the previous cited experiments, the
threshold ?t exceeded the nominal 9 ?s.
"

His methodology was as follows:

"The experiment consists of presenting an approximately
square-wave shaped complex tone, with a 7 kHz fundamental,
through earphones with different degrees of low-pass
filtering (i.e., with different time constants ? ) and testing
a listener’s ability to distinguish this filtered tone from the
unfiltered control tone (?=0).
A. Apparatus
A significant potential bottleneck in a temporal resolution
experiment is the temporal-response speed of
the equipment. Typically the apparatus consists of signal
sources, a switching/gating method used to ramp the signals,
an amplifier for driving the tranducer, and the transducer
itself. In the present work, many different approaches
were initally tried and abandoned, including using digital
synthesis (with 24-bit/96-kHz sampling) for the production
and ramping of signals. It was found that such a digital
method had far too inadequate temporal definition for this
purpose. So instead an analog signal generator (model 4001
manufactured by Global Specialties Instruments, Cheshire,
Connecticutt) was used to produce a 7 kHz square waveform
that had 20 ns rise/fall times (a thousand times faster
than the 23 ?s rise/fall times that characterize the 44.1 kHz
sampling rate of the digital compact-disk).

The electronics used in this experiment was designed
and built in-house because the required combination of response
speed, linearity, power-supply stability, and damping
ability (output impedance) was not found in commercial
headphone-amplifiers. The result was an amplifier with
input and output impedances of 1 M? and 50 m? respectively,
a 3-dB power bandwidth of 0–2.2 MHz, a rise/fall
time of 90 ns, and dc offset voltages under 0.6 mV at all
stages. The linearity of the entire signal chain is essentially
perfect (non-linear errors have sound levels of less than 0
dB SPL; please see below).
"

"
The earphones used were a pair of Grado RS1 (Grado
Laboratories, Brooklyn, New York) supra-aural headphones
which have a frequency response of 12 Hz–30 kHz,
an input resistance of 32 ?, and an efficiency of 98 dB/mW.
Identical signals are fed to both left and right ears to provide
a diotic presentation.
"

"In the main experiment, subjects try to discern differences
between a (7 kHz approximately square-wave shaped)
signal with finite low-pass filtering versus a control signal
with no filtering (waveforms depicted in Fig. 3). The control
tone was perceived to have a sharper or brighter timbre
whereas the filtered one had a duller quality (no difference
in loudness was perceived except for the largest setting of
?=30 ?s). In the blind test, the subject tries to judge
whether an unknown sound is the control or filtered tone
for different settings of ? . It was found in preliminary testing
(especially when ? is close to the threshold) that subjects
needed to listen to the tones for several seconds to
form a lasting impression of the sounds; immediately after
switching the subjects had difficulty assessing whether
anything had changed or not. This again confirms that the
gating itself does not provide a cue.
"

  • Woodinville
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Audibility of phase shifts and time delays
Reply #31
Lets not talk about Kuncher, please.

Instead, generate an FIR highpass filter about 64 samples long that rolls off at 2K, so there is effectively no energy below that.

Calculate the roots. Take all of the roots > 1 and invert them (same frequency response, but minimum phase).

Turn the roots back into a filter.

Now you have a minimum-phase click with no energy below 2kHz.

repeat that click every 10 milliseconds. Make a pulse train. Still no content under 2kHz, note...

Now, delay that by 1 sample at 48kHz in one channel vs. the other. Listen to that vs. the two at the same time, i.e. no delay. Use both speakers and headphones.

See what YOU hear.
-----
J. D. (jj) Johnston

Audibility of phase shifts and time delays
Reply #32
For your inspection, comment and listening enjoyment

in the uploads forum:

Uploads forum post Interchannel delay sampler

(1) A 0.45 second 24/192 file with a unit impulse in the middle was created
(2) It was appended 3 times to create a file with 4 impulses
(3) A number of copies were created with various delays added to the Left channel. 1,2,3,4,5,7,9,11,15,19,27 samples @ 192 kHz. Units of about 5 uSec
(4) All files were low pass filtered @ 2 KHz with a 10th order minimum phase low pass filter
(5) All files were down sampled to 4416
(6) all files were normalized to 90% (-1 dB FS)

  • mzil
  • [*][*][*][*][*]
Audibility of phase shifts and time delays
Reply #33
Oops, I responded in the upload thread so I'll put it here instead.

