On Mon, Feb 16, 2009 at 09:27:18PM +0100, Arnold Krille wrote: > Are they really upsampling? > > I was under the impression that they use one fast converter (far faster then > the sampling rate you hear) and the (de-)multiplex the signal to the various > channels. Should be easier than syncing several converter clocks. AFAIK nobody is doing that today. Some of the very early Sony converters for recording digital audio on video tape used to do this, compensating for the half sample delay on the second channel by the same trick on playback. Just consider this: a very good digital antialiasing filter operating at 8 * Fs (i.e. 8 * upsampling) costs almost nothing if integrated on the DA chip. An analog one for 44.1 or 48 kHz would require precision components, very careful board design, and probably manual alignment of each individual board. The choice is easy to make... > The devices I work with on the other hand go exactly > the other way, they combine several 1GS/s adc's to > have one 4GS/s adc. But they face their own set of > clock-sync problems... I imagine they have !! The fastest one I was involved with was 1G, it drove most of the design team crazy, and required a collection of dirty tricks to get it interfaced to an FPGA, but that was five years ago... Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia O tu, che porte, correndo si ? E guerra e morte ! _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user
