Fokus Posted January 18, 2017 Share Posted January 18, 2017 Can you isolate the primary upsampler's response (magnitude, phase, impulse) for a non-MQA input, and see if it is identical to Fig.6 here Mytek HiFi Brooklyn D/A processor–headphone amplifier Measurements | Stereophile.com and Fig.4 here Meridian Explorer2 D/A headphone amplifier Measurements | Stereophile.com? Link to comment
Fokus Posted January 18, 2017 Share Posted January 18, 2017 They are related. The filter is named 'MQA' on the Mytek. But the interesting thing is that it looks suboptimal, regardless to which filter church you belong, and yet ... the Meridian Explorer2 has a very similar/same filter. If the Blue Sound code again shows the same response (or a very similar one), then we know what sort of reconstruction filter is required by MQA, and this may yield insight in how the band splitting and joining is done, and what compromises were made there which may/may not impact on the quality of undecoded replay as compared to decoded replay. Link to comment
Fokus Posted January 20, 2017 Share Posted January 20, 2017 The trouble with short filters is that they are not steep enough to avoid serious amounts of aliasing. The crux of MQA, when dealing with original rates in excess of 96k, is to allow aliasing, but within limits. Particularly in the baseband the filters are tuned so that the aliasing there does not exceed the programme's innate noise or another suitable masking threshold. In the 24-48kHz band the situation is more dire. What air has to do with anything completely escapes me. It is a marketing slogan. "Our new digital system does as much damage as 10 m of air. Quick. Rush." Link to comment
Fokus Posted January 20, 2017 Share Posted January 20, 2017 4. this get through the cochlea into ones brain Maybe they assume the typical audiophile's head is filled with air? Link to comment
Fokus Posted January 20, 2017 Share Posted January 20, 2017 So "the situation is more dire" where most of the recorded music is? Please read again. My music does not reside in the 24-48kHz band. Maybe Batman's does? Link to comment
Fokus Posted January 20, 2017 Share Posted January 20, 2017 I'm not too familiar with the Meridian filters. Got a pointer to some info? The 'classic' Meridian apodising filter for CD replay is a steep, minimum phase filter starting at 20kHz and cutting deeply before 22kHz, entirely free of image generation. Its rationale is to cut away the pre- and post-ringing of the presumed halfband linear phase production stage AA filter (ADC or downsampler), and replace it entirely with their own post-ringing at a lower frequency. Meridian 808.2/808i.2 Signature Reference CD player/preamplifier Measurements | Stereophile.com This is in part why I asked you before to quantify the CD-rate upsampling filter of the BS code: what has been published in Stereophile for both the Brooklyn and Explorer2 MQA DACs is so suspiciously un-Meridian that this may be a pointer to what the MQA codec does. Meridian Explorer2 D/A headphone amplifier Measurements | Stereophile.com If the Blue Sound code contains the same horrible filter than this must be seen as a part of the MQA standard. If this same filter is used in the unfolding process (for the baseband) then this makes us ask what sort of filters were used in the folding process (LP for the baseband, HP for the ultrasonics) and what this means for possible aliasing distortion in the non-decoded baseband, because one can expect these four filters to be mandatory related to each other. Link to comment
Fokus Posted January 20, 2017 Share Posted January 20, 2017 All in good time. You are doing stellar work. I am asking this now for the sake of documentation, not to put any pressure on you. Link to comment
Fokus Posted February 2, 2017 Share Posted February 2, 2017 Well, it shows that the low bits encode, somehow, the high-frequency content. It shows that the folded part also contains information below Fs/2, possibly suggesting something about the filters used in the lossless spectral split and join at Fs/2. (With Fs/2 meaning 24kHz.) Link to comment
Fokus Posted February 5, 2017 Share Posted February 5, 2017 Not related to the signal processing, but to the format itself. MQA seems sensitive to ID3 tag contents. I took an MQA wav file, and converted it to flac in two flavours: one with ReplayGain writing to the ID3 tag, one without. The Explorer2 refuses to recognise the ReplayGained track as MQA, even though the actual signal in the file nulls perfectly with the original MQA wav. The Explorer2 is fine with the non-RGed track. Link to comment
Fokus Posted February 5, 2017 Share Posted February 5, 2017 You are probably right. Running all these experiments it is easy to forget a setting in Foobar. As to the DAC not seeing the header ... it seems that I also forgot to wake up today. Link to comment
Fokus Posted February 7, 2017 Share Posted February 7, 2017 So you cannot apply any DSP of your choice; like replay gain, digital room correction, headphone cross-feed or stuff like that, unless those are specifically approved/blessed by the MQA company. I know. I wrote that years ago, mere days after MQA plans were released. The point is that the machine/player I use for analysis is not supposed to have any processing going on. When I find the time I will return to it and check if it has RG totally inadvertently enabled. Link to comment
Fokus Posted February 11, 2017 Share Posted February 11, 2017 That sounds like a standard apodising filter. Nothing novel about that. Anyone can do it. Nope. There is a paper or a patent that describes an all pass filter with group delay severely peaking around the recording Fs/2, effectively lifting up the ringing and dropping it down after the main impulse. Their description, not mine. A bit ridiculous, this obsession with a phenomenon that is only really triggered by a tiny portion of the music, and that has to be inaudible to all but the newly born. You would think that if this really mattered at all they would stage a convincing demonstration of it at conferences and fairs, not? Just a single note or hit, with and without the magic applied. But no, what you get are videos of semi-celebs, running down the streets, tossing off their clothes and screaming how they have seen the light. Link to comment