I may be hearing artifacts in the conversion, not the clicks.

Code: [Select]
foo_abx 2.0 report
foobar2000 v1.3.3
2015-04-24 17:11:12

File A: reference impulses 4 2klp 4416 norm.flac
SHA1: fb1ca13adfb07144138724d31d58dcec72a40a47
File B: impulses 4 2klp 4416 norm shift 9 samples .flac
SHA1: 257245f5370395be10108dd0dc99ca4f2bcc3a84

Output:
DS : Primary Sound Driver
Crossfading: NO

17:11:12 : Test started.
17:14:23 : 01/01
17:15:17 : 02/02
17:15:38 : 03/03
17:15:51 : 04/04
17:16:33 : 04/05
17:16:52 : 05/06
17:17:37 : 06/07
17:17:45 : 07/08
17:17:45 : Test finished.

 ----------
Total: 7/8
Probability that you were guessing: 3.5%

 -- signature --
d98599b14d1dc93a24479b6bbb054825e2329d6a

I'm pretty sure I can ace this test if tried again because the difference became more obvious to me towards the end.
 

 

  • Last Edit: 24 April, 2015, 08:28:56 PM by mzil

Audibility of phase shifts and time delays
Reply #34
Oops, I responded in the upload thread so I'll put it here instead.

I may be hearing artifacts in the conversion, not the clicks.

Code: [Select]
foo_abx 2.0 report
foobar2000 v1.3.3
2015-04-24 17:11:12

File A: reference impulses 4 2klp 4416 norm.flac
SHA1: fb1ca13adfb07144138724d31d58dcec72a40a47
File B: impulses 4 2klp 4416 norm shift 9 samples .flac
SHA1: 257245f5370395be10108dd0dc99ca4f2bcc3a84

Output:
DS : Primary Sound Driver
Crossfading: NO

17:11:12 : Test started.
17:14:23 : 01/01
17:15:17 : 02/02
17:15:38 : 03/03
17:15:51 : 04/04
17:16:33 : 04/05
17:16:52 : 05/06
17:17:37 : 06/07
17:17:45 : 07/08
17:17:45 : Test finished.

 ----------
Total: 7/8
Probability that you were guessing: 3.5%

 -- signature --
d98599b14d1dc93a24479b6bbb054825e2329d6a

I'm pretty sure I can ace this test if tried again because the difference became more obvious to me towards the end.

This suggests to me that I need to add the files for 2 and 4 sample delays which I have.

How did you do with the file with a 1 sample delay?

If people reliably detect the file with 1 sample delay, then I need to shift the generation process up to a higher sample rate and provide files with less than 5 uSec interchannel delays.

  • mzil
  • [*][*][*][*][*]
Audibility of phase shifts and time delays
Reply #35
27 samples shift is so easy I just run the Xs. My perception is of a lateral shift across the horizon of the soundstage, of the target sound:

Code: [Select]

2015-04-25 00:26:33

File A: reference impulses 4 2klp 4416 norm.flac
SHA1: fb1ca13adfb07144138724d31d58dcec72a40a47
File B: impulses 4 2klp 4416 norm shift 27 samples .flac
SHA1: 80197bcba7ed96997778e7752deb1cf16babffa4

Output:
DS : Primary Sound Driver
Crossfading: NO

00:26:33 : Test started.
00:26:39 : 01/01
00:26:44 : 02/02
00:26:48 : 03/03
00:26:53 : 04/04
00:26:58 : 05/05
00:27:02 : 06/06
00:27:05 : 07/07
00:27:09 : 08/08
00:27:09 : Test finished.

