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["flash/jitterator-schem1.swf", "jscheme1", 660, 160, "Setup #1: Digital Slave Mode, No Oversampling, No Reclocking", "What you see above is the most simple digital configuration possible. All Jitter which is present at the S/PDIF input is contributing to distortion of the audio signal at the analogue output. This is the typical scenario when a standard external DAC is used in conjunction with a CD player. This scenario is called Digital Slave Mode because the DAC receives its clock signal from the external CD player's oscillator.
To view another digital routing solution, click Setup 2"]
,["flash/jitterator-schem2.swf", "jscheme2", 660, 160, "Setup #2: Digital Slave Mode, 8x Oversampling, No Reclocking", "Oversampling means that the frequency of the original clock signal is multiplied by a whole number (i.e. 2, 4, 8, etc.). The idea is that when the converter chip runs at this higher frequency, the low-pass filter after the converter chip can be a simple second-order filter, as opposed to a sixth or higher order filter needed in the case of no oversampling. This leads to less distortion, which effect is easily audible and is not to be doubted in any way. Based on the Shannon Kotelnikov sampling theorem, the low-pass filtering should ideally cut off the first frequency which is above the Nyquist frequency. Because analogue filtering causes the least amount of distortion when the slope is soft (i.e. first or second order which means 6 or 12 dB per octave, respectively), oversampling helps achieve the conditions where such a filter can be used to achieve the desired attenuation of the first frequency which is above the Nyquist Frequency.
All Sigma/Delta type converters (the kind we don't like) contain Oversampling (whole-number multiple) built in as an integral part of the chip. The PCM 1704, in contrast, is a parallel resistor type converter which requires any Oversampling to be done externally by another chip dedicated to this process. This is more expensive, is more laborious to build, and sounds better."]
,["flash/jitterator-schem3.swf", "jscheme3", 660, 237, "Setup #3: Digital Slave Mode, 8x Oversampling, Asynchronous Reclocking", "This method is used to attempt to reduce the Jitter present in Method #2 above. In this example, Jitter is indeed reduced, because the new oscillator is located right next to the digital signal it is newly clocking. This avoids new introduction of line-induced Jitter. However, we are now dealing with two Master Clocks simultaneously. Part of the problem here is that no two clocks ever tick at the exact same pace. Another part of the problem is that the further clock signal is always more Jittered than the nearer clock signal. Therefore, we encounter an inevitable frequency discrepancy between these two clock signals, and this leads to distortion of the audio signal due to the real-time digital interpolation algorithm employed. Even though Asynchronous Reclocking does not bring about an actual sampling rate conversion, (which would then be called Asynchronous Resampling), the method of Asynchronous Reclocking cannot actively remove all present Jitter artifacts from the original data stream. Indeed, by using Asynchronous Reclocking (two separate clocks), some amount of existing Jitter is actually fed into the newly clocked data stream."]
,["flash/jitterator-schem4.swf", "jscheme4", 660, 237, "Setup #4: Digital Master Mode, 8x Oversampling, Single-stage Synchronous Reclocking", "Here there is only one clock source, and this is located right next to the converter. The source device accepts the same clock as the converter. Not only at the same frequency, but the same clock signal itself. Everything is kept in perfect synchronization, and line-induced Jitter which appears on the long path between the oscillator, through to the source device, then back to the S/PDIF input, is quantized to perfection via the synchronously running oversampling chip. This is called Master Mode because the clock source is within the DAC, and the device contains only this one clock source. This is the method we recommend using, because it is the method which annihilates Jitter the most effectively. The addition of multiple stages of this type of completely Synchronous Reclocking lessens Jitter even more. By appropriately applying the completely Synchronous Clocking and multiple-staged Reclocking solution offered by LessLoss, no digital compromises are made and a Jittered signal coming off of the laser mechanism of a CD player is improved upon in real time as it reaches the DAC chip."]
,["flash/jitterator-schem5.swf", "jscheme5", 660, 237, "Setup #5: Digital Slave Mode, Upsampling and Asynchronous Reclocking Together", "Asynchronous Reclocking With Upsampling is the method used when you want to achieve a frequency other than a whole-number multiple of the S/PDIF input frequency. (i.e. from 44.1 kHz input we want a 48 kHz output (multiple of 1.0884353741496598639455782312925), or from 44.1 we want 96 kHz (multiple of 2.176870748299319727891156462585). This is also known as Sampling Rate Conversion. It is important to note that in Asynchronous Reclocking With Upsampling, a separate clock is used with a different operating frequency than that of the original clock.
Method #5, shown above, is the best possible solution for those who wish not to modify an existing Master CD player (original CD-internal clock remains, running the player). The market is full of such solutions. Although the Jitter reduction using this method is very exceptional, the user will still encounter the traditional audible differences between digital cables, cable lengths, and jittery playback devices. Using Method #5, even with the highest quality digital cabling, it is impossible to achieve the sound quality which is achieved using Method #4 above (Multiple-staged Synchronous Reclocking)."]
,["flash/jitterator-schem6.swf", "jscheme6", 660, 237, "Setup #6: Digital Master Mode, Upsampling and Synchronous Reclocking Together", "The standard frequencies used which allow simple Oversampling (x2, x4, x8) of 44.1 kHz audio are the following:- 11.2896 MHz
- 16.9344 MHz
- 33.8688 MHz
This is so because the numbers 11289600, 16934400, and 33868800 all divide evenly by 44100, which is the typical sampling frequency used in the consumer S/PDIF standard. Because the standard oscillator in DVD players runs at either 27 MHz or 54 MHz (for video purposes), a Phase Locked Loop (PLL) is used within universal DVD/CD players to achieve the 44.1 kHz needed for consumer CD audio. If we want to slave a universal DVD/CD/SACD player to the DAC 2004, the DAC must contain the 27 or 54 MHz generator. Because we are on a 27 or 54 MHz clock, we must therefore use Asynchronous Upsampling, because 27 or 54 MHz cannot be devided evenly by 44.1 kHz. This renders the question of highest possible quality audio from DVD players moot.
In this case, at best, we have a high-quality clock upgrade to a DVD player, coupled with a quite effective Jitter suppressing solution.
The creators of the DVD playback devices sacrificed high quality audio for a frequency compatibility with their video schematic solution (it was probably cheaper that way).
And this is the very reason why, using our method (Setup #4), a higher quality sound is achieved by playing CDs at 44.1 kHz than is possible if we were to offer this mathematically inferior DVD solution at even 96 kHz. Many audiophiles and the more honest dealers tend to agree that well engineered CD playback solutions sound just as good as DVD or SACD. Our solution goes to the extreme and it puts the CD format in the forefront of the new format disc playback solutions. Until higher resolution formats can be played back in both a mathematically synchronous way and through parallel-resistor type converters (PCM audio instead of bitstream), the CD remains the best digital source out there.
Interestingly, the mass-market, in its scramble for attention, has prematurely moved on and has never utilized the CD format to its full potential. Haste makes waste."]
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