My simple analogy is what speaker correction s/w does, correcting known anomalies for a specific design on the fly, implemented for a known Dac
He would rightly argue that it is unnatural with regards to separate analogue amplifiers and passive speakers, due to group delay and phase distortion these bring; one area active speakers like the Grimm Audio are superior and technically more natural......
In his interview with Harley, Stuart said, "When we listen to music, it's analog...." "... converting the sound to digital is an unnatural act." Yet, here's Bob doing a wholesale sellout in the digital realm. I wonder if Bob thinks amplification is a natural act....
The impression I get is that Bob Stuart, Robert Harley, John Atkinson, and others are intentionally all over the map trying to make MQA all things to all people and IMO coming up with some of the most outlandish and hyped claims in an industry already known for its hype. Almost as if to make it so overwhelmingly impossible or exhausting to debunk the technology as a whole.
Personally, I think it's perhaps the greatest charade ever pulled on this industry.
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I think it is important, especially in audio, to separate the claims from the underlying science. For some reason after many years in this hobby, I do that instinctively. The unsubstantiated claims go in one ear and out the other for me. But, my impression is that you are hung up on the verbiage of claims and marketing, then you are going through convoluted logic about those claims. However, you are not really understanding the underlying science and its implications. Amir has provided a good explanation of the possibilities of the science here, also avoiding getting bogged down in what are likely to be fictitious or overblown claims.
In all I have read in the actual words of Stuart, Harley and Atkinson, I see plausible explanations from a scientific standpoint of how this MIGHT be able to provide an improvement in sound quality over digital audio as we know it. That potential improvement is not a result of an increased sampling rate beyond hi res audio as we know it, which is a tangent you have pursued.
And, we are all big boys here. If anyone expects this or any other technology to once and for all totally eliminate the gap between live and recorded sound, they are living in fantasy land. But, honestly, I have seen no such claim for MQA in my readings. All that is promised and all that I expect is an improvement, a step in the right direction, that may take us closer to the mike feed and beyond the limitations imposed by the original analog to digital conversion in the recording process.
Per some of your quotes, I agree that the mike itself does not hear live music as we do. Unless and until that can be overcome, if ever, we have to settle for incremental sonic improvements later in the recording chain, such as what MQA purports to deliver. That is all I see Stuart, Harley and Atkinson alluding to.
Well, objectively (i.e. measurements) it may be able to do that as I explained. Let's create a fictitious example of an A/D converter that drops by 0.5 dB by 20 Khz. If I measure and profile that, then at playback time I can insert an inverse of that restore that droop to a flat line. This would be measureable.MQA can not possibly improve sound quality over the original full high resolution input file that it encodes. Anything along these lines is just marketing smoke which makes me suspicious of everything the promoters and their paid shills say.
I think it is important, especially in audio, to separate the claims from the underlying science. For some reason after many years in this hobby, I do that instinctively. The unsubstantiated claims go in one ear and out the other for me. But, my impression is that you are hung up on the verbiage of claims and marketing, then you are going through convoluted logic about those claims. However, you are not really understanding the underlying science and its implications. Amir has provided a good explanation of the possibilities of the science here, also avoiding getting bogged down in what are likely to be fictitious or overblown claims.
In all I have read in the actual words of Stuart, Harley and Atkinson, I see plausible explanations from a scientific standpoint of how this MIGHT be able to provide an improvement in sound quality over digital audio as we know it. That potential improvement is not a result of an increased sampling rate beyond hi res audio as we know it, which is a tangent you have pursued.
And, we are all big boys here. If anyone expects this or any other technology to once and for all totally eliminate the gap between live and recorded sound, they are living in fantasy land. But, honestly, I have seen no such claim for MQA in my readings. All that is promised and all that I expect is an improvement, a step in the right direction, that may take us closer to the mike feed and beyond the limitations imposed by the original analog to digital conversion in the recording process.
Per some of your quotes, I agree that the mike itself does not hear live music as we do. Unless and until that can be overcome, if ever, we have to settle for incremental sonic improvements later in the recording chain, such as what MQA purports to deliver. That is all I see Stuart, Harley and Atkinson alluding to.
Well, objectively (i.e. measurements) it may be able to do that as I explained. Let's create a fictitious example of an A/D converter that drops by 0.5 dB by 20 Khz. If I measure and profile that, then at playback time I can insert an inverse of that restore that droop to a flat line. This would be measureable.
