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Resampling down to 44.1KHz, Is there a method that will not colour the sound?
MLXXX
post May 6 2008, 12:07
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Having accepted that a well dithered 24-bit source (at least at 44.1KHz or above) appears to sound the same at 16 bits (at ordinary listening levels), I am now focussing my energies on the sample rate question: is 44.1KHz a sufficient sampling rate?

As part of that exercise, I am uploading three versions of a sound file. These three versions are the result of converting a short 96/24 original sound file (of a triangle being struck) to 44.1/32:-

Audition version: Attached File  triangle_2_2496__Audition3_0__conversionTo44_1_quality999.wav ( 1.48MB ) Number of downloads: 477

Cooledit version: Attached File  triangle_2_2496__Cooleditpro2__0__conversionTo44_1__quality999.wav ( 1.48MB ) Number of downloads: 397

R8brain version: Attached File  triangle_2_2496__R8brain1_9_conversionTo44_1__VeryHigh.wav ( 1.48MB ) Number of downloads: 407



If anyone is interested in my amateur comments about these files, the relevant thread is Hydrogenaudio Forum > General Audio > The Emperor's New Sample Rate, MIX magazine wonders if "maybe CD is good enough", at post #72.

EDIT: On reflection I am now suggesting that discussion on this topic is perhaps a bit specialised and might be better suited to this thread. However please note there are already some relevant comments in the Emperor's New sample Rate thread.

This post has been edited by MLXXX: May 7 2008, 17:03
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Martel
post May 18 2008, 19:25
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A 44 kHz digital waveform PERFECTLY describes ANY signal (or mixture of signals), including phase, from 0 to 22049 Hz, if you do not consider distortion caused by finite number of amplitude quantization steps.
Just looking at the waveform, you might get suspicious about accuracy at frequencies near the Nyquist one, since the signal hardly gets 3-4 samples per period. Try zooming in the waveform in Cool Edit up to the sub-sample accuracy. There you will see some interpolated points between actual samples. These are calculated solely by upsampling. No information is lost, you may recalculate the "missing" samples any time. This "upsampling" also happens naturally in DAC upon conversion to continuous-time domain.


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MLXXX
post May 19 2008, 18:11
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QUOTE (Martel @ May 19 2008, 04:25) *
A 44 kHz digital waveform PERFECTLY describes ANY signal (or mixture of signals), including phase, from 0 to 22049 Hz, if you do not consider distortion caused by finite number of amplitude quantization steps.
Just looking at the waveform, you might get suspicious about accuracy at frequencies near the Nyquist one, since the signal hardly gets 3-4 samples per period. Try zooming in the waveform in Cool Edit up to the sub-sample accuracy. There you will see some interpolated points between actual samples. These are calculated solely by upsampling. No information is lost, you may recalculate the "missing" samples any time. This "upsampling" also happens naturally in DAC upon conversion to continuous-time domain.

I do get suspicious when I look at a digital mixdown of 19KHz and 20Khz sinewaves that were created at 44.1KHz. There are so few sample points and yet as you say cooledit manages to create a realistic graphical interpolation (with this relatively simple waveform).

In contrast, when I look at 19KHz and 20KHz sinewaves created at 96KHz and mixed digitally in cooledit, there are so many more sample points in the mixdown that sophisticated interpolation would not be necessary: you could simply join the dots with a most basic form a of integration (a resistor and capacitor). The undulations in overall amplitude at a rate of 1KHz appear to be relatively smooth, at this higher sampling rate. I could readily imagine this undulating signal surviving, despite the addition of other high frequency signals into the digital mix each needing to be 'interpolated'.

This post has been edited by MLXXX: May 20 2008, 09:55
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SebastianG
post May 21 2008, 10:14
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QUOTE (MLXXX @ May 19 2008, 19:11) *
I do get suspicious when I look at a digital mixdown of 19KHz and 20Khz sinewaves that were created at 44.1KHz. There are so few sample points and yet as you say cooledit manages to create a realistic graphical interpolation (with this relatively simple waveform).

