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In Reply to: RE: No revolutions posted by Dave_K on April 11, 2016 at 15:42:31
Dave thanks for the post. I found the whole thing incredibly informative.
Does the very high sampling/oversampling rate in PCM R2R dacs allow for a low order gentle filter like DSD?
Do you think there is anything intrinsically more linear about one format versus the other? I mean in reproducing a waveform, or does it just come down to the sound of filters?
Thanks,
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Big speakers and little amps blew my mind!
Follow Ups:
Does the very high sampling/oversampling rate in PCM R2R dacs allow for a low order gentle filter like DSD?
No for the digital filter, yes for the analog filter.
There is a gentle low order analog filter on the output of an oversampling DAC. It can be gentle because it's not the reconstruction filter. All it's doing is filtering ultrasonic noise. With some R2R DACs, you might not even need it.
But the digital reconstruction filter will still be a brick wall or close to it. Oversampling at higher rates and with longer filters allows for a more ideal filter response. It also allows for better noise shaping. But you're still stuck trying to fit a steep cut-off between 20-22 KHz regardless of how fast you oversample. Such is the nature of Redbook unfortunately, and there is no way around using brick wall filters without accepting aliasing distortion or rolling off below 20 KHz.
Same thing if you're converting 44.1 KHz PCM to DSD, the conversion will involve a brick wall filter.
If you're sourcing DSD from DSD or analog or high sample rate PCM, then you can avoid it.
Do you think there is anything intrinsically more linear about one format versus the other? I mean in reproducing a waveform, or does it just come down to the sound of filters?
Are you comparing typical oversampled Redbook playback vs. Redbook upsampled to DSD? Or R2R vs. delta-sigma? Or more generally PCM vs. DSD?
Thanks for the response.
Hi res PCM vs DSD.
One tenth of an octave (20 to 22kHz) would require one hell of a brickwall filter. JA mentioned in another post that DSD provides an improvement over PCM in time domain errors. I suppose that could be called a form of linearity. Nelson Pass published a paper that has some graphs showing how steep filters for speakers obliterate the phase coherence of a speaker. My limited knowledge leads me to think that phase and time domain response are probably closely tied? Your opinion? I have used both gentle and extreme slopes with speakers, and neither is perfect or perfectly terrible. Just different with different tradeoffs.
Thanks again for answering my basic questions.
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Big speakers and little amps blew my mind!
I suppose he is referring to the ringing impulse responses of the anti-aliasing and reconstruction filters, which is particularly bad for Redbook due to the need for steep filter responses. For higher sample rate PCM, 88.2 KHz and above, the filters do not need to be steep. And even if the filters are made to be steep, there isn't much musical content up at 44 KHz or above to incite ringing, and the ringing would be ultrasonic anyway. So in my opinion, it is impossible to make Redbook anti-aliasing and reconstructions filters truly transparent, but it should be easy to do that for PCM at 88.2 KHz and above.
I enjoy both DSD and hi-res PCM, but I think DSD 64 (the SACD format) is a little bit compromised relative to hi-res PCM. Going forward, I could be happy with either hi-res PCM or higher rate DSD. I think that a DSD-only DAC operating at 256fs could be superior to hi-res PCM due to simpler implementation, but at the expense of storage space and download bandwidth.
I found the following Audiostream interview with Mr. T. Of special interest was the following;
"If we convert from 24-Bit at 352.8kHz (DXD-PCM) to 1-bit at 2.822MHz (DSD) - we need to throw away around 99.96% of the amplitude information the PCM format is capable of, while we are only having 12.5% of the time domain information that the DSD system is capable of. If we convert to DSD from DXD, that is 1-bit at 2.822MHz to 24-Bit at 352.8kHz - we need to throw away 87.5% of the time domain information of DSD, though we can theoretically remap all of this into the amplitude domain. So in effect we get the worst of both formats, rather than the best of one."
Maybe higher rate DSD would even things some, but I find it interesting. Also very few albums for sale at 352kHz. I would bet Sony thinks they can remap into the amplitude domain, or they wouldn't have put it forward as an archival format, but then again maybe they would. I have read on a forum for recording engineers where they say DSD was meant to be converted to PCM, but PCM was not meant to be converted to DSD. They certainly seem to do it though.
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Big speakers and little amps blew my mind!
Thorsten's numbers are misleading because he's equating timing resolution with sample rate and amplitude resolution with sample size. I believe he knows better but is just trying to oversimplify for a non-technical audience.
Ultimately, amplitude resolution and time resolution are characteristics of the encoded waveform, not its encoding format. Amplitude and time resolution are limited by the digital noise spectra. So for example, 1-bit DSD has better amplitude resolution than 16-bit PCM, and 352.4 KHz 24-bit PCM certainly has better timing resolution than DSD (even 192 KHz PCM probably does).
Using delta-sigma modulation with appropriate noise shaping, one can trade sample size vs. sample rate without loss of resolution. That's how delta-sigma DACs work.
Very informative reading here.
nt
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