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Resampling down to 44.1KHz, Is there a method that will not colour the sound?
Martel
post May 25 2008, 18:16
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It would be better to get some NATURAL sound samples and try to ABX a 24 kHz lowpass applied to them. If the third harmonics had the same amplitude as the base frequency, then it is very unnatural sound and unlike anything you're going to listen to. The tweeters are usually not dimensioned to take such amplitudes, so be cautious!
Also, try doing a similar test you did with headphones to rule out environment interaction.

If a 24 kHz lowpass degrades your general listening experience, then you can make a clear conclusion - 48 kHz (and less) sampling rate is not enough for you.


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cabbagerat
post May 25 2008, 19:13
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QUOTE (Martel @ May 25 2008, 09:16) *
It would be better to get some NATURAL sound samples and try to ABX a 24 kHz lowpass applied to them. If the third harmonics had the same amplitude as the base frequency, then it is very unnatural sound and unlike anything you're going to listen to. The tweeters are usually not dimensioned to take such amplitudes, so be cautious!
Also, try doing a similar test you did with headphones to rule out environment interaction.
Well, yeah - real music would be better, but test signals do tell you something interesting - that it's possible to hear the difference in some cases. This makes the search for other (possibly real music) cases more interesting. I agree that a headphone test would be interesting.


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MLXXX
post May 26 2008, 04:06
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QUOTE (Martel @ May 26 2008, 03:16) *
It would be better to get some NATURAL sound samples and try to ABX a 24 kHz lowpass applied to them. If the third harmonics had the same amplitude as the base frequency, then it is very unnatural sound and unlike anything you're going to listen to.


The problem with that approach is that it would probably be inconclusive. Loudspeakers differ in their responses to complex waveforms. A filter to separate out the above 24Khz components from the below 24Khz components would be problematic to implement if an attempt were made to separately amplify and reproduce the above 24KHz components.

The test I did was quite clinical, in creating the harmonic separate to the fundamental. I note that some natural instruments do create strong odd order harmonics.

I did actually find an orchestral recording that sounded better to my ears at its 96KHz sample rate on my system, and I made reference to it earlier in this thread. I did not receive permission to upload it, but even if I had been able to upload it, I'm sure opinions would have varied as to why the sound was different.

The differences involved are very small. I feel I have to move on to other things rather than carry out further amateur 'investigations' into the effects of differing sample rates, with my domestic equipment. I feel satisfied that for many recordings, 48KHz is quite adequate when tested practically, and I note that this is underpinned by a body of expert opinion that 48KHz is quite adequate in theory; as for example expressed in this thread.

One aspect is that studio microphone design may not attach much importance to frequencies much above 20KHz, as such frequencies have been considered of little relevance, and not worth the design effort, and cost of manufacture.

Cheers.
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greynol
post May 26 2008, 06:29
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QUOTE (MLXXX @ May 25 2008, 20:06) *
I did actually find an orchestral recording that sounded better to my ears at its 96KHz sample rate on my system, and I made reference to it earlier in this thread. I did not receive permission to upload it, but even if I had been able to upload it, I'm sure opinions would have varied as to why the sound was different.

You don't need permission to upload. Maybe you've exceeded your allotment, in which case delete some of your older stuff or use a third party provider.


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MLXXX
post May 26 2008, 13:48
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Thanks greynol.

QUOTE (MLXXX @ May 19 2008, 01:29) *
A 96KHz extract

I have always found combined strings a good test for audio equipment. I have come across a recording of an orchestra playing The Earth Overture by Kosuke Yamashita.

The format is 7.1 channel 96KHz 24-bit linear PCM. (The Blu-ray reference disc has been released by Q-TEC.)

The audio quality is very good. I found that when I converted a short extract to 48KHz with Audition 3, the quality was reduced slightly (at least as played back by my AVR). In contrast, many other recordings I have experimented with have revealed no apparent (to me) audible differences when downsampled to 48KHz.

The 48KHz version is not quite as smooth sounding. I find this noticeable in the harmony between the string sections. With the 96KHz version, the sounds blend such that the strings taking the lower part are less noticeable. I'll upload a 9 second extract in this post if possible.