In contrast, when I look at 19KHz and 20KHz sinewaves created at 96KHz and mixed digitally in cooledit, there are so many more sample points in the mixdown that sophisticated interpolation would not be necessary

Sounds like you still think that downmixing makes reconstruction somewhat harder. As 2B already pointed out all of the following operations are linear:
(1) sampling
(2) mixing
(3) reconstruction
It follows that
CODE
reconstruct(sample(x)) + reconstruct(sample(y))e
= reconstruct(sample(x) + sample(y))
= reconstruct(sample(x + y))


QUOTE (MLXXX @ May 19 2008, 19:11) *
[at a higher sampling rate] you could simply join the dots

So? Relevance? Seriously. Grab a good DSP book that explains sampling and reconstruction. All what's been said here has been said many many times before.

QUOTE
This thread started with the downsampling to 44.1KHz question. But if downsampling to even 48KHz is a probem, 44.1KHz would be even more so.

There's no problem with downsampling to 44.1 kHz or 48 kHz (in theory). Reconstruction is also not a problem (in theory). There are simply soundcards out there that manage to screw up reconstruction. That's about it.

This post has been edited by SebastianG: May 21 2008, 10:21
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MLXXX
post May 21 2008, 10:28
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SebastianG, thanks for taking the time to restate this, with precision. However to me it was a side issue. I only mentioned it in response to a post of Martel's [#64]. I may at some stage in my life try to immerse myself in the mathematics that you and others obviously understand so well.

At this point I would like to cut to the chase and ascertain whether there exists a section of a recording of music that people claim is impaired when downsampled to even 48KHz.

If no-one can provide such a clip, and if theory explains why this is so, then I can rest easy when purchasing material that is at 48KHz rather than 96KHz. And I could make my own recordings of musical performances with confidence at only 48KHz.

This post has been edited by MLXXX: May 21 2008, 10:49
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krabapple
post May 21 2008, 16:40
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QUOTE (MLXXX @ May 21 2008, 05:28) *
SebastianG, thanks for taking the time to restate this, with precision. However to me it was a side issue. I only mentioned it in response to a post of Martel's [#64]. I may at some stage in my life try to immerse myself in the mathematics that you and others obviously understand so well.

At this point I would like to cut to the chase and ascertain whether there exists a section of a recording of music that people claim is impaired when downsampled to even 48KHz.

If no-one can provide such a clip, and if theory explains why this is so, then I can rest easy when purchasing material that is at 48KHz rather than 96KHz. And I could make my own recordings of musical performances with confidence at only 48KHz.



For recording, why not just split the diff and record at 88.2/24bit? That's an even multiple SR of 44.1, computationally a snap if you need to downsample to CD rate. And it's well above even the ~60kHz 'safety' rate proposed by Lavry and others for surmounting any real or theoretical problems with suboptimal antialias and anti-image filters. At 88.2 you should have no pangs of anxiety (even though I think it's way overkill).


QUOTE
I do not for a moment question classic Nyquist Shannon concepts. However it is not readily apparent to me how those concepts apply precisely to the human listening experience. It seems to be accepted that the upper frequency continuous sinewave response limit of the human ear (up to about 20KHz) is the relevant bandwidth limit. I have to accept on blind faith that that is all that is relevant and sufficient for the human listening experience.



First, my impression is that understanding the maths behind DSP is really the only way to *truly* understand what's going on (which I do not claim I do). As you see some aspects of DSP really are counterintuitive on their face....like the 'few samples at high frequencies' thing.

Second, every attempt so far to argue for the physiological need for higher sample rates in order to produce realistic audio, founders at the blind test stage. Thus proponents have to resort to arguments like: it's a hypersonic effect that is only detectable by brain imaging! (though the 'effect curiously seems to last much longer than the stimulus, and requires custom made playback gear) or, some musical instruments have lots of energy above 20kHz! (and some visible light sources have lots of energy in the UV or infrared ranges...so?) or , what about bone conduction?! (what about it? it's a vibration effect that requires the source to be very close to the body). The only argument with any solid foundation is: 44.1 puts the onus on engineers to make their brickwall filters very good indeed, or to use oversampling, because at 44.1 the cutoff frequency (22.05) is so close to the audible limit. So shoddy implementation at recording or playback could lead to audible artifacts.

