I have a emagic emi 2|6 DAC (usb, 24/96).
I have tried mad 0.14 2b for 24 bit decoding and I have put the DAC to 96khz, in order to force an oversampling at 96 khz.
The sound seems really better for me.
Is it possible or I'm dreaming?
Oversampling adds nothing to the sound, if properly done. If not properly done, it can degrade the sound. So, unless your DAC works better at 96 KHz than at other sample rates, audible improvements are not due to the actual sound played, but other things.
With 96, the high frequencies should be more natural because you have more points to define it.
With a sweep sound, i can hear higher with 96 and at the end of the sweep in can hear a noise (descending sweep) with 44.1.
In the other hand I have more continuous noise with 96, maybe it's because of an higher frequencie filter in 96, in this case you'r right, it's linked to my DAC.
But what about the theoric precison in high frequencies?
With 96, the high frequencies should be more natural because you have more points to define it.
Well, that's not right, theorically, according to Fourier (or was it Nyquist? I don't remember well...) sampling theorem, there should be no difference, and in fact there's no difference, in good quality equipment and using good quality oversampling (resampling)...
With a sweep sound, i can hear higher with 96 and at the end of the sweep in can hear a noise (descending sweep) with 44.1.
...but it seems that your DAC performance at 44.1 KHz is not as good as at 96 KHz in this respect.
The points that define the sound are in the original file, often 44100 Hz. Any converter, exept some rare crazy-audiophile prototypes, resamples the signal for the conversion. often 64 or 256 times, so that the "points that define the sound" are usually at 64x44100=2,822,400 Hz or 256x44100=11,289,600 Hz.
But they are just artificial points computed from the original 44,100 ones, they don't add any definition to the sound that was not in the original.
Also, if there are descending tones in the 44100 Hz sweep, then the converter is not running at 44,100 Hz ! Descending tones are an alias (an artifact) caused by a bad resampling.
Edit : important points : where does the sweep come from, and how is it played (program/playback settings) ?
Also, if there are descending tones in the 44100 Hz sweep, then the converter is not running at 44,100 Hz ! Descending tones are an alias (an artifact) caused by a bad resampling.
No, the descending sweep is without resampling (44.1->44.1). With 96 (resample), i don't hear it...
Edit : important points : where does the sweep come from, and how is it played (program/playback settings) ?
http://ff123.net/samples/sweep.zip (http://ff123.net/samples/sweep.zip)
WinXP
Winamp 2.81
Nullsoft Waveform Decorder v 2.06
Directsound 2.2.6
i just played the sweep and found that the only way that i can render it correctly is if the sample rate in ssrc is set to 48000 Hz or greater in winamp 2.81.
Well, that's not right, theorically, according to Fourier (or was it Nyquist? I don't remember well...) sampling theorem, there should be no difference
it's the Nyquist-Shannon theorem. when you sample a signal that has been band-limited to fs/2 hz with fs samples per second, you are storing all frequencies up to fs/2 with 100% accuracy.
No, the descending sweep is without resampling (44.1->44.1). With 96 (resample), i don't hear it...
Uncanny...
No problem at 96 kHz, OK...
But an alias at 44.1... The Eamgic 2/6 doesn't look like the kind of soundcard that resamples... would the master clock be on "external" ?... no, wait, it must be Windows DirectSound ! Select Winamp WaveOut instead of DirectSound, and in the WaveOut configuration, don't select the wave mapper as device, but your Emagic itself... This should solve the problem... it would be very interesting to know if DirectSound affects the sound like this.
uhm, i *highly doubt* if selecting "wave mapper"/"primary sound driver"/whatever instead of the device itself does anything, "wave mapper" is provided just to redirect software that doesn't have user interface to select device to the "preferred" device selected in control panel.
i think you people should try toggling "allow hardware acceleration" on device tab in latest out_ds, it seems to affect a lot of weird things; when hardware acceleration is disabled, software mixing is used instead of hardware (which is default for most of directsound apps out there); software mixing seems to "emulate" waveOut (at least on win2k/xp).
