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Simply by changing at random, single samples. Selecting a single sample and entering RND (65536) into the numeric equation for each channel. That's how jitter will be received by the other end. Bren R.




Randomly changing parts of the data stream isn't the same thing as jitter (Although if you damaged every single sample and no one could tell then they aren't going to notice jitter either). That would be simulating read errors and lost data. Jitter is defined as a time based instability. This instability is going to occur on every word from every single sample and the shift is going to be random (sometimes faster sometimes slower.) Your not losing and interpolating the data, your just not reconstructing the sound waves with the same timing (which will equate to shape) they were encoded with. An extreme example of this affect would be to take a turntable that allows fine tuning of the rotation speed and to keep making random small adjustments to the speed.

This is the reason all DAC's require an oscillator crystal. The crystal establishes the time sync with the audio samples for reproduction of the analog sound wave from the digital sample. When transfering the digital samples over spdif, the clock sync is not transmitted and that is where the jitter is introduced.

What really matters of course is if you can you hear the difference. These fluctuations are probably so small that you wouldn't notice, not to mention since the timing instability is random it's not like you would hear the same imperfections (assuming you heared any) if you played the music over and over.