You don't have to accept on 'blind faith' that the ear's passband for sounds transmitted through air extends 'only' up to the mid-20's at very best, these numbers weren't pulled from thin air, there is a scientific literature on psychoacoustics and the physiology of audition dating back a century.

QUOTE
The absence of such uploads to me significantly weakens the credibility of those who claim superiority of 96KHz per se


Well, no kidding! biggrin.gif I don't know where you get the impression that '96 kHz per se is superior' is the consensus on HA.org. I'd say it's quite the opposite. Of course, once we travel beyond the confines of 'the village' here, and out into the woods of other 'audiophile' forums, then we start to see claims that have more foundation in belief than evidence. wink.gif

This post has been edited by krabapple: May 21 2008, 17:02
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MLXXX
post May 21 2008, 17:23
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QUOTE (krabapple @ May 22 2008, 01:40) *
You don't have to accept on 'blind faith' that the ear's passband for sounds transmitted through air extends 'only' up to the mid-20's at very best, these numbers weren't pulled from thin air, there is a scientific literature on psychoacoustics and the physiology of audition dating back a century.

Thx for your various comments krabapple. On this particular aspect, the point I was trying to make is that although it can be said based on decades of testing that the human ear has a bandwidth of around 20KHz when tested with continuous tones, I am obliged to accept on blind faith that that is all that is required as the bandwidth of a digital reconstruction process.

To many people, the two bandwidths are equivalent and no further analysis is necessary.

I am hesitant as real life audio sources can start and stop abruptly and asynchronously. We have a perception of the direction of a sound source as well as its pitch and tonal quality. All of this strikes me as very complex. It is not clear to me (but has to be accepted with blind faith) that because our ears can only hear a continuous tone up to about 20KHz, a bandwidth of 20KHz in the electronics is sufficient for recording and reproducing music.
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2Bdecided
post May 22 2008, 11:49
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QUOTE (MLXXX @ May 21 2008, 17:23) *
Thx for your various comments krabapple. On this particular aspect, the point I was trying to make is that although it can be said based on decades of testing that the human ear has a bandwidth of around 20KHz when tested with continuous tones, I am obliged to accept on blind faith that that is all that is required as the bandwidth of a digital reconstruction process.

To many people, the two bandwidths are equivalent and no further analysis is necessary.

I am hesitant as real life audio sources can start and stop abruptly and asynchronously. We have a perception of the direction of a sound source as well as its pitch and tonal quality. All of this strikes me as very complex. It is not clear to me (but has to be accepted with blind faith) that because our ears can only hear a continuous tone up to about 20KHz, a bandwidth of 20KHz in the electronics is sufficient for recording and reproducing music.
You're doing it again - you're assuming that, in the entire history of psychoacoustics, no one tried low pass filtering impulses to check the limit that way; no one tried manipulating interaural level, time, and frequency to determine the effects; and no one tried recording audio at high bandwidth, and checked for the audibility of various low pass filters.

krabapple put it succinctly...
QUOTE
The 'it strikes me as complex' is close to an argument from personal incredulity. and the answer is: more reading about how digital audio *works*.

...though you're moving into psychoacoustics now. A very fascinating field. Try these:

http://www.amazon.co.uk/Introduction-Psych...e/dp/0125056281
(the £5 link is a bargain!)
http://www.amazon.co.uk/Psychoacoustics-Mo...595/ref=ed_oe_h
http://www.amazon.co.uk/Hearing-Handbook-P...2980&sr=1-1

These are probably superseded by more modern publications - it's 10 years since I read them.