Well, that's not right, theorically, according to Fourier (or was it Nyquist? I don't remember well...) sampling theorem, there should be no difference
it's the Nyquist-Shannon theorem. when you sample a signal that has been band-limited to fs/2 hz with fs samples per second, you are storing all frequencies up to fs/2 with 100% accuracy.
that can't be right... if you offset by some phase (0 < phi < pi/2) the amplitude decreases. for instance, if you sample a wave at fs/2 that's pi/4 out of phase, you have zero amplitude.
But an alias at 44.1... The Eamgic 2/6 doesn't look like the kind of soundcard that resamples... would the master clock be on "external" ?...
Internal clock
no, wait, it must be Windows DirectSound ! Select Winamp WaveOut instead of DirectSound, and in the WaveOut configuration, don't select the wave mapper as device, but your Emagic itself... This should solve the problem... it would be very interesting to know if DirectSound affects the sound like this.
DirectSound or WaveOut, no matter...
I have try 44.1 and 96 on several tracks and I really hear the difference...
when you sample a signal that has been band-limited to fs/2 hz with fs samples per second, you are storing all frequencies up to fs/2 with 100% accuracy.
that can't be right... if you offset by some phase (0 < phi < pi/2) the amplitude decreases. for instance, if you sample a wave at fs/2 that's pi/4 out of phase, you have zero amplitude.
The amplitude of the samples decrease, yes, but the samples are not a wave. They still contain all the information from the original wave, including its original amplitude.
It can be seen in CoolEdit : phase-shift a high frequency wave so that the amplitude of the sample decrease. Cool Edit will still draw the original full amplitude wave between the decreased samples. It is mathematically reconstructed. I can post a picture when I'm back home.
You example at fs/2 is right. It is not possible to record fs/2. Only frequencies strictly inferior to fs/2.
Maybe it's not the frequency itself but how to reproduce it?
You example at fs/2 is right. It is not possible to record fs/2. Only frequencies strictly inferior to fs/2.
No, it's *sometimes* possible to get fs/2 depending on the phase - a signal alternating between +1 and -1 is definitely fs/2 - and if you amplitude modulate any signal with this +1/-1 alternating signal, you "invert" the spectrum as you'd expect (so signals at 0.1fs get shifted to 0.4fs, etc.).
Interesting thread, I wanna add something
I have a DMX 6fire which is capable of 24/96. When I set SSRC to resample from 44.1kHz 16bit to 96kHz 24bit the soundcard driver automatically changes its internal master clock to the resampled frequency and the sound becomes slightly higher or the highs are a little bit more present than the lows. I'm rather sure that the sound changes but I don't know why. Any explanation?
But an alias at 44.1... The Eamgic 2/6 doesn't look like the kind of soundcard that resamples... would the master clock be on "external" ?...
Internal clock
no, wait, it must be Windows DirectSound ! Select Winamp WaveOut instead of DirectSound, and in the WaveOut configuration, don't select the wave mapper as device, but your Emagic itself... This should solve the problem... it would be very interesting to know if DirectSound affects the sound like this.
DirectSound or WaveOut, no matter...
I have try 44.1 and 96 on several tracks and I really hear the difference...
marcan: how did you set the internal clock? 44.1 or 96? and do you resample via ssrc to 96 or not?
how did you set the internal clock? 44.1 or 96? and do you resample via ssrc to 96 or not?
I can change the clock with the Emagic Device Options.
I have tried with out_ds and out_ds_ssrc. With out_ds, the oversampling is done by Windows XP (with High Performance Option). With out_ds_ssrc, the oversampling is done by out_ds_ssrc. Personnaly, I prefer with out_ds.
No, it's *sometimes* possible to get fs/2 depending on the phase - a signal alternating between +1 and -1 is definitely fs/2
No, the longer the signal alternates between -1 and +1, the closer to fs/2 it is.
To be at fs/2 exactly, it must alternate for an infinite amount of time. This is purely mathematical, though.
In practice, there is no point to distinguish between 22,050 Hz and 22,049.9999.
On the Audiophile 24/96, I can measure an improvement when upsampling from 44.1kHz to 96kHz (using Cool Edit Pro) - but I'm not convinced I can hear a difference - maybe - but I've not done a blind test yet!