And now it is possible:
Attached File  earthsong_9seconds.flac ( 1.42MB ) Number of downloads: 147

Attached File  earthsong_9secondsAuditionConvertedto48KHz.flac ( 879.42K ) Number of downloads: 153


There will no doubt be arguments that any differences are due to the playback equipment and indeed that may be so. On my equipment I prefer the unconverted version, but the difference is extremely slight, and only noticeable at all if my ears are fresh. (Note: to keep the upload to a manageable file size, only front left and front right of the 7.1 channels have been extracted for this test file.)

QUOTE (cabbagerat @ May 26 2008, 04:13) *
I agree that a headphone test would be interesting.

*if sending 8333Hz to one ear and switching 24999Hz on and off for the other ear, there was no change I could perceive, whether the 24999 was on or off.
*if sending the 24999Hz to the same ear as the 8333Hz, the sound volume seemed slightly less and the sound quality was different when the 24999 signal was on.


If anyone wants to experiment with the 8333Hz and 24999Hz tones, be mindful that 8333Hz is an irritating high frequency and is likely to lead to a headache for anyone in earshot, if ABXing is attempted. The playback volume should be kept at a modest setting. The 24999Hz tone will not be audible by itself but will stress any tweeter that is attempting to reproduce it - another reason to avoid a high playback volume.

Here are test files in stereo:
Attached File  8333left_nothing_right_192.flac ( 230.01K ) Number of downloads: 147

Attached File  8333left_24999right_192.flac ( 462.23K ) Number of downloads: 121


This post has been edited by MLXXX: May 26 2008, 15:18
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Kees de Visser
post May 26 2008, 22:41
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QUOTE (MLXXX @ May 26 2008, 13:48) *
There will no doubt be arguments that any differences are due to the playback equipment and indeed that may be so.
When I re-convert your 48kHz sample to 96kHz and subtract it from the original, there is almost exclusively energy above 22kHz. The part of the signal in the (presumed) audible band below 22kHz is noise at about 2 LSB@24bit level. This will most likely be swamped by the DAC's internal noise during playback. It seems that if you hear differences between the 48 and 96kHz version, it's caused by the >22kHz part of the signal.
Would you be able to record your speaker's output with a microphone? Needless to say that both need to have sufficient bandwidth for this. It would be interesting to analyse the signal that is hitting your ear.
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MLXXX
post May 27 2008, 02:36
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My equipment is not suitable for measuring high frequency performance. The Rode microphone has very low noise but is limited in its response above 20KHz. I measured the response of the speaker and microphone combination to a sine wave at 25KHz. It was 15dB down compared with 8333Hz. (Also, the oscilloscope revealed a sinusoidal response at 25KHz even if the waveform input was changed from a sine wave to a triangular wave!)

If the human listening experience is different when frequencies above 20KHz are allowed to pass through the recording and reproduction chain, this may be because of indirect effects, e.g. instantaneous changes in standing wave patterns in the listening room that an ‘acoustically athletic’ system is able to stimulate rapidly and with fine detail. I am only speculating. We may perceive the higher frequencies (and perhaps the faster response time to change) indirectly because of the effect on the amplitude and apparent timbre of frequencies we can hear.

As I have indicated, I can perceive a strong third harmonic of an 8333Hz tone indirectly by its effect on the tonal quality I hear. Of course I cannot hear the third harmonic (24999Hz) if it is presented as a continuous tone, without the fundamental.

By the way, with the particular orchestral sample I have uploaded, I sometimes hear the 96Khz version as a little cleaner (less muddied) than the 48KHz version. The difference is slight.

I'd be interested in whether others perceive a difference when ABXing (at moderate volume!) the stereo files I provided at the end of post #105.

This post has been edited by MLXXX: May 27 2008, 08:49
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SebastianG
post May 27 2008, 09:20
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QUOTE (MLXXX @ May 27 2008, 03:36) *
If the human listening experience is different when frequencies above 20KHz are allowed to pass through the recording and reproduction chain, this may be because of indirect effects

say "nonlinear".

