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RE: Then I am really confused...

>> 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.

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  • RE: Then I am really confused... - Charles Hansen 21:34:02 06/03/17 (0)


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