Finally, this is far away from what you're looking for, but it's so great at explaining the in-band limits of human hearing wrt audio coding that I had to include it...

http://www.ece.rochester.edu/~gsharma/SPS_...AudioCoding.pdf

Cheers,
David.

P.S. EDIT: Sampling Theory
http://groups.google.com/group/comp.dsp/ms...hl=en&fwc=1
Forget the maths if you want - the conclusions themselves are interesting. You'll find some of them quoted in this thread. Doubting them is about as useful as doubting that 2+2=4.

This post has been edited by 2Bdecided: May 22 2008, 12:29
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MLXXX
post May 22 2008, 15:24
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QUOTE (cabbagerat @ May 21 2008, 17:00) *
QUOTE (MLXXX @ May 20 2008, 14:51) *

However that might still be a significant result if playback equipment generally available cannot play back well at 48KHz, despite theory indicating 48KHz should be sufficient.
A more likely conclusion is that the equipment works fine at 48kHz, and adds additional distortion when playing material with content above 20kHz. While it would be difficult to rate which one works "better", good recordings of the sounds as played will answer the question of which is more accurate. It wouldn't surprise me if the 48kHz version were more accurate.

Thanks Cabbagerat. It seems that even if a particular file did seem to sound better when played at one sample rate than another, using a particular playback chain, there could be any number of possible reasons for that outcome.

QUOTE (2Bdecided @ May 22 2008, 20:49) *
You're doing it again - you're assuming that, in the entire history of psychoacoustics, no one tried low pass filtering impulses to check the limit that way; no one tried manipulating interaural level, time, and frequency to determine the effects; and no one tried recording audio at high bandwidth, and checked for the audibility of various low pass filters.

2B, I have for years assumed such tests would have been done.

_____________________


Perhaps this thread has reached a natural end, unless there are any actual audio clips at around 96KHz or more that have been identified (and can be linked to, or uploaded ) that appear to sound better at the higher sampling rate than when downsampled to 48Khz. Though if anyone has the courage to identify such a clip, they should be ready for their claim to be challenged!

Thanks again for the various helpful comments,
MLXXX

This post has been edited by MLXXX: May 22 2008, 15:32
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Martel
post May 23 2008, 08:08
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QUOTE (MLXXX @ May 22 2008, 06:24) *
Perhaps this thread has reached a natural end, unless there are any actual audio clips at around 96KHz or more that have been identified (and can be linked to, or uploaded ) that appear to sound better at the higher sampling rate than when downsampled to 48Khz. Though if anyone has the courage to identify such a clip, they should be ready for their claim to be challenged!

Man, this is not about courage, this would be about breaking the human body limits! laugh.gif
And, please, rule out the sampling rate from your considerations, a 96kHz waveform lowpassed at 24 kHz bears exactly the same information as the same waveform downsampled to 48 kHz (if downsampled ideally).
If I were you, I would first "investigate" the possibility of identifying a 24 kHz lowpass filter. Try and start a new thread. laugh.gif


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MLXXX
post May 25 2008, 16:10
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QUOTE (Martel @ May 23 2008, 17:08) *
If I were you, I would first "investigate" the possibility of identifying a 24 kHz lowpass filter. Try and start a new thread.
Martel, I've already spent a lot of time in this thread and what I am about to write is relevant to what has gone before.

1st test:
File 1: Audition 3 used to generate a tone of 8333Hz at -20dB @ 192KHz sample rate (single channel).
File 2: Audition 3 used to generate a third harmonic of 8333Hz at -20dB (i.e. a tone at 24999Hz)@ 192KHz.
File 3 created (single channel): file 1 + file 2, @ 192KHz

When file 1 and file 3 were attempted to be ABXd a problem arose as the tweeter in trying to handle the 24999Hz tone was not able to reproduce the 8333Hz at full amplitude.

[With a microphone at 1m from the tweeter and using an oscilliscope connected to the output of the analogue mixer, the peak to peak voltage was slightly less when playing file 3 compared with file 1. The waveform shape was different as well.]

After temporarily reducing the amplitude of file 1 by a small amount, the files still sounded different when an ABX was attempted.