Fokus Posted February 11, 2017 Share Posted February 11, 2017 Bob Stuart stated more than once in the Q & A on here that MQA does not use apodizing filters, yet many people on this thread are stating that it does. It would be good to get some clarification on this. That is correct. If we define 'apodising' the way Craven/Stuart/Meridian define it (i.e. minimum phase with the cut-off well below Fs/2, so that any pre-existing ringing at Fs/2 is sliced away), then MQA indeed does NOT use apodising filters. It is remarkable that the Explorer2, when reproducing standard non-MQA CD-rate material, also does NOT use a Meridian-style apo filter, but rather an extremely leaky minimum phase ... monstrosity. See figures 2 and 4. Meridian Explorer2 D/A headphone amplifier Measurements | Stereophile.com -- Which brings me back to an earlier question of mine: given that the above 'bad' filter also features in the Mytek DAC we can assume it is part of the MQA standard. Does this filter play a role in the folding/unfolding part, and what does this imply for the replay of non-decoded MQA material? Or, to put it bluntly, if this leaky filter is mandatory part of MQA replay, does this then also imply a leaky filter used during the pre-folding band splitting, thus infecting the baseband with severe aliasing that remains present during non-decoded replay? Link to comment
Fokus Posted February 11, 2017 Share Posted February 11, 2017 Mansr: pretty much all I wrote above was for baserate, not at all about the renderer filters. Mytek/Explorer2 measuments are very relevant because they make one ask why the hell they chose a reconstruction filter for baserate with zero attenuation at Fs/2 and a first null at Fs. It simply does not match Meridian's past philosophy. So what forces them to do this? -- The renderer's filters may be horrible, at least they fit MQA's published philosophy, and they are supposed to be mostly harmless. Link to comment
Fokus Posted February 12, 2017 Share Posted February 12, 2017 just note how the design appears to force the end user into MQA play by the poor handling of non MQA encoded files/music. Do note that this has not been proven. Link to comment
Fokus Posted February 12, 2017 Share Posted February 12, 2017 - I'm not catching Fokus' point about what this implies for the encoding, and hope it can be explained in a little more detail. When MQA folds a 96k recording in a 48k container it needs a low pass filter at 24kHz, to isolate the baseband, and it needs a high pass filter at 24kHz, to isolate the ultrasonics prior to their undersampling (the technically correct term for the folding). Let's call these filters F1l and F1h. Similarly, during replay/unfolding, the baseband (i.e. music up to 24kHz) has to be reconstructed, requiring a filter F2l. The ultrasonics have to be band-shifted, which can be done by oversampling x2 combined with a high pass filter, F2h. The outputs of these two filters are summed, yielding the full 0-48kHz signal. For the folding/unfolding to be lossless (i.e. no errors are introduced compared to the original 0-48kHz signal), you can imagine that the four filters F1l, F1h, F2l, F2h have to obey specific criteria, and more, that these four filters are inter-related. OK. So far for MQA encoding and decoding. Forget it. Now to the Stereophile measurements of the Mytek and Explorer2. You see there, for normal baserate replay, i.e. non-MQA, that both DACs use the exact-same reconstruction filter. This filter is curious, because it is minimum phase and very lazy, cutting only above Fs/2 (!) and reaching zero only at Fs. This is curious because 1) it is a crappy filter and 2) it is very un-Meridian. And yet, both DACs show the same filter ... coincidence? This triggers the question: is this filter actually the same as the F2l mentioned above? And if so, what does this tell us about F1l? Does F2l being crappy imply that F1l also is crappy? If true, this would mean that undecoded MQA would be infected by a crappy anti-aliasing filter, causing damage to the signal that is only corrected when MQA decoding is engaged. But again, this is just conjecture. Link to comment
Fokus Posted February 14, 2017 Share Posted February 14, 2017 Isn't this why vendors must obtain MQA "certification" ? (so that they correctly implement F1l and F2l). No. The split-and-join filters are a key part of the MQA spec, and implemented in the software. The certification is required because MQA have to brew up the renderer filters that undo a particular DAC's sins. Link to comment
Fokus Posted February 14, 2017 Share Posted February 14, 2017 Have you seen any evidence of such? I should have used quotes. Or whatever punctuation symbol is used for mild sarcasm these days. Link to comment
Fokus Posted June 30, 2017 Share Posted June 30, 2017 5 hours ago, mansr said: and nothing points to any real difference. Which is even more interesting if you factor in that the Node 2 and the Dragonfly have totally different DAC chips. Perhaps the 16 (32) filters are generics, designed to match broad families of DACs, and is the DAC tuning not much more than picking the best generic filter(s). Link to comment
Popular Post Fokus Posted January 16, 2018 Popular Post Share Posted January 16, 2018 9 hours ago, Shadders said: Do you know what ADC is prevalent in the studio in the 1980's and 1990's, and currently ? As others said, from the early-mid 90s on all sigma-delta stuff, with first or second order analogue pre-filtering and then digital half-band anti-aliasing filtering in the ADC's downsamplers. And before that linear ADCs sampling at baseband, and thus with analogue AA filters, something like 8th order elliptic. The SDM ADCs with their HB filters are easy to recognise: do a spectral analysis of a CD, when the spectrum runs right up to Fs/2 you are looking at HB: there is always a bit of aliasing. The older ADC class is typefied by the Sony PCM1610/1630 systems. I have quite a few older CDs where the spectrum reaches a lot of attenuation at Fs/2, i.e. free of aliasing. Remarkable. There is not much data on these old ADCs available on the net, but here is some Sony DAT review. If I remember correctly the Pacific Microsonics ADC (or rather, the entire system including HDCD encoding) used 2x oversampling, so part of the CD-production AA filter was done in the digital domain. Oh, and Audio Note had a prototype non-filtered linear ADC at around 2000. I still have a CD with test recordings PQ gave me ;-) semente, Shadders and opus101 2 1 Link to comment
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