 ----------
Total: 8/8
Probability that you were guessing: 0.4%

 -- signature --
a1da43ab4c78a4ecb41725eacf714077512bdee8


9 samples shift is very difficult. I no longer hear a difference in the target sound however from my perception there is an extremely subtle difference in the background hiss. It is so subtle it is one of those things where one can't tell if it is simply a minute level change, like say .2 dB, or if there is actually a change in the spectral balance of the hiss. All I know is there is a tiny difference.

In any event, since these 9 sample shift results aren't in the spirit of what we are truly testing for, I post them simply to expose how we might see some people [like the organic twins] claim to be able to "hear" a difference at your 9 sample shift level even though it might just be some artifact with your conversion process, I'm not sure:

Code: [Select]

foo_abx 2.0 report
foobar2000 v1.3.3
2015-04-24 23:56:52

File A: impulses 4 2klp 4416 norm shift 9 samples .flac
SHA1: 257245f5370395be10108dd0dc99ca4f2bcc3a84
File B: reference impulses 4 2klp 4416 norm.flac
SHA1: fb1ca13adfb07144138724d31d58dcec72a40a47

Output:
DS : Primary Sound Driver
Crossfading: NO

23:56:52 : Test started.
23:57:42 : 01/01
23:57:49 : 02/02
23:57:56 : 03/03
23:58:01 : 04/04
23:58:13 : 05/05
23:58:17 : 06/06
23:58:22 : 07/07
23:58:26 : 08/08
23:58:30 : 09/09
23:58:36 : 10/10
23:58:36 : Test finished.

 ----------
Total: 10/10
Probability that you were guessing: 0.1%

 -- signature --
bbdaa32033488f9bd2442eed5bc22c34151f22cf


I can't hear any difference, of any kind, on the 1 sample shift level.

P.S. What is that oddball file of the five in your folder, recorded at a lower level, also with "9 samples" as part of its name? ... Oh wait. It doesn't say "norm", so that means it hasn't been normalized I guess?... Hmm, maybe your normalizing procedure is what I'm keying on for that change to the hiss at the 9 sample shift level?



Audibility of phase shifts and time delays
Reply #36
(4) All files were low pass filtered @ 2 KHz with a 10th order minimum phase low pass filter
Why did you use a 2 kHz low pass filter ? AFAIK Woodinville proposed a 2 kHz high pass filter for the pulse train. Have I missed something ?

See what YOU hear.
JJ: can your pulse train be made with a DAW ? Your method sounds like a Matlab style recipe to me and I don't think I have the tools. Perhaps you can tell us what scientific literature tells us what we are supposed to hear ? I honestly don't care much what "I" can hear.

Audibility of phase shifts and time delays
Reply #37
(4) All files were low pass filtered @ 2 KHz with a 10th order minimum phase low pass filter
Why did you use a 2 kHz low pass filter ? AFAIK Woodinville proposed a 2 kHz high pass filter for the pulse train. Have I missed something ?


I applied the 2 KHz low pass filter to all the samples, both reference and shifted, not because that seemed to be indicated by JJ's instructions but because it ensured that the listening test was all about timing, and not the least about differences in frequency response.  These test files are IMO not very demanding and they should play reasonably accurately on a wide variety of different kinds of equipment. This makes a good test of listeners, not equipment.
  • Last Edit: 25 April, 2015, 07:45:48 AM by Arnold B. Krueger

Audibility of phase shifts and time delays
Reply #38
27 samples shift is so easy I just run the Xs. My perception is of a lateral shift across the horizon of the soundstage, of the target sound:


That was completely intentional - its the confidence building training file.

Quote
P.S. What is that oddball file of the five in your folder, recorded at a lower level, also with "9 samples" as part of its name? ... Oh wait. It doesn't say "norm", so that means it hasn't been normalized I guess?... Hmm, maybe your normalizing procedure is what I'm keying on for that change to the hiss at the 9 sample shift level?


