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In Reply to: RE: Now I think we are getting somewhere posted by Jim Austin on June 01, 2017 at 15:35:36
If the original file is "only" 44.1/24, this raises two questions in my mind:
1) Why bother to process it through MQA at all? The file size would be smaller if just straight FLAC were used, plus there would be no loss in resolution.
2) If the original graph posted with the "notch" centered at Fs/2 really is just a 44.1/24 file, what is the content seen in the dual-rate band? Clearly it is not just an alias of the original baseband audio, or it would be a mirror-image. Even after passing through the MQA digital filter, the mirror image would only be down -3dB at ~38kHz. The graph does not look like that at all to my eyes. Instead it looks like the spectral content expected in a true 88.2/24 file but with two artifacts - the big notch at Fs/2 plus the mild hump between 40kHz and 45kHz.
As always, these posts are strictly my own opinion and not necessarily those of my employer or second cousin, twice removed.
Charlie - I may be technically all wet, but from what little I know, I do believe what you are looking at is an alias of the original signal. My layman's reading of Stuart's technical papers is that he was willing to accept a certain amount of aliasing, perhaps high by other's standards, in order to get the time domain performance he wanted, all based on his tested psychoustic experimentation, of course.
So, what you may be seeing is precisely that instead of an origami unfolding or other bug.
My limited listening so far, which is of course subjective and anecdotal, makes me believe there is ample reason to apply MQA to a 44 or 48k Master, even if no higher sampling rates are available. In fact, I find that MQA has MORE to offer there vs. the non-MQA original than it does when comparing MQA vs. non- at higher resolutions.
Stuart's published graphs in AES papers and elsewhere attest to the fact that 44k has much greater "temporal blur" than at higher sampling rates, and that impulse response is cleaned up much more by MQA at 44k than it is from, say, 192k masters. My subjective impression is that MQA/44K sounds much more like native hi rez.
I have not done much native hi rez vs. hi rez MQA comparison, but I feel there might possibly be considerably less to be gained from MQA by a hi rez listener like me.
> > I do believe what you are looking at is an alias of the original signal < <
I'm pretty sure that it is not aliasing. When you read a book on digital audio (such as Ken Pohlmann's popular one, "Principles of Digital Audio") they almost always show the aliased signals as mirror images of the original signal, extending upwards in frequency to infinity.
Mathematically this is how it works, but in real life there is no such thing as "infinite frequency". It's shown this way because the digital audio theory presented is based on an imaginary abstract concept called a "Dirac delta", which is defined as an infinitely narrow impulse that still contains a finite (quantized) amount of energy. There have been a handful of DAC chips made with pulse outputs (including the one in the original Sony SACD players, the SCD-1 and SCD-777). While not *infinitely* narrow, they would still create a good mirror image of at least the *first* aliased spectral reflection. With a chip like this, the unfiltered spectrum would have a "V" shape - the actual audio energy decreasing with frequency up to the 20 kHz cutoff of the anti-aliasing filter (in the A/D converter) and then mirror-imaged upwards to the sampling frequency.
However pulse-output DAC chips are extremely rare compared to a "zero-order hold" DAC chip that holds the value of a sample until the next sample is entered. These chips output a waveform that looks like a "stair-step" representation (see Figure 3 in link below), and finally the reconstruction filter in the D/A converter filters out the high frequencies (artifacts of the "steps"), leaving a smooth analog waveform without steps.
The "zero order hold" found in nearly all DAC chips performs an unavoidable combination of low-pass filtering and comb filtering (search for images of the "sinc function"). Specifically the audio will be about -4dB at Fs/2 and gradually falling to zero at Fs. However this curve has a known, specific signature, as does the natural spectral content of musical instruments, as do both "leaky" or "brickwall" or "apodizing" digital filters. One skilled in the art can examine the spectrum of the analog output and identify each of the "fingerprints" left behind by each. In my opinion, the waveform shown does not look like what would be caused by a "leaky" filter, but instead exactly what would be expected if the original file were recorded at 88kHz - except for the notch centered at Fs/2.
As far as "improving" the sound quality of an existing single-rate recording, the picture is far from clear regarding MQA. If you are comparing a 44kHz file to the MQA version of the same recording, the MQA file will use different digital filters during playback - even if listening through the exact same D/A converter. If you prefer the sound of the MQA version, it would seem to indicate that you prefer the sound of the MQA digital filter to the standard one in that same D/A converter. To me that simply confirms something that has been known for decades - that different digital filters sound different. In that case the question becomes "Does the MQA digital filter sound better than *all* other digital filters, or just the other one built into this particular D/A converter?" I believe this last question may explain some of the mixed opinions currently existing.
As always, these posts only reflect my personal opinions and not necessarily those of my employer or the local chief of police.
In my previous post I wrote "Clearly it is not just an alias of the original baseband audio, or it would be a mirror-image.". This would *only* be true if the DAC chip output a series of (ideally infinitely) narrow pulses. However virtually all DAC chips output a given level that is "held" until the next sample (called a "first-order hold"). This alters the frequency response of the output by convolving (basically superimposing) the frequency response of the "sinc" function, where sinc (x) = [sin(x)/x]. Regardless the original chart does not look like that either, so my confusion persists... :-(
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