Remember the linearity property?
f is linear => f(x) + f(y) = f(x + y)
Consider f to be your sound reproduction system. 'x' would be the 8 kHz tone and 'y' the 24 kHz tone.
Since you can differentiate between f(x) and f(x+y) the system 'f' might be nonlinear.

QUOTE (MLXXX @ May 27 2008, 03:36) *
e.g. instantaneous changes in standing wave patterns in the listening room

huh?!

QUOTE (MLXXX @ May 27 2008, 03:36) *
We may perceive the higher frequencies [...] indirectly because of the effect on the amplitude and apparent timbre of frequencies we can hear. As I have indicated, I can hear a strong third harmonic of an 8333Hz tone indirectly by its effect on the tonal quality I perceive. Of course I cannot hear the third harmonic (24999Hz) if it is presented as a continuous tone, without the fundamental.

Be careful with the wording here. There's no indication that you can hear the third harmonic. It's probably only its effect on the recording/playback chain. A nonlinear chain would explain the change in amplitude of the fundamental frequency you experienced. This would be an artefact, an imperfection of the chain.

The interesting question is now: Is your playback chain really to blame? If yes, this would mean that what you hear is not really what it's supposed to sound. You could try to test this by recording the sound you're hearing and analyze whether the third harmonic somehow affects the audible spectrum measurably.

In the presumably unlikely case your playback system is not distorting the sound you might be on to something. I guess this would be an indication for nonlinear effects in the human auditory system that cause intermodulation or something. This is something that I can't completely rule out but is unlikely and to my knowledge nobody has reported such findings.

Cheers,
SG

This post has been edited by SebastianG: May 27 2008, 09:31
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MLXXX
post May 27 2008, 10:10
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In my 2nd and 3rd tests in post #100, the 24999Hz tone was amplified in a separate channel in the Audio Video Receiver and sent to its own speaker.

I was nevertheless concerned that the power drain of reproducing the 24999 tone in addition to the 8333 tone might be slightly affecting the performance of the channel carrying the 8333Hz tone, so in a supplementary test I'll label test 4, I played the stereo test files on an older AVR for the harmonic and a portable cassette recorder for the fundamental. (I used my Audigy 4 hub to decode the 192KHz SPDIF, and used its line level outputs to feed the independent amplifiers.)

I could still hear a difference when 24999Hz started coming from a hi-fi speaker driven by the old AVR, provided 8333Hz was coming from the portable cassette player.

I conclude from this that either or both of my ears are in fact slightly non-linear for a tone of 8333Hz that has a 24999Hz tone piggy-backed on top. If others can try out the stereo test files I have provided, they may find that they too have slightly non-linear hearing.

This post has been edited by MLXXX: May 27 2008, 12:01
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Martel
post May 27 2008, 12:47
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So you say that:
1) If you play back just the 25kHz sinus, you hear nothing
2) If you play the 8333 Hz sinus in addition, you hear something
3) Now, if you turn off the 25kHz one (which you're not able to hear), you clearly hear SOMETHING ELSE?!!

This is kind of weird.


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2Bdecided
post May 27 2008, 13:03
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Test 4 is interesting - thank you MLXXX.

It doesn't work for me, but maybe it'll work for someone.

Of course, the Audigy can still come under suspicion, but I have no idea how realistic such an explanation would be. You could re-capture the output and examine it to remove any possible suspicion.

FWIW The definitive test is with two separate signal generators, feeding two separate amplifiers and speakers.

Cheers,
David.
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MLXXX
post May 27 2008, 14:29
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QUOTE (Martel @ May 27 2008, 21:47) *
So you say that:
1) If you play back just the 25kHz sinus, you hear nothing
2) If you play the 8333 Hz sinus in addition, you hear something
3) Now, if you turn off the 25kHz one (which you're not able to hear), you clearly hear SOMETHING ELSE?!!

This is kind of weird.

Yes but the 24999 Hz was strictly synchronised with the 8333 Hz. On the oscilliscope, the signal from the microphone had a very different waveshape when the 8333 and 24999 from the separate hi-fi speakers combined (the mic was at about 1.5m). It was not the perfect waveshape that cooledit displays for a fundamental plus third harmonic. This was, I presume, because the microphone (and to a degree the tweeter) would have struggled to operate at 25KHz.