That was a mistake - an intermediate file that crept in and is removed from the new data zip file which contains an additional 2 files for 2 sample and 4 sample (@192 KHz) delay testing:

http://www.hydrogenaud.io/forums/index.php...st&p=896783

Contents:

The reference file. No interchannel delay. It's a downsampled version of the file I used to build the files with the interchannel delays. Compare all the other files to it.

27 sample delay = 140.6 uSec - this is a confidence builder - you should be able to complete it easily and accurately. It is also a test of the suitability of your test environment. Do not proceed to the shorter delays until you can do well with this file.

9 sample delay = 46.85 uSec - This is roughly the result that David L. Clark reported.

4 sample delay = 20.8 uSec

2 sample delay = 10.4 uSec

1 sample delay = 5.2 uSec

If you edit these files and zoom in on one of the impulses and compare the L&R channel you should see the delay which will be less than 1 sample wide for the last 3 files.  This should put to rest any misapprehensions about 44 Khz sampling not being able to encode time differences with high resolution.
  • Last Edit: 25 April, 2015, 07:29:22 AM by Arnold B. Krueger

  • bandpass
  • [*][*][*][*]
Audibility of phase shifts and time delays
Reply #39
JJ: can your pulse train be made with a DAW ? Your method sounds like a Matlab style recipe to me and I don't think I have the tools.

You can do it with sox (if I understand JJ's instructions correctly):

Code: [Select]
sox -V -n delay-1.wav synth 1s sq pad 100s 379s sinc -a 82 -t 4k -M 4k repeat 399 channels 2 delay 1s


Assuming that –82 dB is sufficient for the stop-band.  This gives filter order 64; transition-band between 2k & 6k.

Omit the 'delay 1s' for the no-delay version.

Audibility of phase shifts and time delays
Reply #40
JJ: can your pulse train be made with a DAW ? Your method sounds like a Matlab style recipe to me and I don't think I have the tools.

You can do it with sox (if I understand JJ's instructions correctly):

Code: [Select]
sox -V -n delay-1.wav synth 1s sq pad 100s 379s sinc -a 82 -t 4k -M 4k repeat 399 channels 2 delay 1s


Assuming that –82 dB is sufficient for the stop-band.  This gives filter order 64; transition-band between 2k & 6k.

Omit the 'delay 1s' for the no-delay version.



Seems like its time to grind out a set of samples and put them into the uploads forum.  There may be some wisdom to glean from the comparison of how the two sets of samples work.

  • bandpass
  • [*][*][*][*]
Audibility of phase shifts and time delays
Reply #41
Seems like its time to grind out a set of samples and put them into the uploads forum.  There may be some wisdom to glean from the comparison of how the two sets of samples work.

Two files here.

  • mzil
  • [*][*][*][*][*]
Audibility of phase shifts and time delays
Reply #42
4 sample delay = 20.8 uSec
 

  Here's my first shot at it and might I add this is before my first cup of coffee for the day. 

Code: [Select]

foo_abx 2.0 report
foobar2000 v1.3.3
2015-04-25 11:19:24

File A: impulses 4 2klp 4416 norm shift 4 samples.flac
SHA1: 301a8107b8d9e8f0a61ac6f722aacfd86a088722
File B: reference impulses 4 2klp 4416 norm.flac
SHA1: fb1ca13adfb07144138724d31d58dcec72a40a47

Output:
DS : Primary Sound Driver
Crossfading: NO

11:19:24 : Test started.
11:21:46 : 00/01
11:22:04 : 01/02
11:22:21 : 02/03
11:22:55 : 03/04
11:23:15 : 04/05
11:23:34 : 05/06
11:24:40 : 06/07
11:24:50 : 07/08
11:24:50 : Test finished.

