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The importance of Components?

Discussion in 'Sound Science' started by gregorio, Mar 5, 2009.
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  1. gregorio
    Quote:

    Originally Posted by nick_charles /img/forum/go_quote.gif
    I could post some square wave samples I have recorded and the picture is pretty bad even as low as 500hz. It would be interesting to do actual listening tests on generated square waves and recorded ones, maybe when I get some spare time.



    Remember though that digital audio theory (largely the Nyquist-Shannon theorum) only works for sine waves (or complex waveforms made up of sine waves). Pure square waves do not exist in nature and cannot be played back by any speaker or heard by the human ear. If you hear a square wave, it is actually various sine waves modulating together to approximate a square wave, same is true for triangle and saw waves. So, don't be looking for an un affected square wave. BTW, the ear is pretty insensitive in some respects, even an approximated square wave cannot be decerned from a pure sine wave once we get to about 12kHz.

    G
     
  2. olblueyez
    Quote:

    Originally Posted by milkweg /img/forum/go_quote.gif
    You have no point.

    You may think you are doing a service by making wild unfounded claims about audio gear but I see you as doing a disservice because new members give you more credibility than you deserve.




    You must have some serious problems to accuse others of short comings that you obviously have. Is it fun for you to sit and start arguments with everyone? Is that all head-fi is good for to you? I ran into a guy a few days ago who wants to swap me a Telefunken for an Amperex and he is a nice guy. I met another person who was nice enough to send me a message wishing me well after my surgery, once again, really nice person. Someone in my Foobar thread asked for help today and I was thrilled to try and help the guy. I also got to shed some light on some super unintelligent stuff in the cable threads too, it was fun but that is not the only reason most of us are here, certainly not myself. Milkweg, why have you left your home planet and come to invade ours? [​IMG]
     
  3. JaZZ Contributor
    Quote:

    Originally Posted by nick_charles /img/forum/go_quote.gif
    So where is the problem? [​IMG]



    In the solution.
    .
     
  4. gregorio
    Apologies to anyone just joining this thread. There are a few people here on Head-Fi who know nothing about audio whatsoever but seem to take pleasure in hi-jacking threads with false information and then getting into a war of insults with anyone who disagrees with them, just for their own gratification.

    My advice, to make this thread an easier read, whenever you see the user Olblueyez, do what I do and completely ignore it. Eventually he'll get bored and go away. Unfortunately, some of PhilS' posts seem to be going the same way.

    Cheers [​IMG]
     
  5. olblueyez
    Quote:

    Originally Posted by gregorio /img/forum/go_quote.gif
    Apologies to anyone just joining this thread. There are a few people here on Head-Fi who know nothing about audio whatsoever but seem to take pleasure in hi-jacking threads with false information and then getting into a war of insults with anyone who disagrees with them, just for their own gratification.

    My advice, to make this thread an easier read, whenever you see the user Olblueyez, do what I do and completely ignore it. Eventually he'll get bored and go away. Unfortunately, some of PhilS' posts seem to be going the same way.

    Cheers [​IMG]




    I would just say to go back and read from the beginning and make up your own mind. If you do that you will see who stands in what camp.
     
  6. PhilS Contributor
    Quote:

    Originally Posted by gregorio /img/forum/go_quote.gif
    Apologies to anyone just joining this thread. There are a few people here on Head-Fi who know nothing about audio whatsoever but seem to take pleasure in hi-jacking threads with false information and then getting into a war of insults with anyone who disagrees with them, just for their own gratification.



    Yes, it is indeed quite annoying. I hope the thread gets back on track, as I said all the way back in post #77. The initial question was interesting and the responses that are on point are interesting also.

    P.S. Compare:

    "There are a few people here on Head-Fi who know nothing about audio whatsoever . . . "

    and

    "There are a few people here on Head-Fi . . . who seem to take pleasure in . . . getting into a war of insults."

    The irony is just delicious. [​IMG]
     
  7. Pio2001
    I've got some pictures illustrating what's going on with amplitude modulation a bit more, and especially with time smearing, but I''ve not decided yet if I'm goig to post them in this forum or another.
    I'll see tomorrow, if the trolling have stopped.
     