There may have been a small intermodulation product at 16666Hz (the difference frequency, and the 2nd harmonic) causing the different timbre for my hearing, but I really don't know how non-linearity works with human hearing.

For my ears, the effect was slight, but ABXable.

QUOTE (2Bdecided @ May 27 2008, 22:03) *
It doesn't work for me, but maybe it'll work for someone.

Indeed. Thanks for trying it out.

______________________

EDIT:
Using Google and the search words 'non-linear' and 'ear' I've found a dozen or so relevant sites, though many sources are only fully downloadable for a fee. Here are a couple of weblinks that refer to difference tones that can be heard because of the non-linearity of human hearing:-

http://www.isvr.soton.ac.uk/SPCG/Tutorial/...-difference.htm - note: the embedded test file on this webpage does not appear to be available

http://www.mp3-tech.org/programmer/docs/no...man_hearing.pdf - a more technical article


This post has been edited by MLXXX: May 27 2008, 16:47
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Martel
post May 27 2008, 18:01
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QUOTE (MLXXX @ May 27 2008, 05:29) *
Yes but the 24999 Hz was strictly synchronised with the 8333 Hz. On the oscilliscope, the signal from the microphone had a very different waveshape when the 8333 and 24999 from the separate hi-fi speakers combined (the mic was at about 1.5m).
It might be strictly synchronized on the soundcard output, but as soon as those signals come from two places far apart, there will always be a phase-shift field. The 8333 Hz signal should have the wave-length of about 4 centimeters (1/8333 s *330 m/s), so, theoretically (loudspeaker being an ideal point source of sound), it spawns a concentric spheric field (any sphere with center in the loudpeaker should exhibit the same signal phase). The same holds for the 25 kHz sine coming from another speaker but the wavelength is one-third (so the imaginary spherical field is more "dense"). By walking around with the microphone, the shape of the recorded wave should vary alot, depending on the relative phase of the signals at the microphone position. Moreover, wall reflections may actually reinforce or damp each of the signals and make the sound field even more complicated.
You would have to test that in an anechoic chamber but I doubt you have any at your disposal.
Borrow some good headphones and try listening to the mixed signal (both signal in both channels) versus just 8333. If you don't hear the difference like you did with the loudspeakers, you may rule out your ear nonlinearity.


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MLXXX
post May 28 2008, 08:49
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Martel, I think I have already established that non-linearity applies to my hearing, by using separate loudspeakers and two separate power amplifers (see my reference to test 4 in my response to SebastianG a few posts above).

If I remain perfectly still whilst the stereo test files are playing end to end, the quality and volume of the sound I hear varies slightly when the 24999Hz wave (from speaker 2) starts. There appear to be only two explanations for that:
(a) I can hear the 24999Hz from speaker 2 directly in its own right (this surely must be discounted as I cannot hear the 24999Hz when it is played in isolation);
(b) I hear the 24999Hz indirectly because of a slight non-linearity in my hearing (a recognised characteristic of human hearing) that allows my ear's response to speaker 1 to be influenced by the sound coming from speaker 2.


After reading the internet material on the non-linearity of human hearing, I do not find the experimental outcome surprising. However I would emphasise that the effect was very slight.
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Martel
post May 28 2008, 09:37
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QUOTE (MLXXX @ May 27 2008, 23:49) *
After reading the internet material on the non-linearity of human hearing, I do not find the experimental outcome surprising. However I would emphasise that the effect was very slight.
Do not trust everything you read. The fact that the paper was presented in a conference doesn't mean much. Do not hesitate to confirm the claims using headphones to rule out environment infulences. I would also advise to analyze the signal coming out of the DAC directly with the oscilloscope just to be sure it was as intended.


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MLXXX
post Jun 9 2008, 08:45
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Martel, I did some tests with simultaneous tones around 22KHz and 23Khz. No difference frequency (e.g. 1KHz) was audible if separate speakers were used.

