UPDATE after getting a comment indicating this answer for using Foobar2000. (I know I can change them one by one in Foobar, copying from the file's Metadata to its Properties How to mass-change the names so that they match the titles? - And can I do that in Foobar2000? I am relying on the MQA core decoder built into Roon, so there is only one unfold.I have some audio files for which I used Foobar2000 to get the proper tags (Title, Artist, Album, etc), so now they have their proper titles when loaded in Foobar and other players, but they still do not have the proper names as seen in Windows file manager under 'Names' column. The music is being played using Roon through a Boss DAC on a Raspberry Pi. They are being powered by a low to mid level Cambridge Audio amplifiers. We have a set of medium quality PSB bookshelf speakers with an Energy sub. I am not using really high level equipment either. The MQA version of Supertramp’s Crime of the Century sounds similarly amazing. If I close my eyes I feel like I could reach out and touch them. The MQA version sounds like the band setup a stage just behind my speakers in my living room and sat down and started playing. Using Fleetwood Mac’s Rumours as an example, the difference between the CD quality version of the album and the MQA version at 96 kHz is considerable. I am also not sure what kind of system you are using to listen to MQA files, or what recordings you are listening to, but my experience with them has been very different. If the audio engineer doing the recording is at all competent at their job the end result will be better sound, meaning it is definitely not a wasted effort to source music in higher resolution for playback unless the original recording was at a lower sampling rate to begin with. The takeaway for me is that a higher sampling rate makes it possible for the “filtering and digitising damage”, as you called it, to happen at a frequency that is not audible. Neither of those things is possible when recording at 44.1 kHz. A more gradual filter over a wider range of frequencies gives better results, and if the filter is in a frequency range that is not audible then it is even better. Second, it is not currently possible (and may never be possible) to sharply filter out sound above a certain frequency without causing distortion. First, a sampling rate of 44.1 kHz is not fast enough to capture audio without errors in timing that are audible to humans. You can have a good recording at a lower sampling rate, but it cannot capture as much information as a higher sampling rate.īased on my understanding of the video, if you record at 44.1 kHz the quality of the digital recording will be compromised in two ways that would not occur with a good quality recording at 192 kHz. Sampling rate is an element in the quality of the recording. This same argument applies to MQA, and the truth to that is obvious-if I compare the MQA and ‘straight’ version of a recording when both are available on Tidal, the top-end finesse is lost in the MQA version, and they vary from down-right awful to having definite audible high-frequency artefacts that destroy the stereo imaging. Inaudible high-frequency noise seems benign, but when it is added more than once to the signal by over-processing, it will easily add up to be audible noise. So taking the additional trouble to source higher resolution music is wasted effort-at best it will just more faithfully reproduce the damage that is already there in the source material, at worst it places more processing burden and noise into my system from computer through to tweeters. From my experience, the quality of the sound we end up hearing is more related to the quality of the recording than the sampling rate, and this video reinforces my suspicion that filtering and digitising damage is likely to have been done in the studio or distribution chain long before it ever gets to Audirvana let alone my DAC. Thanks I very much enjoyed that Hans Beekhuyzen video, and his explanation sounds convincing.
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