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.
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.
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.
A 96KHz extractI 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.
I agree that a headphone test would be interesting.
There will no doubt be arguments that any differences are due to the playback equipment and indeed that may be so.
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
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.
So you say that:1) If you play back just the 25kHz sinus, you hear nothing2) If you play the 8333 Hz sinus in addition, you hear something3) 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.
It doesn't work for me, but maybe it'll work for someone.
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).
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.
% 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=150000c=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 valueswavespace=int16(totalsamples/60)tranche2=int16(wavespace/3)tranche3=int16(wavespace*2/3)offset=0for e=1:wavespace -2if (e==tranche2) offset=1 elseif (e==tranche3) offset=0; endiff=(60 * e)+offsetfor h=1:60c(f+h,1)=d(h);endforendfor%The next section adds fadeins and fadeouts.for z=1:totalsamplesv=c(z,1);% To disable the optional sine wave, add a percentage sign at the start of the next linev=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; endforwavwrite('1samplewidthat96KHz---10.42microseconds---clickswithrepetitionrateof1600Hz---Optionalsinewaveat1595Hz.wav',c,96000,16)disp('***Completed***')
For those who like a graphical presentation, here is how Audition displays the files after the conversion to 48KHz:
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.
[...] I am now looking at another aspect: time resolution affecting the downsampling process.
With difficulty I could even detect a difference of advancement or retardation of one sampling interval at 96Khz, i.e. 10.42?S.'Aha!', I thought. Now I will be able to create a test to demonstrate that 48KHz is inadequate.
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.)
[...] 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).
[...] (no dither) [...]
There is a wavering quality because of the inclusion of a low level sine wave at approximately 1595Hz.
I find the sine wave acts as a reference level for the ear making it easier to pick that the middle burst is different.
You will probably find that with file 2, the middle burst is softer and has a different tonal quality.
Quote from: MLXXX on 09 June, 2008, 03:45:37 AM[...] 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.
... 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. ...
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.
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 from: MLXXX on 09 June, 2008, 03:45:37 AMThere is a wavering quality because of the inclusion of a low level sine wave at approximately 1595Hz.What is "wavering quality"?
Quote from: MLXXX on 09 June, 2008, 03:45:37 AMYou will probably find that with file 2, the middle burst is softer and has a different tonal quality.What would your explanation be?
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.
I downloaded a trial of iZotope RX but wasn't able see how to activate an advanced reconstruction display.
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.
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.
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/