This contrasts with tones much lower down such as 2500Hz and 3000Hz which for my ears produce a 500Hz difference if I listen carefully, even with a separate speaker for each tone. [There are many webpages that refer to this phenomenon of a difference frequency being audible at these lower test frequencies.]

What the explanation is for 24999 affecting the hearing of 8333 using separate loudspeakers I cannot say. Since making my previous post in this thread, I've tested using an entirely separate tone generator and found it is not necessary for the frequency to be exact for the presence of the higher frequency to be detectable.

For my ears, it is not so much a case of 'hearing' the higher tone, as being conscious of it. The effect on hearing can be a little unpleasant, and not dissimilar to having a cold. Anyway I am now looking at another aspect: time resolution affecting the downsampling process.

_____________________________________________

MLXXX's 96Khz 'repeating clicks' test file

96/24 is very topical in the home theatre community. Sound cards are only just now coming on to the market that will enable high-definition sound from Blu-ray or H-DVD to be played back on a Home Theatre PC through HDMI.

"How 'useful' is this?", we might ask, if it cannot be shown that 96Khz is superior to 48Khz.

I thought it relevant to enquire into the ability of human hearing to detect slight differences in timing as between the sound reaching one ear, and the other. I discovered articles on interaural timing differences, and interaural level differences. Some articles referred to the capacity to detect differences of tens of microseconds, whereas others referred to differences in the order of milliseconds.

With my own stereo speakers, I discovered I could detect a difference of one sampling interval at 48Khz, i.e. 20.83μS (microseconds), when listening to a click coming from both speakers, if the click was advanced or retarded by one sampling interval.

With difficulty I could even detect a difference of advancement or retardation of one sampling interval at 96Khz, i.e. 10.42μS.

'Aha!', I thought. Now I will be able to create a test to demonstrate that 48KHz is inadequate. So I ran the 96KHz test file containing the stereo click through a resampling conversion (using Adobe Audition) to 48Khz and played the click back, expecting not to be able to hear the difference of one sampling position between the left and right channels. But I was wrong. It was audible, possibly even more audible than in the original 96KHz file!

How was this possible? Well, on examining the dowsampled file I saw that Adobe Audition had flattened out the waveform if it was displaced by one sampling position. Cooledit did the same. So did Audacity. (The resampling algorithm used by N-track did not produce as noticeable a difference.)

The click sounded different at 48Khz than at 96Khz but it depended on whether the click occupied an odd sample number or an even sample number in the 96KHz file, as to how different the click at 48KHz sounded!

It is geekish to listen to single sample clicks. To make this phenomenon more accessible, I decided to create them in a regular rhythm so that they could be heard as a continuous tone. I found it easier to write my own code rather than attempt to use the tone generators in Cooledit etc.. I chose to repeat the click every 30 samples at 96Khz, and to alternate the polarity . This was equivalent to a 1600Hz square wave (with a very low duty cycle).

Knowing that readers of HA tend to be [very] keen on proof, I supply the following code that will generate a test file at 96KHz in three bursts, using GNU Octave as the software. The middle burst is displaced one sample position compared with the first and third bursts:

CODE
% Creates a waveform in accordance with the waveform values for one cycle appearing in line 5.
% The final output is at a sample rate of 96KHz in  three bursts with the middle burst offset by one sample compared with the other two.  The offset can make a difference when subsequently downsampling to 48KHz.
totalsamples=150000
c=zeros(totalsamples,1);
d=[0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;.5;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0;0
;0;0;0;0;0;0;0;0;0;0;-.5;0] #60 values
wavespace=int16(totalsamples/60)
tranche2=int16(wavespace/3)
tranche3=int16(wavespace*2/3)
offset=0
for e=1:wavespace -2
if (e==tranche2) offset=1
     elseif (e==tranche3) offset=0;
         endif
f=(60 * e)+offset
for h=1:60
c(f+h,1)=d(h);
endfor
endfor
%The next section adds fadeins and fadeouts.
for z=1:totalsamples
v=c(z,1);
% To disable the optional sine wave, add a percentage sign at the start of the next line
v=v+.05*sin(1.995*pi*z/60);
if (z==1) | (z==50000) | (z==100000) fade=0
     elseif (z==46000) | (z==96000) | (z==146000)  fade=-4000
     endif
     fade=fade+1;
     fd=abs(fade);
     if (fd<2000) v=0;
         elseif (fd<4000) v=v*(fd-2000)/2000;
endif
    c(z,1)=v;
    endfor
wavwrite('1samplewidthat96KHz---10.42microseconds---clickswithrepetitionrateof1600Hz---Optionalsinewaveat1595Hz.wav',c,96000,16)
disp('***Completed***')