However I was concerned that the tweeter might be creating spurious effects, so I changed the experimental setup.

2nd test:
Stereo file A created with file 1 (8333Hz) as the left channel and file 2 (24999Hz) as the right channel.
Stereo file B created with file 1 (8333Hz) as the left channel and zero signal for the right channel.

Playback volume of the left speaker was tested with the microphone 1 metre in front of the tweeter and feeding the oscilloscope. Amplitude of the waveform from the left speaker remained constant whether or not the right channel was playing, i.e. whether file A or file B was played.

At a reasonable listening distance, A and B sounded different (file A seemed louder and a little richer).

I was concerned that the separation of the speakers was so great it was creating a sound field full of peaks and troughs. The wavelength of 8333Hz is only a little over 4 centimetres.

3rd test:
In the interest of science, I moved the front left and front right home theatre speaker enclosures so their sides were touching, and played files A and B in an endless loop.

Even at a distance of 8 meters on axis from the speakers there were very noticeable nodes in the sound field. As in test 2, file A seemed louder and little richer. This was clearcut. (However it was important not to move as the loop played.)

As a type of control, I created a file 3L, which had the contents of file 3 in the right channel, and nothing in the left channel. When this was played, the sound field was full of nodes. This was to be expected. Our living room is not an anechoic chamber.

Also, with the speakers adjacent, I positioned the microphone about 1.5 metres away and observed the oscilloscope. The waveform was not perfect but it was very different to a sine wave when one speaker was reproducing the 8333Hz tone and the other the 3rd harmonic.

Conclusions:

Although the third harmonic of a tone at 8333Hz cannot be heard when played by itself (i.e. as a tone of 24999Hz) by adult human beings, it can have an impact on the human listening experience, when the fundamental frequency is also being reproduced by a loudspeaker system in a home environment.

If the 24999Hz tone is absent, the listening experience can be different. Subjectively (for me) it is slightly less rich. Also I found that when the harmonic was present, I perceived the pitch as sounding slighter sharper if my ears were fresh, but flatter if I had been ABXing for a while. [This certainly didn't assist the ABX process!]

The effect was subtle.

Some audio cannot be downsampled to 44.1KHz, or even 48KHz, without affecting the perceived sound.

Equipment used:

Software: Audition 3, Cooledit 2, foobar
AVR driven from PC with coaxial SPDIF at 192Khz.
Medium price hi-fi speakers [Magnat "Vintage 350", rated 20Hz - 35KHz]
Rode NT1-A microphone
Behringer analogue mixer
Dated oscilloscope

*******************

As it is late, I will not attempt to upload any of the test files. They are quite easy to generate using cooledit or audition, anyway. [Edit: Stereo test files are now at post #105.]

I imagine that these results are no surprise to many readers, but will surprise some others.

How this type of experiment relates to the proposition that an audio bandwidth of around 20KHz is sufficient for the human listening experience I will leave to others to comment on, if they so wish.

I note that by sending the third harmonic through a separate amplifier, I avoided the issue of intermodulation distortion in the amplifier and the speakers [though not any possible IMD in my own hearing]. I listened at what I'd term a 'moderate' level, certainly not a loud level for listening to music. The 24999Hz waveform when displayed on the oscilloscope looked quite smooth (a sinusoid) when only it was being played. Similarly when only the 8333Hz waveform was played, there was a smooth sinusoid. However when the combined waveform was played through one speaker [or when played with two adjacent speakers each taking a separate frequency], the shape of the waveform altered on the oscilliscope, and the quality of the sound changed slightly for my ears.


ABX results:

foo_abx 1.3.1 report
foobar2000 v0.9.5.1
2008/05/26 00:20:21

File A: \\star8\shareddocs\sineplus3rdharmonic\8333inleft&8333_3rdharmonicinright@192.wav
File B: \\star8\shareddocs\sineplus3rdharmonic\8333inleftnothinginright@192.wav

00:20:21 : Test started.
00:20:40 : 01/01 50.0%
00:20:52 : 02/02 25.0%
00:23:49 : 03/03 12.5%
00:43:17 : 04/04 6.3%
00:43:57 : 05/05 3.1%
00:44:12 : Test finished.

