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New DVD-Audio music releases and talk about the latest players.

Yep,

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32 bit floating point PCM DSP's with good algorithm's can give very good results. After the math is applied the signals much be rounded back into 24 bits and dither is used to linearize the result.
The cost for this is 2 bits from the 24bits available.
The signal degradation is insignificant if all is done right.

There is no such thing as native DSD DSP's.
After a processing step DSD always ends up as pcm. To get back to a 1 bit stream dithering is needed wich needs at least needs a 3 bit PCM sample otherwise the signal will be compromized to much.
Too much processing steps with dsd and signal degradation becomes noticeable very quickly. This includes ALL processing steps in the complete chain. The ill effects of dsd editing during mastering and the steps applied in the player, for BM filtering or even level adjustments in the digital domain, are cumulative.

Another problem is signal headroom. If two or more signals are added the dsd signal can saturate. They strongly advise to keep the signal level below -6dB of the full input range to prevent trouble.
PCM processing doesn't have this problem, a 24 bit signal handled by a 32 bit processor has plenty of processing offset.

There are solutions for simple crossfades in dsd 'dsp' implemented.
The output is switched between the native dsd stream and the processed stream during the crossfade and if that's finished the signal is switched to the other native dsd stream. During this switching offset clicks at -50dB signal level are produced.
According two the philips engineer who presented this solution these clicks are insignificant.

Frank



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