On Sun, 15 Feb 2009 02:44:10 -0700 "Bearcat M. Şandor" <hometheater(a)feline-soul.com> wrote: I'm not sure but it sounds like the following problem: Formats like flac, ogg vorbis and probably others support more than two channels but pretty much no-one bothered to implement support for more than two channels since must music only uses two channels. I don't know if the problem lies in the application or the underlying libraries most use. My guess is that video players are more likely to work, so I'd try with xine, mplayer, gstreamer and the likes and see if one of them works. The only way to get this fixed is to file bug reports or feature requests and contact the developers directly to discuss the issue at hand, else no-one will bother to implement it. It's a similar problem with video players and jack support, pretty much all support jack but in a manner that is it pretty much useless. No-one will fix a problem he is not aware of. Best regards, Philipp

On Sun, 15 Feb 2009 18:50:12 +0800 Ray Rashif <schivmeister(a)gmail.com> wrote: I haven't tried but I somewhat doubt that 24bit is the problem, my guess is that it's only the channel count that's the problem for most players. Philipp

On Sun, 15 Feb 2009 18:45:58 -0700 "Bearcat M. Şandor" <hometheater(a)feline-soul.com> wrote: Glad it works for you. But: Upsampling can't make it better(=closer to the original), you can't magically get stuff that's not there (even if companies like creative claim so). If it sound better now than before something was wrong before or you like some resampling artifacts ;) Best regards Philipp

On Monday 16 February 2009 09:05:08 Sampo Savolainen wrote: ok, so i'm confused. If you're saying that upsampling is generally a bad thing why does one pay extra for it in a product like this: http://www.meridian.co.uk/product-model/g-series/g082-upsampling-compact-di… player.aspx ? -- Bearcat M. Şandor Bearcat(a)feline-soul.net Jabber: bearcat(a)feline-soul.net MSN: bearcatsandor(a)hotmail.com Yahoo: bearcatsandor AIM: bearcatmsandor

On Mon, 2009-02-16 at 10:50 -0700, Bearcat M. Şandor wrote: Say you have a CD with nothing but 74 minutes of "perfect" 440Hz sine wave. As CDs are, the data is at 44.1k, which means that it can contain only frequencies below 44.1k/2 (< 22050Hz). If you upsample this sine wave to 96k, you will have 96k PCM data that contains a "perfect" sine wave at 440Hz. You've added nothing to the signal so nothing has happened. .. except for the fact that the sample rate conversion will cause distortion, noise and phase issues. There are really good algorithms (libsamplerate) or obscenely good ones (check out izotope's scores on http://src.infinitewave.ca/ ). But nevertheless none of them are perfect and none of them cad add more "sound" to the original. You can expand this naive example to any sound contained on a CD. The CD will always contain only frequencies below 22050Hz, any upsampled version will only have frequencies below 22050Hz however high of a sample rate you choose. Except for the distortion caused by the algorithm which will contain frequencies above 22050Hz. Good marketing? I find it hard to believe that upsampling PCM data does anything that a proper DAC doesn't. Many DACs use techniques to give the same benefits what the meridian web page describes as the benefit of upsampling: "filtering to take place far beyond the range of human hearing". People who know more about DACs can pitch in and provide more precise information if needed. Let's ask the question the other way around: if upsampling is a superior way to convert from digital to analog, why do we see upsampling being mentioned only in obscenely expensive hi-fi CD players? Why Lynx, Apogee, RME, etc. don't use this technology in their highly praised converters? Sampo

On Mon, 2009-02-16 at 20:19 +0100, Fons Adriaensen wrote: That was the point I was trying to make in the previous paragraph from the one you quoted. The point here was that it seems a lot of companies making high quality converters don't feel upsampling is worth mentioning. Sampo

On Mon, Feb 16, 2009 at 09:27:18PM +0100, Arnold Krille wrote: 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... 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 !

On Tue, Feb 17, 2009 at 08:13:02AM +0100, Arnold Krille wrote: Yep, 10 MHz is the standard for reference frequency distribution for lab instruments. Getting decent GHz clocks from that requires careful PLL design. Returning to the original matter: multiplexing AD/DA do exist of course, and they are the best solution if you require lots of low bandwidth channels. One reason you won't find many of them in pro audio is the channel isolation requirement (> 100 dB) - it would be very hard to make a fast analog multiplexer reaching anything near that figure. Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia O tu, che porte, correndo si ? E guerra e morte !