Anti-jitter RAM buffer in DACs, Which have one ? Options

[Pio2001]

In, http://www.tc.umn.edu/~erick205/Papers/pap...html#principles we can read about RAM buffers :

This process is referred to as timebase correction and as stated before, any quality piece of equipment will implement it.

That has been the ritual answer to jitter concerns since I'm on the Internet.

But... if "any quality piece of equipment will implement it", then some non-quality pieces of equipment don't implement it. My question is : which ones ?

The statement "quality piece of equipment" is completely subjective and doesn't give any information. For my grandfather, a "quality" CD reader will be a portable radio-cassette-CD-speakers combo for 100 $ in the supermarket. The "non-quality" one will be the Fisher-Price toy for children from 3 to 7.
For an audiophile friend, the "quality" CD player will be the 25,000 $ Wadia CD Transport+DAC, while any player under 1,500 $ is just worthless crap.

So, do a 200 $ CD Player always have a RAM buffer ? Or is it not considered a quality piece of equipment ? What about a 300 $ CD Player from 1990 ?

[2Bdecided]

Thoughts?

Well, I've only one further thought on this: how can people buy this product when it has such a fishy website?

What, the one I posted? www.jitter.de? Oh the answer's simple - they'll buy it because it's had a good review in a Hi-Fi magazine.

And everything on the site seems to be scientifically valid.


I just had an interesting email from Frank Klemm about ASRCs in movie playback. He says... (I'll remove it if it turns out that I wasn't supposed to forward this here!)

The sync engine has the task to display and play the decoded stuff.
There are three problems:

  - the Video Card typically don't run with a multiple of (actually not
    exact known) frame frequency
  - The audio device typically don't run on the exact frequency which is
    setup
  - There are displacements between the time calculated from the
    samplerate * samplecount and the time stamps.


Sources of this errors are:

  - sound card and video card are derived from different quartz oscillators
  - noone really cares about exact frequncies, a 100 Hz frame rate can be
    99.253536 as well as 101.8378626 Hz, a 48000 Hz can be a 48662.06896 Hz
    (SB 128 PCI)
  - rounding and accumulated errors due to time quantization (typically
milliseconds)
  - errors in A/V streams
  - the typically view "It nearly works, why should I improve it"
  - the typically "100 implementations and every is a little bit different"


Typically the video display engine has a simple job and only waits until the
time of a video timestamp is reached, then wait again until the CRT video
frame is finished and then switch to the next image.

The audio engine typically uses some called a "asyncronous sample rate
converter" or "Software PLL" which converts the audio stream to the sample
rate of the output device. This includes:

  - estimation of the *real* sample rate of the output device
  - estimation of the *real* sample rate of the input stream
  - very smoothly adjusting small difference changes between these two frequencies
  - adjusting medium difference changes between these two frequencies
  - immediately adjusting large difference changes between these two frequencies

This is typically done by a control signal which is derived from the samples
in the sample fifo buffer. The sample count is normally lowpass filtered by%xact frequncies, a 100 Hz frame rate can be
    99.253536 as well as 101.8378626 Hz, a 48000 Hz can be a 48662.06896 Hz
    (SB 128 PCI)
  - rounding and accumulated errors due to time quantization (typically
milliseconds)
  - errors in A/V streams
  - the typically view "It nearly works, why should I improve it"
  - the typically "100 implementations and every is a little bit different"


Typically the video display engine has a simple job and only waits until the
time of a video timestamp is reached, then wait again until the CRT video
frame is finished and then switch to the next image.

The audio engine typically uses some called a "asyncronous sample rate
converter" or "Software PLL" which converts the audio stream to the sample
rate of the output device. This includes:

  - estimation of the *real* sample rate of the output device
  - estimation of the *real* sample rate of the input stream
  - very smoothly adjusting small difference changes between these two frequencies
  - adjusting medium difference changes between these two frequencies
  - immediately adjusting large difference changes between these two frequencies

This is typically done by a control signal which is derived from the samples
in the sample fifo buffer. The sample count is normally lowpass filtered by
three ... five saturating lowpasses.

    filter_effect            = function ( lowpass_filtered_signal - input_signal)
    lowpass_filtered_signal *= ( 1 - filter_effect )
    lowpass_filtered_signal += filter_effect * input_signal

function (x) may be something like:

    function (x) = ( x*x ) / (1 + x*x)

This suppresses high frequency jitter which may be audible.

--
Frank Klemm

Cheers,
David.