[I find QtOctave convenient as a graphical user interface. It is available as a free all-inclusive zip download, as a binary for Win32. The code above runs in about 30 seconds on a 3.0 GHz single core PC using Windows XP.]

For those preferring to download the file direct, here it is as a 96Khz 16-bit wav file:
Attached File  1samplwidthat96KHz___10.42microseconds___clickswithrepetitionrateof1600Hz___withlowlevelsinewaveat1595Hz.wav ( 293.01K ) Number of downloads: 148


Here it is resampled to 48Khz 16-bit (no dither) using Audition at the maximum quality setting:
Attached File  1samplwidthat96KHz___10.42microseconds___clickswithrepetitionrateof1600Hz___withlowlevelsinewaveat1595HzRESAMPLEDto48kHz.wav ( 146.53K ) Number of downloads: 118


There is a wavering quality because of the inclusion of a low level sine wave at approximately 1595Hz. This interacts with the clicks. The clicks come every 30 samples, and last for 1 sample period, and alternate in polarity. I find the sine wave acts as a reference level for the ear making it easier to pick that the middle burst is different. (If desired, in the code, the sine wave can be disabled by using the comment character '%').

You will probably find that with file 2, the middle burst is softer and has a different tonal quality.

For those who like a graphical presentation, here is how Audition displays the files after the conversion to 48KHz:

Attached Image


The top right-hand graph is without any sine wave added (simply a click every 30 sampling periods at 96KHz, resampled to 48KHz). The drop in level according to the audition output meters is about 3.5dB for the middle burst.

So what do I conclude? I conclude that if a brief click is captured at 96KHz, and then downsampled to 48Khz there may be a change in the tonal quality and/or apparent volume. An electronic instrument could produce such sounds continuously, just as the Octave software has done.

I would conjecture, though I haven't tried this out on anything other than files of repeating clicks, that if downsampling a sound file from 96Khz to 48Khz, there can be a different result [both in the arithmetic values, and how they sound when reconstructed] if one sample of silence is added at the beginning, prior to commencement of the resampling process. This would presumably only be the case for brief clicks or other transient content.

This post has been edited by MLXXX: Jun 9 2008, 11:22
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2Bdecided
post Jun 9 2008, 11:48
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You won't be surprised to hear that...

a) I can't hear any difference, and
b) I still think there's some other explanation


If you take your 48kHz file, and resample it back to 96kHz, you'll find there's almost no difference in the waveform of the centre burst compared with the first and last bursts.

I'd be interested to know if you hear a difference between the three bursts in that
"re-sampled back to 96kHz" version.

Cheers,
David.
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Kees de Visser
post Jun 9 2008, 12:03
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QUOTE (MLXXX @ Jun 9 2008, 08:45) *
For those who like a graphical presentation, here is how Audition displays the files after the conversion to 48KHz:

Attached Image

The top right-hand graph is without any sine wave added (simply a click every 30 sampling periods at 96KHz, resampled to 48KHz). The drop in level according to the audition output meters is about 3.5dB for the middle burst.
Apparently Audition displays the sample values, which isn't an accurate representation of the analog output of the DAC. iZotope RX has an option to display (in red) the reconstructed analog waveform.
Attached Image
As you can see the difference is gone now. Looking at waveforms can be very instructive, but never forget that they are just an approximation of the DAC's output.
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SebastianG
post Jun 9 2008, 12:38
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QUOTE (MLXXX @ Jun 9 2008, 09:45) *
[...] I am now looking at another aspect: time resolution affecting the downsampling process.

btw: I'm sure you'll find plenty of "time resolution" threads on HA.org. Basically people who don't understand the sampling theorem start these kinds of discussions.

