JR_Audio

Hi Chris. Wow, this is a great project. Congrats! Juergen

Hi Mitch. Great Report. Thanks. One question, as this seems to work “dynamically”: If you would make some steady state frequency response measurements with different levels, will you then be able to see changes / correction in the resulting frequency response? Juergen

Hi Mitch. Happy New Year. Once again, I appreciate your thorough work. Yes, it takes a while to read all the "pages", but it's worth every paragraph. Thumbs up.

Happy New Year Chris and a great idea with going into the audiophile style 😉

Hi Chris. Oh man. A lot of work. Good luck. Juergen

This could have 4 different reasons: Microphone, Mic Pre-Amp, Tape Head or Venue Noise This could be the SMPS noise of the polarization voltage that could go into the signal for 2 reasons: Either the signal transformer / path is not fully balanced, so that the SMPS noise is not canceled out, that would otherwise be the case with balanced signal path (if the in-phase path has for example 1k1 and the off-phase path 1k0 input resistance), or what I have had with older Neumann Tube of FET mics is, that the filtering electrolyte got a cold solder point. It is always good to watch the signal of the recording setup with tools like SpectraFoo for abnormalities. I have sent back some mics, that have not been fully balanced, as also some mic preamps, that have not been fully balanced. PS: Some tape machines do have also those spices due to the AC bias of / in tape head, but I don’t think this is the case in the above example, that looks like a tape free recording. Another reason could be a noise in the recording venue: I had a life recording in a church and have had a 40.5 kHz spike in my setup and finally found out, that this was an “anti-bird” noise, that was send out with some piezo tweeters in the church, to prevent birds coming in / flying into the church. In another case, I have had an ultra frequency smoke detector at around 36 kHz. As I said above, it is always good to check the signal that is to be recorded. I hope this helps. Juergen

Hi Mitch, Hi Archie. Very nice. I would have liked to join you for this event. It would have been fun, for sure. Thanks. Juergen

For my Genelecs 8260 in my Mixing / Mastering Room, that are coaxial from low mid on, so very good sound stage and localization for near / mid field mixing and mastering, but not time coherent, I have the miniDSP Dirac DDRC-22D (Digital-In to Digital-Out) and use this for Room EQ (that could the Genelec GLM also) and also to make them time coherent, and use the digital volume control of the miniDSP to control the SPL. Yes, I know, the miniDSP does run internally with 96 kHz sample rate, and does SRC all other sample rates, but being time coherent up the 48 kHz (96 kHz / 2) is of much higher of value, than having for example 192 kHz all the way through, but not time coherent. Juergen

Hi Mitch. Wow, another great review from you. Congrats on that. I enjoyed every paragraph you have written. I am familiar with both speakers (Dutch & Dutch 8c and Kii THREE) and do agree with many points you have described. But for most, it is great to see such thorough review of speakers with the focus, how they behave at home, and not mainly in anechoic chambers. Juergen

Hi Mitch. What a great work and write-up. My full respect for this level of clean analysis and writing. This is extraordinary. Thanks. Juergen

Congratulations Chris, also from my side. Great. My best wishes for you for the upcoming future.

Hi Pal Good, that you join this group. My two points, are not adressed against you in any way but: MQA is lossy, where plain PCM would not, and concerning the sound quality of actual releases. If I would remove every actual release, that do have inter sample overload or native overloads (leaving aside the much to heavy dynamic compression), then nearly no pop / rock release would remain. Juergen

New Masters? And if every recording has to be re-mastered individually, to get the MQA approved sound, why not just remaster into "regular" PCM files, just with less dynamic compression and less inter sample overloads and have still fully compatible pure High Res PCM files, that can be used with digital room corrections or with slide volume control (speaking volume matching) and still have regular 96k or 192k rate, that work in High Res on every regular High End DAC? Juergen

Hi Chris This question can be answered due to the 24 Bit transparency of AIFF, ALAC, WAV and FLAC. You can convert (or should I say, you should be able to) the MQA FLAC file one hundred times between AIFF, ALAC, WAV and FLAC to any version, and when playing back, the MQA led should / must “shine”. More interesting would be, will there be a software decoder, that could convert the MQA “packed” file, into a “pure” 352 or 176 kHz PCM file, or speaking, unfold the MQA origami for non MQA DACs or for DAWs. Juergen

I have already mentioned my two questions on the Archimago Blog, that is really worth reading: Archimago's Musings: MEASUREMENTS: MQA (Master Quality Authenticated) Observations and The Big Picture... In this blog you will see, that from the technical point of view, MQA have around 13 Bit of “lossless” information and everything below 14 Bit is “lossy”. Doesn't mean that is will not sound good, it just means, that this is not a lossless codec, it is lossy (from the technical point of view). Here are some Screen shots of the noise in the MQA data, compared to a plain dithered 16 Bit and to a noise shaped dithered 16 Bit stream. But here are my two questions: Will MQA work with Digital Volume Control and also with Digital Room Correction? 1. If I reduce the level of the digital signal within the Playback Software to just about 1 dB (in the digital domain), will the MQA light will still lit up and will the MQA stream still be "unpacked", or will this be end up in the 44k1 24 Bit digital stream, as when having non MQA certified DAC? Nearly all playback software do still sound good, if you adjust the level of the tracks just within some dB. 2. My thinking is, then also no digital room correction will work in that way, that it will benefit from the MQA unpacking, and all files will end up with just 44k1 24 Bit container, but mainly only with 13 Bit of audio information, no matter what sample rate the original file has had in the MQA encoding process. In comparison a lesser compressed master files with real 24 Bit 96 kHz sample rate from start to finish (in a real lossless codec) will do a more practical job, in our daily homes, especially when you plan or already use a Digital Room Correction (in the digital path, between or in the playback software, and before the DAC). Also in the above mentioned blog you will find some additional comments from me, about my listening impression of the the Non Decoded MQA Listening (on non MQA DACs) and some speculations about Linear Phase and Minimum Phase filtering. Juergen