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Everything posted by beerandmusic

  1. The biggest issue is the time in engineering costs. It's not the BOM....people forget this....new designs have to include "cost of engineering".
  2. I think you could express your point without using "attack" words...take a moment and see if you could word it differently? Sounded "off the cuff"
  3. That sounds almost border-line like personal attack? Not saying you are wrong...just sounds personal.
  4. no problems with ripping if you use dbpoweramp (one of only a dozen programs i have ever paid for)...shows if rip is accurate along with checksum. i am sure it has to do with timing and power. I am not sure i have ever got better sound from a source than playing a bluray audio or sacd from a marantz sacd player regardless of what dac i had it hooked up to....and i know there are a lot more expensive transports out there....even plain cd players. Edit to add.... Even Mike Moffat (sp?) of Schiit DAC engineering, is working on a CD transport that he
  5. many times a cd transport can sound better (either using their internal dac or an external one) than playing the same file from a software player.
  6. Any salesman that admits they have sold snakeoil may be worth listening to...
  7. Very fun...watch, especially about blind test infiniti kappa bookshelf vs wilson https://www.therobinsonbrand.com/journal/3-life-changing-moments-in-hi-fi-for-me
  8. INRE topic, every time i try something new in my audio chain, i always go to eva cassidy fields of gold. Her voice just pulls you in like no other artist i have ever heard, but there are usually 1 or 2 "shrilling" moments...don't know if its the recording or just her voice, but that one or two notes kind of kills the song....wondering if have a tube preamp might help. I don't know if she has any really good recordings...seems like most out there are amateurish recordings, which is a shame she never signed with a major label....
  9. i changed the video above..no dave in that one....I also misspelled...it's kenrick not kendrick. If you haven't checked out what that guy does with vintage speakers, you ought to check them out...amazing.... Here is video with restored ALTEC's... I could still kill myself for selling a pair of Vintage JBL's that went through my hands, all for a quick $300 flip... When I heard them, I thought to myself, wow, these sound really good for their age,and off to my next flip....my biggest mistake in audio.
  10. Depends on who is doing the attacking...I have seen a lot go on here by regular users. You have a great site, but I see bias regularly. I only wish you would be more unbiased in personal attacks you allow...but i must admit, it's been a couple years since i have been on, and it does seem much more quiet now than i remember...so maybe you have cleaned up the act a little...I certainly hope you don't allow personal attacks anymore...i certainly remember seeing many a day against many different people regularly in the past.
  11. I still like listening to LP's from time to time, and am always astonished when I do.... ATC's mid dome hasn't changed much in MANY years, and they still make some of the best sounding speakers imho. Vintage JBL's refurbished by kendrick are absolutely STUNNING!! LS3/5A still carry their weight in gold. I am sure there are many other gems out there still today. I would say of ALL engineering, audio engineering has progressed the least. Computing power has been dramatic. Hell, I remember telling my boss to get into the internet and I set up one of the first sm
  12. Very informative regarding what I am speaking about::: http://www.realhd-audio.com/?p=5971 esp...I can confidently state that you’ve never heard a timbral or fidelity change when connecting a digital source and a destination using a USB cable. (with crc errors, you may get glitch, but unlikely, and you would hear it as a dropout, not as change in sq).
  13. Or perhaps, let me ask it in a different way. If you believe that the data is streamed differently than the bit-pefect flat file (lets just assume flac file pcm), how would you know that it is not transmitted without corruption, and what would it sound like, if a few bits were received incorrectly?
  14. I didn't click disagree with you UNTIL AFTER you clicked disagree with a question i had. It makes no sense to disagree with a question. I have removed my "disagrees with you". Please just put me on ignore.
  15. There is NO application problem in the transfer of bits where the bits are received with 100% accuracy. Everyone to date has agreed that the bits are received accurately. And it also has been stated that if bits are dropped that it would be heard as a tic, not as "clarity" in music as you have suggested.
  16. no, you have childishly clicked disagree with every post i make, without offering any reasoning, even on posts that i never made a statement that you can disagree with. In one such post I said I am trying to understand the logic of something, and that is all i said in that post, and you clicked disagree with. There was NOTHING in the post to disagree with, i was asking a question. You were just being annoying on purpose. Next time read what i write, and if you do not understand what I am asking, then either don't respond, or ask for clarification, but don't disagree with a "question" i mak
  17. Any software engineer (or hardware engineer) would state what I have...that the bits are transferred perfectly along with a reference voltage and noise. I am not referencing potential issues with upsampling.
  18. No, I am not arguing there is no difference, I am questioning the logic and reasoning. You would think a DAC engineer would be able to provide a "logical" explanation...but even DAC engineers shrug like they do not know why. Keeping in sight these basic principles. The only thing transferred to the dac from the pc are: 1. Digital bits (which everyone agrees the DAC receives with 100% accuracy) 2. A reference voltage 3. Noise 4. If you take into consideration that the DAC receives its bits with 100% accuracy and that the PC's reference voltage
  19. "My" point is that I am moving more toward audio/video currently and that said, the majority of what i am listening to now is less than 192K, and i am moving toward pcm via usb.
  20. I do believe that higher than 192K is audible in recordings (regardless of theorems), but there is just so little of it in mainstream. I even have a failrly large library of native dsd. I will maintain dsd over dlna for when i get the "itch", just like i maintain a PL550 for when i get an "itch" for vinyl...but for the "masses" i am going to just resort back to 192K PCM via usb. Nice to see 2 different supporters of 2 different native dsd sites though (wink).
  21. i believe that most hi-res out there is just upsampled 44.1k, esp when you consider all the hobbyists these days that are upsampling. I don't believe most mainstream recording studios record at higher than 96K PCM. That said, i am rethinking what i want. DLNA also does seem to hiccup where usb PCM just seems so much more "solid". After being a hardcore dsd over enet for 7 or 8 years now, I am actually rethinking what i want now is just 192k pcm via a quiet usb. 7 or 8 years ago, I have always thought that native dsd over dlna sounded better than pcm over
  22. Sounds great and nice site especially if you like carmen gomez... i am going to guess that most recording studios for mass stereo recording these days still record in 2 channel 44.1k or 48k unless i find differently....especially since it is mostly universally accepted that recording at higher than cd rate is beyond most people's hearing capabilities.
  23. if you google...what sampling rate do recording studio sample at I found 44.1 in 2011 Then in 2015 I find....48k in many cases for a/v... but don't find anything concrete for today...but i am still searching. 2015 If 44.1kHz Captures More than We Can Hear, Why Use Higher Sample Rates? There are a couple of reasons that higher sampling rates can be advantageous; the first is that while 44.1kHz is the standard for audio CDs, 48kHz is the standard for audio for video. Studios who regularly work in film and television may use 48kHz as their in-house st
  24. the digital signal is a representation of a sampling of the music. You can't improve on accuracy by taking samples more frequently than the recording was first recorded....you just have a sample of some guess, not of anything actually recorded....any software can have a logarithm that can guess and some may do a better job at guessing, but it cannot be any more accurate than the actual sampling. The d-a conversion wave shaping will also "connect the dots", and it is possible that wave shaping "could" be more accurate than what the alorithm "guessed". But there is never anything more accurate
  25. iimproved timing is likely needed for upsampled DSD rates which is not at least what i am looking for. I just want to know if a technology such as unsion that includes isolation and regeneration is sufficient without getting into the debate of high-res which i consider a totally different topic, that i have my own and separate debate about.
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