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
With difficulty I could even detect a difference of advancement or retardation of one sampling interval at 96Khz, i.e. 10.42μS.

'Aha!', I thought. Now I will be able to create a test to demonstrate that 48KHz is inadequate.

Where's the connection? It sounds like you think sub sample delays can't be represented in a discrete time signal. (A related question in many "time resolution" threads.)

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
How was this possible? Well, on examining the dowsampled file I saw that Adobe Audition had flattened out the waveform if it was displaced by one sampling position. Cooledit did the same. So did Audacity. (The resampling algorithm used by N-track did not produce as noticeable a difference.)

Does it matter how it looks? I'm sure that Cooledit and Audacity (when zoomed out) only show max and min samples which doesn't tell you the actual max/min of the reconstructed waveform.

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
[...] I chose to repeat the click every 30 samples at 96Khz, and to alternate the polarity . This was equivalent to a 1600Hz square wave (with a very low duty cycle).

It's not equivalent (in terms of the harmonics' magnitues) to a square wave unless you feed a leaky integrator with this.

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
[...] (no dither) [...]

Why not?

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
There is a wavering quality because of the inclusion of a low level sine wave at approximately 1595Hz.

What is "wavering quality"?
Let me recap: Pulses at 96000 Hz sampling frequency, alternating polarity, placed 30 samples apart.
This certainly leads to a waveform period of 60 samples => fundamental frequency = 96000/60 = 1600 Hz.
This kind of periodical wave form contains the frequencies 1600 Hz, 1600*3 Hz, 1600*5 Hz, ..., 1600*29 Hz which share the same amplitude (unlike square waves). When properly downsampled to 48 kHz it should reduce to frequencies 1600 Hz, 1600*3 Hz, ..., 1600*13 Hz. The anti-alias filter might damp the last one (1600*13=20800) slightly.

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
I find the sine wave acts as a reference level for the ear making it easier to pick that the middle burst is different.

Is it different apart from the way it looks?

QUOTE (MLXXX @ Jun 9 2008, 09:45) *
You will probably find that with file 2, the middle burst is softer and has a different tonal quality.

What would your explanation be?

Cheers,
SG

This post has been edited by SebastianG: Jun 9 2008, 13:51
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cabbagerat
post Jun 9 2008, 14:27
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MLXXX, can you please record the output of your sound card when playing these samples? Using a microphone to record what is playing through your speakers would be best, but recording the output of the DAC with a loopback cable at 96kHz would probably be best.

QUOTE (SebastianG @ Jun 9 2008, 03:38) *
QUOTE (MLXXX @ Jun 9 2008, 09:45) *

[...] I am now looking at another aspect: time resolution affecting the downsampling process.

btw: I'm sure you'll find plenty of "time resolution" threads on HA.org. Basically people who don't understand the sampling theorem start these kinds of discussions.

To save you the effort of searching, this is one of the more recent, and more complete threads. Woodinville (amongst others) cover the time resolution thing really well in some posts to that thread.

This post has been edited by cabbagerat: Jun 9 2008, 14:28


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2Bdecided
post Jun 9 2008, 15:46
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QUOTE (cabbagerat @ Jun 9 2008, 14:27) *
To save you the effort of searching, this is one of the more recent, and more complete threads.
Are you trying to torture us?! wink.gif

Cheers,
David.
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MLXXX
post Jun 9 2008, 20:43
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QUOTE (2Bdecided @ Jun 9 2008, 20:48) *
... If you take your 48kHz file, and resample it back to 96kHz, you'll find there's almost no difference in the waveform of the centre burst compared with the first and last bursts.

I'd be interested to know if you hear a difference between the three bursts in that
"re-sampled back to 96kHz" version. ...
I think so, but I will have to do a proper ABX test and report results.


QUOTE (Kees de Visser @ Jun 9 2008, 21:03) *
Apparently Audition displays the sample values, which isn't an accurate representation of the analog output of the DAC. iZotope RX has an option to display (in red) the reconstructed analog waveform. As you can see the difference is gone now. Looking at waveforms can be very instructive, but never forget that they are just an approximation of the DAC's output.
I think Audition does an approximation to a reconstruction. In contrast, Audacity makes no attempt at reconstruction. (This evening, I downloaded a trial of iZotope RX but wasn't able see how to activate an advanced reconstruction display.)