For example, if you play a 21kHz tone sampled at 44.1kHz, you get a quiet tone output at 44.1-21=23.1kHz (the first alias). If you resample to 96kHz (well, 88.2 is more sensible) first, then the higher quality resampling in CEP removes this alias completely. It gives poorer time-domain resolution though. Whatever - I can't hear this high!
Cheers,
David.
It gives poorer time-domain resolution though.
What does? 88.2 kHz has better time-domain resolution than 44.1 kHz.
Definitely, with a blind test, I can hear a difference, particularly in high frequencies!
Definitely, with a blind test, I can hear a difference, particularly in high frequencies!
Then I'm technically correct if I call you a son of a bi....
But Marcan said that there were low frequencies (descending sweep) at the end of the sweep file.
I have a descending sweep (not low frequencies) with 44.1, not with 96.
You example at fs/2 is right. It is not possible to record fs/2. Only frequencies strictly inferior to fs/2.
No, it's *sometimes* possible to get fs/2 depending on the phase - a signal alternating between +1 and -1 is definitely fs/2 - and if you amplitude modulate any signal with this +1/-1 alternating signal, you "invert" the spectrum as you'd expect (so signals at 0.1fs get shifted to 0.4fs, etc.).
sorry for staying so off-topic, but....
yeah, i see what you're saying, but that wasn't my issue. my problem was that, given a signal at fs/2, you couldn't ever tell what the actual amplitude of the signal was, since you'd be lacking the phase information. as pio pointed out, my problem was that i was only looking at the fs/2 frequency. at anything else, you get beats---amplitude of the beats is close to the amplitude of the signal (it may be equal, but my intuition about discrete signals is rather poor), and the frequency of the beats is the difference between the sampled frequency and the sampling frequency, unless i'm terribly mistaken...
so here's my next question: how do you tell the difference between a signal with an oscillating amplitude at fs/2 and one with constant amplitude at some other frequency. or is there no difference?
how do you tell the difference between a signal with an oscillating amplitude at fs/2 and one with constant amplitude at some other frequency. or is there no difference?
The difference is that the oscillating signal has some frequencies above Nyquist's limit, while the constant one has not.
In fact, the mathematical concept of frequency is a bit different from what the intuition suggests. A given frequency is a sinewave that never starts and never stops, and never changes its amplitude.
If we modulate the amplitude of a sinewave, we get a new wave. Its mathematic "period" is changed. It repeats itself only after the amplitude has been completely changed.
Say that we modulate a 22050 Hz wave in a sinusoidal way, one time per second. The new wave is a function of time that repeats itself each second. Therefore its fundamental frequency is 1 Hz, and the harmonics are 2,3,4,5 etc Hz. In fact nearly all its harmonics have an amplitude of zero.
Only the 22049 and 22051 Hz harmonics have an amplitude different of zero.
Thus 22049+22051 Hz is the same as 22050 Hz + amplitude modulation each second. They are two different ways of writing the function of time that the wave is. But only the first one describes it in sum of pure frequencies. The second one describes an altered frequency.
Then the sampling theory comes into play. A sampled signal such as this one can be generated by an infinity of different waves. For example, 22049+22051+88200 Hz. The 88200 component will not affect the sampling at 44100 Hz, the result will be the same as 22049 + 22051. In the same way, the 22051 Hz component only affects the amplitude of the result. A single 22049 Hz sine gives exactly the same shape when sampled at 44100 Hz.
That's the meaning of "complete description" of frequencies under 22050 Hz. Frequencies above are only partially described, because different frequencies can give the same sampled result. I think also that to get a complete description, the original signal must not have any frequency above 22050 Hz. Then, there is a unique way of reconstructing the original wave from the sampled one.
In our example, the reconstruction is a steady 22049 Hz sine. I can't be an oscillating 22050 Hz one, nor even, say, an oscillating 22049.9 Hz one, because this have frequency components above 22050 Hz, and this possibility has been discarded when the 44100 Hz sampling rate was chosen.
yeah... thanks for the excellent, very detailed response. i just realized it was a stupid question, though...