----------
Total: 5/5 (3.1%)


This post has been edited by MLXXX: May 27 2008, 12:20
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Posts in this topic
- MLXXX   Resampling down to 44.1KHz   May 6 2008, 12:07
- - Martel   QUOTE (MLXXX @ May 6 2008, 03:07) is 44.1...   May 7 2008, 10:13
|- - MLXXX   Martel, thx for your comments. QUOTE (Martel ...   May 7 2008, 16:55
|- - 2Bdecided   QUOTE (MLXXX @ May 7 2008, 16:55) B. If c...   May 7 2008, 17:18
||- - MLXXX   QUOTE (2Bdecided @ May 8 2008, 02:18) [No...   May 7 2008, 17:29
|- - lvqcl   QUOTE (MLXXX @ May 7 2008, 19:55) A. If ...   May 7 2008, 18:09
|- - Glenn Gundlach   QUOTE (lvqcl @ May 7 2008, 09:09) QUOTE (...   May 10 2008, 05:01
|- - krabapple   I think what he meant was there is no comparison d...   May 10 2008, 06:00
|- - lvqcl   QUOTE (krabapple @ May 10 2008, 09:00) I ...   May 10 2008, 09:46
- - cabbagerat   QUOTE (MLXXX @ May 7 2008, 07:55) A. If ...   May 7 2008, 17:15
- - MLXXX   Thanks Ivqcl. The graphs look fine. However I wo...   May 7 2008, 19:09
- - KikeG   The original triangle file has a problem, and it i...   May 7 2008, 19:48
|- - MLXXX   Kike, Thanks for preparing and uploading these fil...   May 8 2008, 10:38
|- - MLXXX   QUOTE (KikeG @ May 8 2008, 04:48) Edit: t...   May 9 2008, 18:54
|- - KikeG   QUOTE would that be a basis for concluding that if...   May 9 2008, 18:59
- - MLXXX   Will be pleased to do that Kike, but may be 24 hou...   May 7 2008, 20:00
- - AndyH-ha   I have no suggestions about what the Current Creat...   May 7 2008, 20:04
- - MLXXX   This really must be my last post for now! My ...   May 7 2008, 20:23
|- - Martel   QUOTE (MLXXX @ May 7 2008, 11:23) This re...   May 7 2008, 21:32
|- - MLXXX   QUOTE (Martel @ May 8 2008, 06:32) [You p...   May 8 2008, 15:11
- - AndyH-ha   It sounds like you are confusing resampling with o...   May 7 2008, 22:10
|- - SebastianG   QUOTE (AndyH-ha @ May 7 2008, 23:10)...   May 8 2008, 10:52
- - cabbagerat   I have uploaded some files to test, MLXXX, which y...   May 8 2008, 18:18
- - KikeG   MLXXX, I think there is something wrong going on w...   May 8 2008, 18:18
|- - Alex B   QUOTE (KikeG @ May 8 2008, 20:18) MLXXX, ...   May 8 2008, 19:43
|- - Kees de Visser   QUOTE (Alex B @ May 8 2008, 19:43) Someth...   May 8 2008, 22:13
||- - Alex B   QUOTE (Kees de Visser @ May 9 2008, 00:13...   May 8 2008, 22:52
||- - 2Bdecided   QUOTE (Alex B @ May 8 2008, 22:52) I loca...   May 9 2008, 10:19
||- - MLXXX   QUOTE (2Bdecided @ May 9 2008, 19:19) So ...   May 20 2008, 23:51
||- - 2Bdecided   QUOTE (MLXXX @ May 20 2008, 23:51) QUOTE ...   May 21 2008, 11:43
|- - MLXXX   QUOTE (Alex B @ May 9 2008, 04:43) My Ter...   May 8 2008, 23:10
- - SoleBastard   QUOTE (MLXXX @ May 8 2008, 15:11) Indeed,...   May 8 2008, 18:26
- - cabbagerat   For your ABXing pleasure - versions reduced by 3dB...   May 9 2008, 08:46