Using an oscilloscope to view the analogue output, there was no drop in apparent level when the samples were offset, so indeed the Audition reconstruction (similar to Cooledit) must only go some distance towards indicating the actual reconstruction in the hardware DAC.



QUOTE (SebastianG @ Jun 9 2008, 21:38) *
Where's the connection? It sounds like you think sub sample delays can't be represented in a discrete time signal. (A related question in many "time resolution" threads.)
Yes, I was doubtful, but am beginning to understand the concept of reconstruction.

QUOTE (SebastianG @ Jun 9 2008, 21:38) *
QUOTE (MLXXX @ Jun 9 2008, 09:45) *
There is a wavering quality because of the inclusion of a low level sine wave at approximately 1595Hz.
What is "wavering quality"?
Simply the beat at approximately 5Hz between the 1600Hz and the 1595hz (approx). [For some tests this beat is an unnecessary distraction. I have been doing further testing with the sine wave off.]

QUOTE (SebastianG @ Jun 9 2008, 21:38) *
QUOTE (MLXXX @ Jun 9 2008, 09:45) *
You will probably find that with file 2, the middle burst is softer and has a different tonal quality.
What would your explanation be?
Am still investigating. There may be a slight fault somewhere in the signal processing. Certainly the audible differences are very small.

You may be referring to the fact that suggestion can play a surprisingly powerful role in hearing perception. I recognize that. It is one reason ABX testing is so important, even when we think we can hear a difference.



QUOTE (cabbagerat @ Jun 9 2008, 23:27) *
To save you the effort of searching, this is one of the more recent, and more complete threads. Woodinville (amongst others) cover the time resolution thing really well in some posts to that thread.
Very relevant indeed, that 18 month old thread. Thanks, cabbagerat. I'm half way through it and I think I am beginning to understand how subsample timing is effectively encoded to and recoverable from wav files.
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AndyH-ha
post Jun 10 2008, 04:03
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I am pretty sure that the CoolEdit/Audition code produces the theoretically perfect reconstruction display. There is no reference to any particular hardware, you get the same display no matter what soundcard you use. There was some extended discussion of this on the Audiomaster’s forum, should you care to look for it.
http://www.audiomastersforum.net/

This post has been edited by AndyH-ha: Jun 10 2008, 04:04
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Kees de Visser
post Jun 10 2008, 06:40
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QUOTE (MLXXX @ Jun 9 2008, 20:43) *
I downloaded a trial of iZotope RX but wasn't able see how to activate an advanced reconstruction display.
I'm using the Mac version, but presume the pc version to be similar. Go to Preferences, click the Misc tab and enable the option Show analog waveform.
Quotes from the iZotope RX user manual:
QUOTE
When digital audio is played back, it gets converted to analog. The peak values in the analog waveform can be larger than the peaks in the digital waveform, leading to "analog clipping" which can be problematic in some cases. When "show analog waveform" is enabled, RX will compute an analog waveform in the background. Any peaks will be highlighted in red on top of the existing digital waveform.
In the Misc tab there's also a "Waveform interpolation order" setting, ranging from 0 to 64.
QUOTE
If you zoom into the waveform so that individual samples become visible, RX will display an upsampled analog waveform as well as the individual digital samples. The interpolation order controls the quality of upsampling. Higher values yield more accurate analog waveforms at the expense of CPU usage.
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krabapple
post Jun 10 2008, 07:16
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QUOTE (AndyH-ha @ Jun 9 2008, 23:03) *
I am pretty sure that the CoolEdit/Audition code produces the theoretically perfect reconstruction display. There is no reference to any particular hardware, you get the same display no matter what soundcard you use. There was some extended discussion of this on the Audiomaster’s forum, should you care to look for it.
http://www.audiomastersforum.net/


In corroboration of this, I have compared waveforms of rips to those of ADC recordings (m_audio 2496) of the same track, using Audition, and they look and measure extremely similarly once peak levels are matched.
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