  8. olblueyez
    Quote:

    Originally Posted by Pio2001 /img/forum/go_quote.gif
    I've got some pictures illustrating what's going on with amplitude modulation a bit more, and especially with time smearing, but I''ve not decided yet if I'm goig to post them in this forum or another.
    I'll see tomorrow, if the trolling have stopped.




    I would recommend starting a fresh thread with no baggage to drag you down. I am looking forward to sseeing what you have.
     
  9. moogoob
    I'll go back to the original concept of the thread (re-rail? [​IMG]) and post what I consider the effects of the following to be, ranked from most to least important in my system:
    Note: Joke entries added to distance certain components more realistically. [​IMG]

    1)Phones
    2)Recording quality
    3)Amp
    4)DAC
    5)Beer
    6)Analog cabling
    7)Comfy seat
    8)Mmmm, tasty sandwich!
    9)Digital interconnects
    10)Planetary alignment
    11)Plastering my walls in ERS paper
    12)Power cords
     
  10. 883dave
    1. 50% Music library
    2. 12.5% Transducers
    3. 5% Room treatments...when speaker based
    4. 12.5% Source component...garbage in garbage out
    5. 10% Amplification
    6. 10% Miscellaneous...ie. cables (interconnects and power cords), vibration control, tubes, stands, record cleaners and storage


    Recording quality is mote in my eyes...if a recording is bad, nothing you do will make it better...so just choose not to listen to the bad recordings
     
  11. Pio2001
    Quote:

    Originally Posted by Pio2001 /img/forum/go_quote.gif
    The second part will adress time smearing.



    Before going on with time smearing, I'd like to add one thing about aliasing and amplitude modulation.

    Those who, by chance, read french, should read the presentation made by GBO about oversampling here : homecinema-fr.com • Voir le sujet - FAQ sur le suréchantillonnage

    Very interesting and very well explained. Here is a short part, right on topic :

    Quote:

    [​IMG]

    I'd like you to pay attention to the presentation. I have willingly represented the samples as small isolated circles. Each circle standing for the value of the given sample (coded on 16 signed bits, as should be on a CD). And there is NOTHING between the samples.
    It happens that some software link these points with imaginary straight lines, which is maybe convenient, but is source of important misunderstandings that are, alas, still found in some audiophile websites.



    And this is very true. Look at the same wav file, displayed in Samplitude :

    [​IMG]

    And here, in CoolEdit pro :

    [​IMG]

    These are two arbitrary pictural representations of the same file. The second one takes the sampling theory into account, and represents the waveform as it is intended to be.

    But both are incorrect : the real representation of the content of the wav files is just a serial of binary digits !
     
  12. Pio2001
    Quote:

    Originally Posted by JaZZ /img/forum/go_quote.gif
    A resonance doesn't just affect the narrow band with maximum intensity, but extends quite largely to both sides. A simple, descriptive example may show you how impossible your scenario is: Let's take some 11-kHz tone bursts, let them pass a sharp 22-kHz low-pass filter. That's one octave below the filter resonance. The 11-kHz tone burst will not stop immediately, unlike it would with full bandwidth, but show significant delay of decay, with a few additional (11-kHz!) cycles before complete silence. That's in the nature of low-pass filtering: Immediate signal stop would call for full bandwidth. Now tell me that's not transient corruption and time-smearing! [​IMG]



    This is a very good idea, and I have done it.

    But I had to find a filter sharp enough to produce significant amounts of ringing. The resampling algorithms of SoundForge did not. Neither its parametric equalizer.
    But Foobar2000 0.8.3's graphic equalizer has very sharp transitions between its frequency bands. Setting the 20 kHz slider to the minimum (-20 dB) on a white noise gives this frequency response (65536-samples FFT analysis) :

    [​IMG]

    Complete attenuation is reached only 30 Hz above the last unaffected frequency !

    Taking a 96 kHz wav with 11 kHz bursts, and equalizing it this way produces ringing, because of the sharpness of the transition in the frequency domain :

    [​IMG]

    Left, the original file, with sharp transition between silence and sound. Right, the lowpassed file. We could expect that it is the same, since everything below 16730 Hz is unaffected, and the file only contains bursts of 11 kHz sines.
    But a burst of 11 kHz sine is not a real 11 kHz sine. The start point and the stop point of the sine are places where the frequency decomposition shows a lot of other frequencies whose effect is to allow the transition between silence and sound.
    These frequencies are affected by the lowpass, and the transition looks different. There is ringing, visible in the right picture.