|- - MLXXX   Cabbagerat, thanks for the first set of uploads. ...   May 9 2008, 13:25
- - Martel   My guess is, if you meet all the "perfection...   May 9 2008, 10:40
- - AndyH-ha   It isn't unusual for the signal level to go ab...   May 9 2008, 12:34
- - cabbagerat   QUOTE (MLXXX @ May 9 2008, 04:25) Cabbage...   May 9 2008, 14:00
- - MLXXX   Well that noise sample of yours cabbagerat is very...   May 9 2008, 14:17
|- - MLXXX   QUOTE (MLXXX @ May 9 2008, 23:17) 2. Kike...   May 10 2008, 06:01
|- - krabapple   QUOTE (MLXXX @ May 10 2008, 01:01) I have...   May 10 2008, 06:40
|- - MLXXX   QUOTE (krabapple @ May 10 2008, 15:40) Yo...   May 10 2008, 07:04
|- - Martel   QUOTE (MLXXX @ May 9 2008, 22:04) krabapp...   May 10 2008, 14:31
- - Alex B   Here is the difference signal sample that I used i...   May 9 2008, 14:31
- - KikeG   I have identified the lowpasses of the files I pos...   May 9 2008, 17:45
- - AndyH-ha   Some CD players and stand-alone DACs "upsampl...   May 9 2008, 21:22
- - MLXXX   My results were:Audigy 4 module: a click was audib...   May 11 2008, 01:15
|- - Martel   QUOTE (MLXXX @ May 10 2008, 16:15) My res...   May 11 2008, 08:37
- - cabbagerat   QUOTE (Martel @ May 10 2008, 23:37) As fa...   May 11 2008, 08:46
- - lvqcl   MLXXX, can you record this 'click' with, s...   May 11 2008, 09:04
|- - MLXXX   As we do not appear to have any posters with exten...   May 11 2008, 09:56
|- - 2Bdecided   QUOTE (MLXXX @ May 11 2008, 09:56) This i...   May 12 2008, 10:12
|- - Martel   QUOTE (MLXXX @ May 11 2008, 00:56) SoleBa...   May 13 2008, 14:17
- - MLXXX   [Martel, I have not tried to find out exactly how ...   May 14 2008, 14:34
|- - pdq   QUOTE (MLXXX @ May 14 2008, 09:34) Put an...   May 14 2008, 15:36
|- - MLXXX   2Bdecided, I have looked through the FAQs but ther...   May 14 2008, 15:52
|- - 2Bdecided   QUOTE (MLXXX @ May 14 2008, 15:52) 2Bdeci...   May 14 2008, 16:17
|- - MLXXX   Thanks cabbagerat; your specific explanations in r...   May 15 2008, 02:51
|- - Martel   QUOTE (MLXXX @ May 14 2008, 17:51) But I ...   May 15 2008, 09:46
||- - Kees de Visser   QUOTE (Martel @ May 15 2008, 09:46) 96kHz...   May 15 2008, 11:37
||- - Martel   QUOTE (Kees de Visser @ May 15 2008, 02:3...   May 16 2008, 08:48
|- - Kees de Visser   QUOTE (MLXXX @ May 15 2008, 02:51) If we ...   May 15 2008, 10:55
|- - 2Bdecided   QUOTE (MLXXX @ May 15 2008, 02:51) I coul...   May 15 2008, 11:34
- - 2Bdecided   To me, it sounds like you still haven't read t...   May 14 2008, 14:54
|- - 2Bdecided   QUOTE (2Bdecided @ May 14 2008, 14:54) To...   May 20 2008, 12:55
- - cabbagerat   2Bdecided is right - you need to do some backgroun...   May 14 2008, 15:57
- - MLXXX   QUOTE (cabbagerat @ May 15 2008, 00:57) N...   May 18 2008, 16:29