Where it gets tricky is that Bob attempts to correct what he says are the timing distortions created in capture despite the high sampling rates. The audibility of such distortions is not accepted currently. So while objectively we may be able to show this, subjectively, i.e. when we listen, these differences may be well below audibility thresholds.
Well, objectively (i.e. measurements) it may be able to do that as I explained. Let's create a fictitious example of an A/D converter that drops by 0.5 dB by 20 Khz. If I measure and profile that, then at playback time I can insert an inverse of that restore that droop to a flat line. This would be measureable.
Where it gets tricky is that Bob attempts to correct what he says are the timing distortions created in capture despite the high sampling rates. The audibility of such distortions is not accepted currently. So while objectively we may be able to show this, subjectively, i.e. when we listen, these differences may be well below audibility thresholds.
I see your point but the process they going down the path of, takes care of this in that the producers of music will have to a) help profile their ADCs and b) perform quality testing to tell if this is an improvement or not, as to then recreate all of their stereo digital masters for distribution.The problem is that the recording may have been optimized by the mastering engineer to sound the way he and the producer liked based on the particular distortions of the ADC that they used. Second guessing this process amounts to "automatic remastering". Actually, doing this is impossible. The filtering and equalization (or microphone placement for a purist recording) were made based on listening to the final product through a particular playback chain. The filters in the DAC, not to mention the mastering speakers and room, made the recording sound good, but the information about this playback chain is not encoded on the recording.
Looks like I created the problem with that analogy . Per above, there is no frequency response fixing but rather, other processing to better preserve timing resolution.There is another problem with second guessing the ADC filters. Taking your specific example, that 0.5 dB boost can certainly be applied, but doing so does not just restore the amplitude response to be flat up to 20 kHz. It will have affects on the response at other frequencies as well as the time domain response. These effects may be audible and their evaluation may depend on the specific type of recording and musical genre. Perhaps there will be some future artificial intelligence technology that will automate the mastering engineer function, but I doubt that MQA is such a technololgy.
Well, it is better sound if one reads the effect of processing in conversion to long word PCM (i.e. 24 bit) to have impacted time domain/transient response. This is highly disputed as I mentioned but is the nature of what they are trying to do. As they demonstrate in the paper, this can be measured. It remains to be seen if it can be heard.It is certainly possible that MQA can do a better job of fitting audio into a fixed bitrate than merely adapting a lower PCM format with associated lossless coding. However, I put this in the category of "saving bits" not "better sound" and this is about what is good enough, not what is best.
Great point. Having one's purchased music in this format and having the company discontinue its promotion, could prove highly costly to consumers who would at some have to buy their music all over again.Incidentally, unless MQA comes with a complete specification and open source code for the encoder and decoder, together with free and unrestricted licensing, I have no intention of even investigating the technical details. We have too many formats as it is to justify another one that is propritary. (I never even considered SACD because of its use of DRM and new patentent technology. I believe this contributed heavily to SACD's failure in the marketplace.)
That's an interesting point. To the extent Meridian inserts itself into the heart of production, there can now be assurances regarding provenance of the content, and purity/avoidance of resampling of low-res content.Also I thought the MQA also deals with the real world label-studio-distribution that could do upsampling and/or downsampling (we know it happens), where again it is possible for it to be messed up (definitely happens).
Cheers
Orb
stehno
You seem to be getting all excited and hung up on descriptions/write-ups of what MQA will or won't go. I suggest you take the opportunity to hear it before you write another 1000 word essay giving us your opinion as to why it can't work.
The demo I heard may not have knocked me off my chair but I did hear SQ on a couple of tracks that were significantly different/better than any version I have heard in the past.
The problem is that the recording may have been optimized by the mastering engineer to sound the way he and the producer liked based on the particular distortions of the ADC that they used. Second guessing this process amounts to "automatic remastering".
I, for one, have absolutely zero interest in hearing a new proprietary format. I don't even want to hear files that have been put through such a process, since this is open to all kinds of selection or tweaking. I consider even listening to these files to be a bad idea. The industry needs a new proprietary format like a fish needs a bicycle.
If I am given source code for the CODEC that allows me to experiment with my own choice of source files and which comes with a free and open license for the code and all patented technology, then, and only then, would I be interested in evaluating this format. [I am not singling out Meridian. I went through the same due diligence with FLAC.]
That's an interesting point. To the extent Meridian inserts itself into the heart of production, there can now be assurances regarding provenance of the content, and purity/avoidance of resampling of low-res content.