    If we zoom out horizontally, so as to see the whole bursts, and zoom in vertically, so as to see the smallest amount of ringing, we can see that the ringing extends so far that there is no more silence at all between the bursts. The post-ringing of all bursts joins the pre-ringing of the next :

    [​IMG]

    However, I can't hear the difference. I can hear that there are a lot of other noises than just 11 kHz bleeps, but they sound the same in both versions. They are the clicks that start and stop each piece of sine.

    Let's look more closely at a given pre-ringing :

    [​IMG]

    Here, we can see something very interesting : the oscillations of the ringing seem closer spaced than the ones of the 11 kHz burst !

    Unfortunately, SoundForge is not a practical software to look at this. The zoom is at maximum here. Analyzing frequencies on such short bits of wave is not accurate, but we can ask for the statistics about zero-crossings. First in the burst :

    [​IMG]

    Then in the ringing :

    [​IMG]

    The software shows that the ringing is not an extension of the 11 kHz sine at all, but rather a parasite 16 kHz resonance added.

    The spectrogram view shows it much better. Here is a spectrogram showing three bursts in the original file (256 samples FFT) :

    [​IMG]

    The bursts of 11 kHz sine are represented by the three horizontal red lines. the vertical green and blue bars show the sudden burst of all kind of frequencies that account for the brutal "clicks" at the sharp starting and ending of each sine.

    Now, here is the lowpassed version :

    [​IMG]

    The vertical bars still extend to the top frequencies. Remember that the filter applied is just an equalization of -20 dB, and that it doesn't delete completely high frequencies.

    And the ringing is perfectly visible as a continuous blue line around 16 kHz, which is the transition frequency of the filter (remember the white noise spectrum above).

    We can note two important results :
    -The ringing is completely restricted to the transition frequency of the filter. It consists in a pure sine wave.
    -The sharpness of the frequencies below (and above) the ringing is completely preserved. The verical bars are still there, and the horizontal red lines are not extended the slightest amount, compared to the large duration of the ringing.

    With an antialias filter designed to cut frequencies above 20 kHz, the ringing is also completely restricted to this frequency, and the sharpness of the musical content is completely preserved, in spite of visible oscillations that extends from it in the waveform view.

    We can also see a curious thing that the waveform did not show : the ringing extends not only outside, but also inside the 11 kHz sinewaves.

    I'd like to post another picture by GBO. He did the same experiment, but with a musical sample. It is the recording of a triangle, made at 96 kHz. He applied a brickwall filter at 22 kHz.

    Here is an analysis of the result :

    http://3141592.pio2001.online.fr/pic...nglecomp2D.gif

    Up, the original, down, the filtered version.

    The pre-ringing is clealy visible, does not extends below 20 kHz, and the content below 20 kHz has no time smearing. There is just some noise caused by the processing. Unfortunately, there is no scale on the diagram for us to show the amplitude of this noise.
    There was noise also in my stecrograms, but I purposely set the color scaling just above.

    I've not illustrated it, but I read that ringing is only triggered if there is some content at the filter frequency in the original signal. That's why he used the recording of a triangle. He needed some 22 kHz harmonics in order to cause ringing.
    In our 11 kHz sine, it's the clicks (the vertical bars) that feature the frequency content that triggers the ringing. If our file had already been lowpassed below 16 kHz, the application of the 16 kHz brickwall would have caused no ringing at all (and the absence of time smearing below this frequency would have been obvious).
     
  13. Publius
    You're doing the lord's work Pio. Unless you are not a deist. In which case, the flying spaghetti monster thanks you.

    I would like to point out that high-Q filters (with very small transition bands) can be a very big deal at lower frequencies. In fact, a while ago on HA I was able to ABX the foobar2000 eq in a nulling configuration (-20db a band; +20db same band; ringing remains audible). The band was in the 7khz range - far more audible than 21khz or even 11khz.