 ----------
Total: 7/8
Probability that you were guessing: 3.5%

 -- signature --
07a9ed0ac5de3bdebe80d0761ce9e408ee91b465


The good news is here on this test, above,  I'm truly, at least to the best of my ability, relying solely on directional cues of the target sound and ignoring any differences in the background hiss . [But who knows, maybe I am keying on it at a subconscious level]. Would you please speak to why I'm hearing, or thinking I'm hearing, differences in the background hiss on my earlier test. Thanks.

Interestingly my perceived HRTF mental processing for directional cues is claiming not just a lateral shift but also a tiny vertical shift, almost as if you snuck in a small EQ alteration to elicit that, ha-ha.

Tomlinson Holman says we can discern about 10,000+ distinct directions. I wonder if checking his work, and that of others, to determine in what direction and frequency we are most sensitive to tiny differences in directional arc, measured in degrees, and then designing these tests using those frequencies in those directions, makes better sense to maximize sensitivity? For instance, maybe we should be using 3.5 kHz tones at the time spacing and L vs. R balance level which directionally implies slightly to the side of center? Just a guess.

This  4 sample shift wasn't easy but instead of taking 5 minutes to test I suspect a shorter delay of half that, 2 samples, would take an hour of concentration or more, if I can even discern it at all, so I don't think I'll be trying that any time soon. Do you have a 3 sample shift file? That might be doable. Please post it and thanks for these tests. They are fun and educational.
  • Last Edit: 25 April, 2015, 03:12:59 PM by mzil

Audibility of phase shifts and time delays
Reply #43

  • mzil
  • [*][*][*][*][*]
Audibility of phase shifts and time delays
Reply #44
Thanks, I haven't tackled that 3 sample one yet, but here's my second stab at 4 sample shift, after coffee, to verify the first try wasn't just dumb luck. 9/10 score not too shabby:

Code: [Select]

foo_abx 2.0 report
foobar2000 v1.3.3
2015-04-25 12:52:35

File A: impulses 4 2klp 4416 norm shift 4 samples.flac
SHA1: 301a8107b8d9e8f0a61ac6f722aacfd86a088722
File B: reference impulses 4 2klp 4416 norm.flac
SHA1: fb1ca13adfb07144138724d31d58dcec72a40a47

Output:
DS : Primary Sound Driver
Crossfading: NO

12:52:35 : Test started.
12:53:18 : 01/01
12:53:44 : 02/02
12:54:09 : 03/03
12:54:36 : 04/04
12:55:06 : 05/05
12:55:45 : 05/06
12:56:13 : 06/07
12:56:47 : 07/08
12:58:17 : 08/09
12:58:51 : 09/10
12:58:51 : Test finished.

 ----------
Total: 9/10
Probability that you were guessing: 1.1%

 -- signature --
695a168ab31fd7a7d6d6e73a44b7c27772740f85


--

WOW! They've added keyboard shortcuts so A, B, and X can be selected by their keyboard letters! Woo-hoo. News to me. That really does reduce cognitive load and greatly adds to my sensitivity. Has that always been there but I didn't notice it?

Previously, these were the mental steps:

Thought: "I want to hear the alternate source B now"

A) Stop listening to music

B) open eyes

C) activate visual cortex

D) locate mouse cursor on screen

E) place hand on mouse

F) move hand in analogous fashion to desired cursor motion

G) make minor tracking adjustments as cursor moves across screen

H) stop on "B" box

I) correct for overshoot if applicable

J) click "B" box

K) disengage visual processing

L) close eyes

M) listen to newly selected B source

N) compare to "A" sound in mind's memory, from a couple of seconds ago, prior to the gymnastics to get to "B", and pretend that there was no loss of concentration from having to go through that multi-step process, steps A through M above, which involved switching to an alternate sense, sight, and then returning back to hearing.

NOW, thanks to the keyboard shortcuts:

Thought: "I want to hear the alternate source B now"

A) press downward on right hand  finger already in position, resting on keyboard letter "B"

B) listen to newly selected B source, INSTANTLY, to see if the transition was detectable. "Memory" not really being used, in my opinion.

