appar111
Headphoneus Supremus
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- Jun 4, 2002
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I'm a complete newbie when it comes to DAC's. What's the difference between a DAc that can do 24/192 and one that can do 24/96 khz? I don't even know what those terms mean
Originally Posted by mofonyx /img/forum/go_quote.gif of course there is. |
Originally Posted by Dave_M /img/forum/go_quote.gif Higher sampling rate is better because on the output of a DAC chip is usually a low pass filter to make the square wave output into smooth analog waveforms. This filter should ideally be a brick wall filter, but that is not possible to make so the filter will have a steady roll off. If the sample rate is low, then the filter might start rolling off at the top end of the audible frequency range. |
Originally Posted by appar111 /img/forum/go_quote.gif What's the difference between a DAc that can do 24/192 and one that can do 24/96 khz? I don't even know what those terms mean |
Originally Posted by Jon L /img/forum/go_quote.gif This completely depends on what you exactly mean by "do 24/192." Many modern DAC chips are spec'd to be capable of decoding 24/192, but this really doesn't give you any benefit unless: A) Your song/track is actually 24/192, hopefully recorded in 24/192 not just upsampled, AND B) Spdif receiver of your DAC can actually accept 24/192. Many spdif receivers can't pass 24/192; most can pass up to 96kHz only. When you are playing redbook 16/44.1 material, all the parts above only passes 16/44.1, so it's a moot point sonically. Yes, you can manipulate 16/44.1 data by increasing word-length to 24 bit and/or upsample to 88.2/96/176.4/192, but no new data can be created. As far as whether there's a sonic difference between 24/96 and 24/192 tracks when the tracks were natively recorded in that format and decoded unmolested, the answer is a resounding YES! I've had opportunities to A-B same tracks recorded in 24/96 vs. 24/192 natively in studio (not commercially available), and the improvements are quite significant with 192. However, even properly recorded 24/96 tracks simply Blow away 16/44.1 hands down. |
Originally Posted by Garbz /img/forum/go_quote.gif That's exactly what oversampling fixes. |
Originally Posted by Fryguy8 /img/forum/go_quote.gif Nyquist's theorem says this is impossible, unless I'm misinterpreting. Can you please elaborate on this and educate me? I'm under the impression that nyquist's theorem basically proves that anything greater than 44.1 is useless for audio playback. |