Jump to content

Tony Lauck

  • Content Count

    512
  • Joined

  • Last visited

1 Follower

About Tony Lauck

  • Rank
    Sophomore Member

Recent Profile Visitors

The recent visitors block is disabled and is not being shown to other users.

  1. The preach level is entirely justified when it comes to the importance of precise level matching whenever subjective sound quality comparisons are performed. People who don't do this are either ignorant or dishonest shysters. This one factor goes back throughout the history of audio, in my case all the way back to the late 1950's when I started getting interested in hi-fi. If you want to measure levels accurately between two formats where a proprietary decode process is bundled into a DAC you will have to conduct measurements at the analog output of the DAC. You can do this by t
  2. Subtractive dither is more than a ruse, however. Compared to TPDF dither, subtractive dither provides higher audio quality (removing all correlation between signal and dither noise and not just first and second order correlation). In addition compared to TPDF dither it provides an approximate 6 dB gain in S/N ratio or, equivalently, saves approximately one bit. I found that subtractive dither used to convert 96/24 audio to 96/8 audio sounded musical, albeit noisy like an old 4 track 7.5 ips pre-recorded tape. At 96/12 the noise was similar to that on high quality analog tape. In
  3. I was discussing the need for the scrambling function needed to whiten the low order bits that represent the folded high frequency information. Since these bits appear in the code space for the undecoded playback they need to appear to the receiver and listener as uncorrelated with the music. That way, they will be heard as random noise, rather than distortion. (It's actually slightly more complicated because changing the low order bits to random values will still be correlated to the music in the form of noise modulation, but these are details of the dithering algorithms and are unrelated
  4. Encryption is not a necessary part of the folding from an audio perspective. In the MQA "ecosystem" encryption serves three purposes: a technical purpose associated with the folding, a control purpose which adds difficulty to competing product implementations, and a legal purpose enabling criminal prosecution of control violations. When encoding information that is correlated to the music (and this includes the folded high frequency content) the encoded information must be scrambled or "whitened" into pseudo-noise to convert distortion artifacts into noise artifacts. This can b
  5. This is complete and utter BS. There are many formats that I have in my library and I can play most of these on my DAC, converting them where necessary by software. There are things that I do with DBpoweramp, HQPlayer, SoundForge and iZotope RX that allow me to deal with these formats and where necessary I can download new CODEC software to access different formats. However, if I were to be forced to use MQA I would not be able to do things that I presently do, such as to do digital room correction for both PCM and DSD using HQPlayer. I don't want hardware. I don't want ch
  6. Depends. Unfortunately, many labels have victimized their artists, who are generally innocent of all of this skullduggery. But some labels have been on my do not buy list for decades, generally because they have a track record of poor sound quality. I suspect quite a bit of overlap, here.
  7. All the money spent buying new DAC hardware is going to vultures. Consumers have limited funds and any money that doesn't go to the musicians and song writers and the engineers who made the original recordings is going to parasites. MQA is nothing but a parasitical scheme involving the legal system to extract rent from music lovers at the expense of everyone else. (Legal aspects include proprietary formats backed up with non-disclosure agreements and patents and the use of encryption that can't legally be "circumvented" due to fascist legislation such as the US DMCA law.) IMO
  8. DAC designers have to make a tradeoff. Assuming the digital sampling has extra guard bits they can afford to provide headroom to prevent clipping. However, unless the output of the upsampling is then reduced down to the actual resolution of the converter circuitry there will be clipping. Here's the conflict: if they provide little reduction than "hot" music will sound distorted, but the measured S/N of the DAC will be good. If they provide more reduction then "hot" music will be clean, but the measured S/N of the DAC will be inferior. Pro audio equipment often allows for the converter to
  9. Easily done with audio workstation software such as Audacity, Izotope RX, or Soundforge. There is no way to tell whether or not a DAC will produce intersample clipping on a recording that has no clipping except by listening (or measuring the inpulse response). For listening, you can use a digital volume control in the computer, such as HQPlayer or a digital volume control in the DAC (if it's been done correctly) and see if this eliminates the harshness. (You will need to make corresponding analog volume control changes to get a fair comparison.) If you are a decent recording eng
  10. No bats. Just a non-linear mechanism, my ears. Not to mention non-linearity in the DAC, amp, speakers and air. Take a non-clipped waveform with energy close to 22 kHz and with peaks close to 0 dBFs. Now put this waveform through a steep filter (or just about any low pass filter for that matter). You will now have a waveform that has peaks above 0 dBFs if you do the calculations in floating point or otherwise with headroom. Now, when you put the result out in a regular PCM format without a gain reduction you will get clipping, and this will often result in audible distortion.
  11. This is all BS. The filter used for playback is going to interact with the filter used for recording. In the event of compromised systems, such as 44.1 kHz PCM, these filters have subtle but audible interactions. It is not possible to have a system that has full frequency range (e.g. up to 20 kHz), is free of ringing, and does not create spurious frequencies due to aliasing. The optimum filter for playback of 44.1 recordings is going to vary according to the filter used for playback and upon the quality of the original recording, even the type of music. This is not a matter of "proper" or
  12. One of the problems I had with the bass was room related. The speakers needed to be placed appropriately for decent midrange and highs, but this created some problems with room modes. I tried many different positions from the back walls and adjusting the cross-over controls on the main speakers, and none were satisfactory. The sub allowed another degree of freedom, particularly with respect to the vertical room mode. In addition to gain controls, the sub had a phase control and adjustable cross-over. Unfortunately, all of these knobs required crawling about on the floor and forgetting
  13. Let me guess. You are not a computer hardware engineer familiar with design of state of the art digital equipment, whether it be computer chip design, computer motherboard design, storage device design, or communications equipment. All of these disciplines involve pushing the limits of signal integrity and all the people working competently in this space are well aware that "bits are bits" is a convenient mathematical abstraction for some purposes, but irrelevant when designing equipment that has to work reliably in the real-world. And if you are a programmer, I hope you aren't working on
  14. I took some of these older piano recordings and digitized some of them. Also some live concert material where there won't be "in the room" credibiilty since there will be venue sonics. These are on a web site: http://www.susanlauck.com/ Enjoy these free downloads. On the studio recordings you may notice that the piano image is excessively large. This will happen if the physical spacing of hyour speakers is wider than the spacing of the speakers that I used, where the setup had prevously been determined as a compromise with many high quality recordings that I used for playback setup at the
  15. I have done this several times, with the exception of the curtain. Apart from the different location of the speakers and the piano, no way to tell. However, in addition to obsessing over playback setup it took quite a bit of work finding the right microphone positioning. The first time I did this was in the mid 1970's, later in the 1980's. Speakers the first time were AR-3a's, the second time Snell AIIIs. Two caveats: the recording and playback were in the same room and the music did not use the bottom three notes on the keyboard. In both cases a 7.5 IPS 2T Tandberg recor
×
×
  • Create New...