Hallelujah!
  • Last Edit: 25 April, 2015, 04:27:09 PM by mzil

  • mzil
  • [*][*][*][*][*]
Audibility of phase shifts and time delays
Reply #45
I tried and I am unable to hear 3 samples shift. Looks like under these conditions my best is 4 sample shift.

Audibility of phase shifts and time delays
Reply #46
I tried and I am unable to hear 3 samples shift. Looks like under these conditions my best is 4 sample shift.



Take heart! Clark's test had results that showed the threshold of detection as being about twice that. I ascribe the difference to our casting it as an ABX test which was not nearly as easy to work out way back when.

Hopefully others will try it and report their results so we are not reliant on the results of just one person. I tried it and my old ears barely make it through the first test with 95% confidence.

  • bandpass
  • [*][*][*][*]
Audibility of phase shifts and time delays
Reply #47
Attached files generated by:

Code: [Select]
for d in 0 1; do sox -V -n delay-${d}.flac synth 1s sq pad 100s 379s sinc -a 82 -t 4k -M 4k repeat 399 channels 2 delay ${d}s norm -1 trim 0 4; done


My analysis is that the file with the delay adds a 1 sample (22 uSec) delay to the left channel where the pulse rate is 100 Hz.

Good, I believe that's what JJ intended.

Can this methodology produce delays that are other than 1 sample?

Yes, any positive integer can be substituted.

Can this methodology produce delays that are less than 1 sample?

I think a linear-phase sub-sample delay requires a rate change (but that's easy enough to add).

Of course, as shorter delay could be introduced simply by running the chain at a higher rate (but bear in mind possible resampling in the OS or hardware).

What are the typical ABX results?

Don't know if I'm typical, but I couldn't ABX the two files.

Audibility of phase shifts and time delays
Reply #48
Attached files generated by:

Code: [Select]
for d in 0 1; do sox -V -n delay-${d}.flac synth 1s sq pad 100s 379s sinc -a 82 -t 4k -M 4k repeat 399 channels 2 delay ${d}s norm -1 trim 0 4; done


My analysis is that the file with the delay adds a 1 sample (22 uSec) delay to the left channel where the pulse rate is 100 Hz.

Good, I believe that's what JJ intended.

Can this methodology produce delays that are other than 1 sample?

Yes, any positive integer can be substituted.

Can this methodology produce delays that are less than 1 sample?

I think a linear-phase sub-sample delay requires a rate change (but that's easy enough to add).

Of course, as shorter delay could be introduced simply by running the chain at a higher rate (but bear in mind possible resampling in the OS or hardware).

What are the typical ABX results?

Don't know if I'm typical, but I couldn't ABX the two files.



FWIW (not much given the age-attenuated state of my hearing) neither can I. 

However, the one value of delay we have at hand is one that has been only slightly bettered by other means, and is in the range where audibility may be elusive for many listeners.

The next logical step would appear to be to produce a family of files similar to those already posted on the downloads forum for the other methodology, covering a range of delays, starting with one that is so large as to be unmistakable (e.g. about 140 uSec).

  • bandpass
  • [*][*][*][*]
Audibility of phase shifts and time delays
Reply #49
In short, 10 microseconds is nowhere near the actual abilities of 4416 to convey timing differences among
 signals is:  1/(44100*65536) =

3.460042871315193e-10  seconds. Less than a nanosecond.
Let's see how this pans out in a simple experiment:
  • Input 1 = single-sample pulse at the 1GHz sample-rate, band-limited to CD rate (i.e. the impulse becomes a sinc pulse).
  • Input 2 = Input 1 delayed by 1 sample i.e. 1 ns.
  • Outputs 1 & 2 obtained by converting inputs 1 & 2 to redbook then back to original rate/depth.
Does ‘degrading’ to redbook preserve the 1ns difference?

