Re: [SlimDevices: Audiophiles] Re: Upsampling
PhilNYC wrote: ...but if you are changing the sample rate of the data, you will need two clocks, no? One clock runs at the original sample-rate (44.1khz) and the second runs at the new sample rate (96khz). or one that runs at 44.1*48 and select the proper signal samples off a common clock. There was a time when 44.1kHz was a challenge. By the time SACD came out, 2.82 MHz was not a challenge. At least if you are not trying to use a Tube clock :-) So having one clock is not a big deal. But changing from 44.1 to 48 or 96 is a really bad idea, IMHO. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
PhilNYC wrote: Pat Farrell;131091 Wrote: Nyquist showed that sampling at twice the bandwidth allows reconstruction. That is why the RedBook spec uses 44.1 kHz. For decades, the hfi world used a bandwidth of 20 hz to 20kHz as the limits of human hearing. Sampling at 44.1kHz allows a little over. My understanding of the benefit of oversampling/upsampling is primarily to get the digital artifacts resulting from imprecisions in the DAC process to a higher frequency so that they can be more easily filtered in a frequency range that won't impact the audible range. I'm not an audio design engineer, so I could be wrong. But I understand it exactly the opposite of this. Digital processing has to have analog filters to cut out unwanted signals and noise. If you use a 44.1kHz sample, you need a radical filter to cut off signals about 20kHz. The standard implementation uses 12dB/octave or even 18dB / octave filters. These do evil things to phase. So if you over/re/up-sample at 96kHz or 192kHz, you can use digital filters (IIR, etc.) for the worst parts, and then use gentle single order analog filters down in the 20-20kHz range. One technical problem with the SACD spec was that it used noise shaping to move the inevitable noise into relatively low frequencies (50kHz, and up). which had potentential to have audible interactions. Someone smarter than me can probably shed some light. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
reeve_mike wrote: Sorry if it disappoints but I know well the work of Claude Shannon and Harry Nyquist ... Opps, sorry. What I was trying to contribute was that the samples on the CD do not faithfully represent the musical waveform, they only represent a version of it filtered at 20.5KHz, which seemed to be relevant at the time but now I can't remember why ... I would not expect it to be 20.5kHz. The sample rate is 44.1, so in theory, you could have signals as high as 22.05kHz. Realistically, there is nearly nothing above 20kHz to start. Not only is 20-20kHz the standard spec but most microphones have serious roll off above 17kHz or so. And all of the preamps used for the microphones, especially the 'vintage' ones that people swear sound best. Neve, SSL, etc. And analog signals don't stop at clean numbers like 20.5, they just roll off at X dB per octave. As an aside, just because 20Hz-20KHz has been used for years doesn't make it 'right', there is increasing psycho-acoustic experimental data that suggests that even though pure tones above approx. 20KHz cannot be heard directly their presence in music signals can be 'detected' in some way - as pointed out by Pat ... Right is an interesting concept here. I believe that the idea of a brick wall fall off at 20kHz is dumb, I believe that there are harmonics and interactions. I don't know when the 20-20kHz idea became popular, but by the early post-War days, when Hi-Fi was invented by signal corpmen going to engineering school on the GI-bill (same guys who made ham radio be real), it was established. It is next to impossible to get a brick wall filter using coils and capacitors. I think 24dB/octave is about it, but there might be more. So you would expect that a good filter can only cut the signal by 24 dB going from 20kHz, up an octave to 40kHz. But the mics, preamps, Neumann cutting lathes, and all the anti-feedback controls all combine to each throw away a couple more dB per octave. There just isn't much up there. Arbitrary technical limits are often post-rationalizations of the technology limits of the time (e.g. 44.1/16), and one might point to the historic limits of valve-amp output transformers, speaker drive unit technology etc. (and the immature state of psycho-acoustics) for leading us to 20Hz-20KHz ... All the early HiFi stuff was tubes and transformers. Getting even plus or minus 3dB 20-20kHz was hard and expensive. And no speakers before the early 80s tried for 20-20kHz. One of the better speakers of the era was the Quad 63, which has a writeup in Stereophile on their website http://stereophile.com/floorloudspeakers/416/index11.html The scale is not detailed enough to quote where the -3dB points are, but my eyeballs estimate it as 50-15kHz. Things like Bozak systems went lower, and getting into the 30 hz zone was not all that hard (but took a lot of power/space). All in all, I think that 44.1/16 was a good engineering choice at the time. The key was that CDs were designed to replace casettes, and be better quality, longer living, and harder to replicate. That is why the labels wanted them. They did replace cassettes. And in all but a few cases, replaced vinyl. The current Stereophile (ro maybe it was TAS) has a quote from Boothroyd/Stewart big wig who said that if they had chosen 20 bit and 50kHz (or maybe 55kHz) that we would have enough to have perfect sound forever. :-) -- Pat Farrell PRC recording studio http://www.pfarrell.com/PRC ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
reeve_mike wrote: Agreed in general, but I've used some nice vintage Neumann mics that go way up high ... This is way off topic, but which ones? And how vintage? Most of the classic Neumann's like the U87 or M50 fall off pretty seriously. Now my KM184's go up high, but they aren't vintage. Pat Farrell;131153 Wrote: One of the better speakers of the era was the Quad 63 As an aside, I would claim that it is still a good speaker in this era No argument from me. I just don't have a room suitable for a pair. Pat Farrell;131153 Wrote: All in all, I think that 44.1/16 was a good engineering choice at the time. I think that it was a good (and at the time the only available) product engineering choice but I don't think that it was a good sound/music engineering choice ... :-O Sound? Who cares about sound? Labels care about money. Warner Bros just closed their Classical label: http://www.stereophile.com/news/082106classical/ Soon it will be all Britney and hip-hop. -- Pat Farrell PRC recording studio http://www.pfarrell.com/PRC ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
seanadams wrote: then upsample it to 0,0,0,0, 2,2,2,2, 3,3,3,3 at four times the rate. No, it interpolates. So you get something maybe like: 0, 0.5, 1, 1.5, 2 and so on. Thanks for the clarification. What you're talking about is oversampling - just another name for upsampling, but usually used in reference to what modern DACs do internally. It is fundamental to how they work and yes, the smaller steps require less filtering (and yield better linearity, lower noise etc). The DAC in transporter oversamples by 128x, so a 44.1 signal is actually converted to analogue at a sample rate of 5.6 MHz... a high resolution indeed. So it is more than twice as good as the SACD single bit rate of 2.82 MHz, eh? Any chance that the DAC in the Transport actually is 5.64 mHz? Now, what's stupid is taking 44.1 CD rips, resampling them to 96KHz and then re-saving to disk, thinking you've given it more breathing room or opened up the high end or whatever. It's total nonsense, exactly like on CSI where they zoom in on a single pixel, click ENHANCE and then read a license plate from a mile away. It don't work that way. Next you are going to start claiming that little wooden feet that hold your cables off the floor don't improve the bloom and remove a veil. More seriously, I don't understand why anyone thinks 96kHz is a good thing to do to RedBook. For ADAT sources, sure. But taking it to a non-integer multiple makes no sense. If nothing else, it will screw up the dither. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