Thanks too for the very detailed answer, but why I (and the others persons) hear the difference?
Let me guess, Marcan, you and Neck are using some late SB soundcards?
The reason you hear those downward tones are becase they are false tones
being created by the resampling process that some cards have to do, since they
do not operate at 44.1Khz.
SB Live and Audigy1 and some others are like this, they run at 48 Khz internally,
so they have to resample up to 48.. and they don't do it very cleanly.
If you run at 48Khz to begin with, you don't go thru the process, and don't have
the errors..
Hope it helps..
Jon
Thanks Jon.
My DAC is an Emagic emi2|6 (professional). I can choose the internal conversion rate (44.1, 48, 96).
We can easily hear the difference on professional equipment (Genelec, Yamaha NS10, Quested, ...).
Marcan, burn this sweep on a CD and try it in a CD player. I don't understand how your soundcard work !
Pio, the sweep sound isn’t really important. When we hear a track, we can easily hear a difference between a 44.1 and 96 khz conversion. The high frequencies are really better.
I think the difference is because the anti-alising filter is higher and because the high frequencies are reproduced more naturally.
It gives poorer time-domain resolution though.
What does? 88.2 kHz has better time-domain resolution than 44.1 kHz.
Yes. But if you take a 44.1kHz sampled signal, and let the DAC do the reampling, then a more gentle filter
may be used, which would give better time domain resolution. The resampling in Cool Edit, set to a reasonably "high quality" setting will give excellent frequency response, but poor temporal accuracy. This is the "theoretical" best result, but in practice a compromise between time and frequency domain accuracy may sound better.
(Even this statement is misleading - the theoretical best resampling which Cool Edit almost manages is "perfect" in both time and frequency domain, within the limits of the system. Unfortunately, these limits may give infinite ringing at 22.05kHz - that's what a compromise may be better!)
Cheers,
David.
http://www.David.Robinson.org/ (http://www.David.Robinson.org/)
Pio, the sweep sound isn’t really important. When we hear a track, we can easily hear a difference between a 44.1 and 96 khz conversion. The high frequencies are really better.
I think the difference is because the anti-alising filter is higher and because the high frequencies are reproduced more naturally.
If the audible high frequencies are better sampled at 96k than 44.1k, then there's something wrong at 44.1kHz!
But isn't this the whole point - we live in the real, imperfect world. 44.1kHz should be enough - but with real world (even very good) equipment, 96kHz sounds better.
It doesn't prove that Fourier or Nyquist were wrong, or that we all have Bat-like hearing. It just shows nothing is perfect.
Maybe!
D.
If the audible high frequencies are better sampled at 96k than 44.1k, then there's something wrong at 44.1kHz!
If information > 22 KHz is really inaudible, then there has to be something really wrong in any system to sound so different from 96 KHz. Marcan, As Pio suggests, try burning the sweep and play it in a good cd player.
But isn't this the whole point - we live in the real, imperfect world. 44.1kHz should be enough - but with real world (even very good) equipment, 96kHz sounds better.
I don't think this can be considered the general case, certainly not with good equipment. I'd suggest to try some ABX tests with 44.1 KHz data vs. 88.2 or 96 KHz resampled data (SSRC, CoolEdit, others?) to see if this happens with anyone's setup. I'd say that I can't hear a difference, but, to be honest, I haven't tried seriously yet.
As for true 24/96 signals sounding better than 16/44.1 ones, anyone can try http://www.pcabx-pro.com/technical/sample_...rates/index.htm (http://www.pcabx-pro.com/technical/sample_rates/index.htm)
I think that there is no need to ABX. Descending tones are characteristic of resampling. It doesn't mean that the DAC works better at 96 kHz, it means that it doesn't really operate at 44.1 kHz, there is something tampered with. The most probable is that the DAC is in fact 96 kHz fixed, and when a 44.1 kHz frequency is set, a sample rate converter processes all the data from 44.1 to 96 kHz.