Results of three runs (showing sample values around the peak):
Code: [Select]
            Time    Input 1         Input 2         Output 1        Output 2
           -----    -------------  ------------    -------------  ------------
               0    0.99999992736  0.99999989569  0.9953168775    0.99532626430
           1e-09    0.99999995297  0.99999992736  0.99531690404  0.99532629550
           2e-09    0.99999997392  0.99999995297  0.99531692453  0.99532632064
           3e-09    0.99999998789  0.99999997392  0.99531693989  0.99532634020
           4e-09    0.99999999674  0.99999998789  0.99531694921  0.99532635370
           5e-09    0.99999999953*  0.99999999674  0.99531695293*  0.99532636208
           6e-09    0.99999999674  0.99999999953*  0.99531695107  0.99532636441*
           7e-09    0.99999998789  0.99999999674  0.99531694362  0.99532636115
           8e-09    0.99999997392  0.99999998789  0.99531693012  0.99532635184
           9e-09    0.99999995297  0.99999997392  0.99531691102  0.99532633740
           1e-08    0.99999992736  0.99999995297  0.99531688634  0.99532631692

               0    0.99999992736  0.99999989569  0.99531617435  0.99532265775
           1e-09    0.99999995297  0.99999992736  0.99531620182  0.99532268988
           2e-09    0.99999997392  0.99999995297  0.99531622371  0.99532271642
           3e-09    0.99999998789  0.99999997392  0.99531624001  0.99532273691
           4e-09    0.99999999674  0.99999998789  0.99531625025  0.99532275181
           5e-09    0.99999999953*  0.99999999674  0.99531625491*  0.99532276113
           6e-09    0.99999999674  0.99999999953*  0.99531625398  0.99532276485*
           7e-09    0.99999998789  0.99999999674  0.99531624746  0.99532276252
           8e-09    0.99999997392  0.99999998789  0.99531623488  0.99532275461
           9e-09    0.99999995297  0.99999997392  0.99531621672  0.99532274110
           1e-08    0.99999992736  0.99999995297  0.99531619297  0.99532272201

               0    0.99999992736  0.99999989569  0.99532299675  0.99532591924
           1e-09    0.99999995297  0.99999992736  0.99532302469  0.99532594904
           2e-09    0.99999997392  0.99999995297  0.99532304704  0.99532597326
           3e-09    0.99999998789  0.99999997392  0.99532306381  0.99532599188
           4e-09    0.99999999674  0.99999998789  0.99532307498  0.99532600446
           5e-09    0.99999999953*  0.99999999674  0.99532308057*  0.99532601144
           6e-09    0.99999999674  0.99999999953*  0.99532308010  0.99532601284*
           7e-09    0.99999998789  0.99999999674  0.99532307405  0.99532600865
           8e-09    0.99999997392  0.99999998789  0.99532306241  0.99532599840
           9e-09    0.99999995297  0.99999997392  0.99532304518  0.99532598257
           1e-08    0.99999992736  0.99999995297  0.99532302190  0.99532596115
Input 1 shows the peak at 5ns in.  As expected, input 2 shows exactly the same samples as input 1 but delayed 1ns.

The outputs vary between runs and are noisy (as expected, due to the random TPDF dither applied at the redbook conversion stage) but look good: in each case, the peak value time matches that of the respective input (peak values have been highlighted with asterisks).

The slight attenuation in the outputs may be due to headroom being applied at the dither stage and/or HF roll-off during resampling but in any case, it's timing we're interested in here.

So the conclusion is yes, redbook does preserve timing at least down to 1 ns.

Code used (N.B. sox v14.4.2):
Code: [Select]
sox -r 1000000k -n in1.wav synth 1s sq pad .1 .1 rate -vtf 44100 norm rate -vtf 1000000k norm
sox in1.wav in2.wav delay 1s
sox in1.wav -b 16 tmp.wav rate -vtf 44100
sox tmp.wav -b 32 out1.wav rate -vtf 1000000k
sox in2.wav -b 16 tmp.wav rate -vtf 44100
sox tmp.wav -b 32 out2.wav rate -vtf 1000000k
sox -M in1.wav in2.wav out1.wav out2.wav -t dat - trim .099999995 11s