    I like to think of transition bands in terms of Heisenberg's Uncertainty Principle, in a sense. To anthropomorphize the situation, the filter needs a certain amount of time to distinguish between two closely-spaced frequencies in the transition band. As the transition band shrinks, the time needed increases. ie, there's a tradeoff between knowing the frequency of a signal and the temporal location of a signal - just like there's a tradeoff between knowledge of position and momentum in quantum physics.

    But none of this actually matters for sample rate arguments because a) nobody has ever proven the audibility of different antialiasing filters and b) there are extremely good theoretical arguments, as Pio has pointed out I believe, for that proof never existing.

    The remaining carrier tone on the AM spectral plot makes more sense given what you're plotting:

    y(t) = (1 + cos(wAM*t)) cos(wt)
     
  14. gregorio
    Quote:

    Originally Posted by Pio2001 /img/forum/go_quote.gif
    And this is very true. Look at the same wav file, displayed in Samplitude :

    [​IMG]

    And here, in CoolEdit pro :

    [​IMG]

    These are two arbitrary pictural representations of the same file. The second one takes the sampling theory into account, and represents the waveform as it is intended to be.

    But both are incorrect : the real representation of the content of the wav files is just a serial of binary digits !




    True, to an extent. Those binary digits are arranged into digital "words", those Words (in 16bit) allow the binary digits to encode roughly 65,000 different values per sample. As far as a graphical representation of a digital file is concerned the Samplitude graphic would be accurate. The CoolEdit graphic is likely to be roughly correct as it is "guessing" how a DAC is going to reconstruct the sampling data back into analogue sound waves.

    BTW, when the reconstruction is done, it's a little more sophisticated than just drawing a line between the dots (sampling points). The values between the sampling points are interpolated with some fancy maths which apply dither algorithms. This results in a perfectly linear (without error) recreation of the original waveform, plus a little un-correlated noise down at the LSB level.

    G
     
  15. JaZZ Contributor
    Pio...

    ...thanks again for you effort. Working through all the data was a fascinating journey for the mind.

    In the meantime I've done my own recordings of tone burst of different high frequencies. Made with Wavelab Lite, at 192 kHz/20 bit (for a closer approach to the analog sound, avoiding the obligatory amplitude modulation with 44.1-kHz sampling). The signals were created from downloaded hearing-test frequencies edited to a series of bursts, then burned onto a CD, played back through a Philips DVD 963SA (upsampling disabled) and finally recorded from its analog output with an E-MU 1212M. The process thus comprises a standard antialiasing/«reconstruction» filter at ~21 kHz. So let's see what the player/DAC has made out of the stored tone-burst signals:

    11 kHz
    [​IMG]

    12 kHz
    [​IMG]

    14 kHz
    [​IMG]

    16 kHz
    [​IMG]

    17 kHz
    [​IMG]

    18 kHz
    [​IMG]

    The signal shapes look surprisingly intact, apart from the inevitable pre- and post-ringing. The latter effectively seems to merely consist of ultrasonic frequencies, foremost the filter frequency. Actually I have expected the sharp start and stop points to make for the dominating effect, at the same time I have expected and hoped to additionally find a visible decay effect with the burst-signal frequency. That is not the case, with none of the samples.

    So my scenario with the filter resonance smoothing the amplitude modulation in fact doesn't seem to apply. Aparently (and astonishingly) it's the aliasing frequency which seems to be «responsible» for the beat and the high-frequency drop-off, so the «reconstruction filter» with its implicated sharp low-pass filter does the trick -- just by suppressing the mirror frequency --, without obvious resonance/time-smearing effects in the audible range.

    I will certainly reserve some reconsiderations on the matter, but as it looks like, the CD format is (near-)perfectly able to store the usable audio signal -- from a theoretical perspective. I would have wished a result more corresponding to my sonic experiences -- which aren't invalidated by these new technical insights, though. And after all the several filter settings on my Symphony make for significant sonic differences, also those with identical frequency response. So the filter ringing must be audible nonetheless, anyhow... And certainly the fact that the SACD format sounds fundamentally different from redbook CD (not by much, but still...) doesn't really allow me to change my mind. When I was about to buy the UDP-1, the salesman demonstrated the two formats and wanted to test if I could identify the SACD layer. I could. And it wasn't the only occasion.

    Well then... Thanks for an interesting, fruitful debate nonetheless! [​IMG]
    .
    vbattach15170.jpg
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