There is no descending tone at all in the sweep when it is really played at 44.1 kHz. A treble tone is rising, and quickly disappears in complete silence. That's what is played in a pure 44.1 kHz converter, like in a CD player. I've just checked with the Marian Marc 2 soundcard (advertised for digital in/out without dither nor resampling), digital out, the DAT deck outside detects properly a 44.1 kHz SPDIF incoming stream. As an example, the SPDIF output of the SB live is detected at 48 kHz no matter what.
Given this, you certainly have a better sound at 96 kHz. The point is that you should get as good a sound burning the files on CD, and playing them in a CD player, at 44.1 kHz.
Hmmm... some people have mentioned the fact that Nyquist proved that sampling at Fmax*2 is "enough". Well, I disagree. Being just a senior in Electrical Engineering Technology I can't really disprove the guy, and I am sure in theory he is correct.
In practice tho, I think he is not. Two examples. One is the digitizing scopes we use in lab. At the high end of the frequency range (I think 500MHz for the one's I am using this semester) they are taking 2Gsamples per second. That is 4x the max freq. I asked my professor why, if Nyquist blah blah and he said that for all practical purposes you NEED 4X to get an accurate representation. I mean, real world systems.
Another one is that in my digital class, we had a really simple A/D and a D/A system. We put in a sine wave in on one side, goes thru the A/D, into a shifter, thru a serial cable, into another shifter and out the D/A. Anyway, don't remember at what frequency we were running, but the output was distorted way before we got to the 2X spot.
So I really think that sampling at 96kHz is a great idea. But then again, I can't HEAR any artifacts or anything related to sampling at 44.1. Who knows... just wanted to share
Thanks LoKi128,
So I'm not dreaming...
That is 4x the max freq. I asked my professor why, if Nyquist blah blah and he said that for all practical purposes you NEED 4X to get an accurate representation. I mean, real world systems.
Maybe for getting an accurate *graphical* representation. But with the use of a reconstruction filter, which any proper DAC has, this is unnecesary, as Pio and others have explained for Nth time.
Hey, just do some tests with Cool Edit!! It is capable of drawing *perfectly* any 21 KHz waveform at 44.1 KHz, because it simulates graphically this reconstruction filter. (
Edit: Well, it seems that Cool Edit is not so "perfect" at doing this... Anyway with a proper reconstruction filter, there's no need for higher sampling rates to properly reproduce high frequencies.)
But then, there's one difference between theory and real world, and is that, in practice, it is very difficult to have perfect response up to fs/2, that is up, to 22050 Hz in case of cd. But, there is no problem in getting nearly perfect response up to 20 KHz and even 21 KHz with today's DACs.
Edit: With the use of external software upsamplers such as SSRC, you simply get closer to the ideal case (theory), instead of relying on the more "approximate" oversampling that all DACs perform nowadays. However, as 2bdecided says, this more ideal case has other effects that can cause problems in real world, such as long ringing at the limit upper frequency.
So I'm not dreaming...
Nobody is saying you're dreaming. Again, if you hear descending tones playing the 44.1 KHz sweep, then there are obvious flaws in your DAC, because this doesn't happen on mine. Maybe it lacks a proper reconstrucion filter that filters at 22 KHz, I've seen this happens with my 96 KHz capable DVD player. In my case this causes high amplitude aliases over 22 KHz, that in you case could intermodulate with the original sweep tones, leading do descending products that fall into the audible range. Or maybe there is just some bad resampling-like process.
I think that there is no need to ABX.
I was refering to 96 KHz sounding better in general, not for this particular (obvious?) case.
problems in real world, such as long ringing at the limit upper frequency.
What kind of problems long ringing should cause ? Intermodulation ?
problems in real world, such as long ringing at the limit upper frequency.
What kind of problems long ringing should cause ? Intermodulation ?
Yes, but I guess it is a *possible* problem, depending mostly on the equipment. But I really don't know how important is it in practice. Maybe in some cases it is, maybe it is not most of the times. I have simulated the effect into Cool Edit Pro, and with some induced distortion and synthetic signals (impulses and sines) it is quite noticeable. But I haven't done any tests with real music and real equipment, so I don't know.