Here you go: Chris Montgomery - Wikipedia Christopher "Monty"[1] Montgomery (born June 6, 1972[2][3]) is an American programmer and engineer. He is the original creator of the Ogg Free Software container format and the Vorbisaudio codec and others, and the founder of The Xiph.Org Foundation, which promotes public domain multimedia codecs. He uses xiphmont as an online pseudonym. He holds a B.S. in electrical engineering and computer science from the Massachusetts Institute of Technology and a M.Eng. degree in computer engineering from the Tokyo Institute of Technology.[4] A multimedia programmer, free software advocate and musician,[5] Monty resides in the Bostonarea. He previously worked for Red Hat on improving the quality of the Ogg Theora format and decoders. In October 2013, he announced[6] his almost immediate switch to Mozilla. Work on Daala will be an important part of his work there.[needs update] I think I'll trust him.
huh? Nup. Take an recording, run it through converters at whatever sample/bit rate you desire. You have a facsimile of a recording. Your sampling it. You CANNOT and DO not retain 100% of the original. This is not arguable.
Yeah, I've done that with numerous ECM CDs. Very often, the CDs sounds practically identical to the vinyl (not surprising, as ECM has a reputation for putting out some really good-sounding CDs).
It's so sad that after 40+ years people still don't understand the sampling theory! With 2 samples you can recreate PERFECTLY any soundwave (as at any given frequency only 1 soundwave can pass through those 2 "dots"!) Deal with it! Or study! PERIOD!
I have studied it, and your wrong. You are not sampling anything perfectly, load of toss. Do you understand ringing? Do you understand this has on your "perfect" 100% sample of the waveform? Eh whatever, its the usual amateur digital scientists in here, I got better things to do.
you don't sample perfectly but you can reconstruct it PERFECTLY. Study how you can achieve that! You're talking to someone who has a Phd in physics...who'the amateur then?! Bye
Me, I keep listening to CD, having a blast. Have several high quality CD players and dacs, to try, change and mix different sound presentations. Also I continue searching and buying the silver stuff, as there is always more music available than my musical curiosity can pursuit… and I love my huge collection. Long live CD
The keyboard you used to type that message is also sampling your keystrokes at a specific rate. Is it possible for the sampling to miss some keystrokes? Yes, but not at a rate that any human can type. The same is true with digital audio. As long as the sampling is happening often enough to capture everything a human can hear, then no information is lost.
Printing the image and viewing the Raw file are two different things. Most people post-process Raw files then have to convert to Jpeg which has less information then the Raw file. Printing depends on the quality of your printer. I can print a Raw file but I would need a high-end printer to capture the quality of the Raw file.
You are correct my programming experience is not in audio but more in Databases and networking. Most people can understand the difference between something that is sampled from the non-sampled. It comes down to logic and common sense.
Sorry, it comes down to knowledge...sampling audio is different from sampling something else! It's sampling sound waves, they behave like waves...they're predictables (their shape is) while images are not predictables
Right, enjoy both for what they have to offer. There is not much hassle in caring and feeding for LP's. Just clean them when they need to be, get off the dust and play them. The cost of new LP's is expensive but not so much for used in decent condition.
What do you think light is - series of waves and frequencies. audio, waves and frequencies. Some of the most iconic photos were captured on film not digital cameras. digital video is similar to digital audio.
Nope, you cannot use the Nyquist theory to sample an image. To capture an image you add pixels to pixels, there are shapes, colors, light intensity to measure...it's much more complicated... To capture an audio wave all you need is 2 samples at a given frequency. But this is not a physics class...please study seriously then write here. Everything else is just unfounded opinions...biased by common audio myths.
Ringing is a non-issue. No properly recorded, mixed, and mastered music will have issues with ringing. Aliasing is a different story. Recording in higher sample rates lets you use a gentler analog filter while recording to avoid aliasing artifacts.
If CD allows for the analog signal to be captured perfectly (I think this is what people have been arguing about - is 44.1 enough?), then we can think of that as the input. The full analog waveform is intact but it's digital. Now the DAC has to convert that and spit out an analog signal. How accurately is that done? We have so many DAC options, so many different approaches. Do they all fail to some extent and keep getting better (like we are told)? It just seems strange to me I guess that the ADC step is perfect but the DAC step is not. And if it is then why have so many DAC options and ever changing technology? If all DACs can spit out a perfect analog signal then it seems to me we are just finding the coloration we like best (which can also be done further down the line of course). All the music is there, it just has a slightly different presentation depending on the DAC used. I was recently comparing my Apollo to my Oppo. The Apollo is noticeably more relaxed in its presentation. Warmer, smoother, easier to listen to. But the Oppo sounds more alive and instruments have more weight and presence. There is more detail and transparency. It just feels like more of the music is coming through, but it can be a bit relentless. I do wonder if using the Rega DAC with the Apollo would bring out more detail and transparency while keeping the overall presentation smooth and relaxed. I imagine there are better DACs out there though that could probably pull that off.
Well worth noting for completeness that this is true give certain conditions -- a bandwith limited signal and a sampling frequency twice the upper bandwidth limit.
The fault is not in the CD format. It's in how it's mastered for CD and the DACs presentation of that. I have been making Cassette mix tape recordings from HiRes digital streaming and it is arguably more enjoyable and more musical. Like in jazz music perfection in timing is boring and unpleasant, but if you can deviate slightly but keep perfect time over a larger gradient, then it becomes musically pleasant.