- - Martel   A 44 kHz digital waveform PERFECTLY describes ANY ...   May 18 2008, 19:25
|- - MLXXX   QUOTE (Martel @ May 19 2008, 04:25) A 44 ...   May 19 2008, 18:11
|- - Martel   QUOTE (MLXXX @ May 19 2008, 09:11) I do g...   May 20 2008, 08:38
|- - SebastianG   QUOTE (MLXXX @ May 19 2008, 19:11) I do g...   May 21 2008, 10:14
|- - MLXXX   SebastianG, thanks for taking the time to restate ...   May 21 2008, 10:28
|- - SebastianG   QUOTE (MLXXX @ May 21 2008, 11:28) At thi...   May 21 2008, 11:32
|- - krabapple   QUOTE (MLXXX @ May 21 2008, 05:28) Sebast...   May 21 2008, 16:40
|- - MLXXX   QUOTE (krabapple @ May 22 2008, 01:40) Yo...   May 21 2008, 17:23
|- - krabapple   QUOTE (MLXXX @ May 21 2008, 12:23) QUOTE ...   May 22 2008, 05:16
|- - 2Bdecided   QUOTE (MLXXX @ May 21 2008, 17:23) Thx fo...   May 22 2008, 11:49
|- - MLXXX   QUOTE (cabbagerat @ May 21 2008, 17:00) Q...   May 22 2008, 15:24
|- - Martel   QUOTE (MLXXX @ May 22 2008, 06:24) Perhap...   May 23 2008, 08:08
|- - MLXXX   QUOTE (Martel @ May 23 2008, 17:08) If I ...   May 25 2008, 16:10
- - cabbagerat   QUOTE (MLXXX @ May 18 2008, 07:29) I...   May 19 2008, 08:33
|- - MLXXX   Rereading the post leaves me with the same impress...   May 19 2008, 12:06
- - pdq   Let me see if I can provide an analog-domain equiv...   May 19 2008, 16:10
|- - 2Bdecided   QUOTE (pdq @ May 19 2008, 16:10) Let me s...   May 19 2008, 16:56
- - greynol   Key word here is product. Simply summing two sign...   May 19 2008, 17:06
- - pdq   I could be wrong about this, but I thought that wh...   May 19 2008, 17:08
- - greynol   You have to multiply the two signals or subject th...   May 19 2008, 17:18
- - pdq   Sorry, post corrected.   May 19 2008, 17:25
- - greynol   Reconstruction using a sinc pulse at every sample ...   May 19 2008, 18:25
- - cabbagerat   QUOTE (MLXXX @ May 20 2008, 14:51) Howeve...   May 21 2008, 08:00
|- - MLXXX   This thread started with the downsampling to 44.1K...   May 21 2008, 09:46
- - greynol   If we're talking sound cards, the problem is w...   May 21 2008, 08:43
- - pdq   @MLXXX: You seem to want someone to prove to you t...   May 21 2008, 12:37
- - MLXXX   2B, yes I can understand that some of my posts may...   May 21 2008, 12:58
|- - Martel   QUOTE (MLXXX @ May 21 2008, 03:58) 2B, ye...   May 21 2008, 13:36
|- - Kees de Visser   QUOTE (MLXXX @ May 21 2008, 12:58) If no-...   May 21 2008, 13:53
||- - MLXXX   QUOTE (Kees de Visser @ May 21 2008, 22:5...   May 21 2008, 14:52
|||- - krabapple   QUOTE (MLXXX @ May 21 2008, 09:52) QUOTE ...   May 21 2008, 17:08
||- - 2Bdecided   QUOTE (Kees de Visser @ May 21 2008, 13:5...   May 21 2008, 15:19
|- - pdq   QUOTE (MLXXX @ May 21 2008, 07:58) It see...   May 21 2008, 13:57
- - sld   Don't you think that it is futile to try to fo...   May 21 2008, 18:27
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