IIRC: The HDCD coding contains explicit information to select the "best" filter and as such modifies the impulse response constantly. Are you suggesting that the HF information in MQA is there to make the filter reproduce the time domain correctly?
No, it's not in the HF information. For as far as I understand it now, they have developed a way to store the impulse respons of the complete recording chain in metadata. They are rather secretive about it and questions I posted to Meridian remained unanswered. But Steward and Craven are extremely knowledgable and have been following research on spycho acoustics and neuro science intensely. If they managed as they claim, it really will set new standards on audio quality.
Re: upsampling: I know many people who do that and with success. But they are only compensating for poor reconstruction filters and thus not improving the recording. They are merely working around a hardware problem. Which is ok btw.
Re: resolution. I was talking about time resolution and thus filtering
Re: hires recordings. I own about 500 SACD's that I play 'ripped'. The most part contains information above 20 kHz and thus must be hires. Classics like Jacques Brel (tape transcript) but also Herbert Grönemeyer which definitely is recorded in hires (I know the mastering engineer Ronald Prent personally and heard the recording in the mastering suite).
RE MQA and time smearing: here you're absolutely wrong. There yet is no other 'codec' (essentially a wrong moniker here) that does contain metadata on the impulse respons.
Recordings made at 44.1/16 are what they are. There is no way to improve the resolution. Upsampling might give improvements when your DAC's reconstruction filter is poor (it's easier to build a filter for 192 kHz). Buying a better DAC would be a better option.
What I don't understand is your statement that there is no hires music. Many studios record at 96 or 192 kHz and when digitizing analogue recordings 192 kan be selected too.
MQA might improve 44.1 recordings since it seems to have a way to compensate for artifacts of the anti aliasing filter.
Excuses accepted. Good of you to respond this way. You must know my track record; I rather had my magazines go broke than sell my soul to the devil. It was therefore that I needed to respond. But enough about that.
The difference between PCM and DSD? The jury is still out. DSD is extremely difficult to use in production, almost always post production is done in PCM at 352,8 kHz/32 bit. The main advantage is that DSD64 has equal resolution to PCM at 352.8 kHz sampling at 32 bit depth while having a bitrate equal to 192 kHz at 24 bit depth. Another advantage is that - even when doing post in PCM - the anti aliasing and reconstruction filters cause less time smearing since they only have to filter at very high frequencies. The nice thing about MQA is that, while filtering at 192 kHz, the time smearing artifacts reportedly are compensated for while the file size comes close to a PCM at 44,1 kHz and 16 bit. Meridian (ie Craven and Stewart) had already developed a reconstruction filter that to a degree compensated for time smearing (the apodizing filter) that sounds very 'right'. Hence my enthusiasm for MQA.
Thanks for your kind judgement on my youtube video. Indeed, I'm trying to hype MQA and I love to have Likes. But there are only a few things you got not completely right: - I was streaming audio when the rest of the world still thought it could never equal cd-players. BTW I also published an ibook about it (File Based Audio aka Streaming Audio). - I do love DSD (as you could have found out easily if you had visited my site). I was amongst the first to have a DAC that supports DoP (Chord QDB76HDSD). - I do know what can be achieved, I was the editor in chief of the Dutch Pro Audio Video magazine for 21 years, visited many studios in the Netherlands, Belgium Singapore, Japan and the UK. - I did read all technical documentation, including the AES papers and patents. have you?
It's easy to judge on other people, it's hard to give decent comments. Q.E.D.