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JEJ 에게 이메일을 썼쑴미다
HIFI게시판 > 상세보기 | 2012-05-08 13:49:47
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JEJ 에게 이메일을 썼쑴미다

글쓴이

김영민 [가입일자 : 2004-04-01]
내용
Related Link: http://www.hometheaterhifi.com/contact-us.html

정확히는 위 홈페이지 에디터에게 썼구요. JEJ 는 이메일이 없길래..



제가 기술에 문외한임에도 이번에 배우게 된 내용을 쥐어짜서 다음과 같이 보냈는데, 고칠 내용이 없나 함 봐주세요.

-----------

Hi. I'm Young Min Kim, one of audiophiles in (South) Korea.



I'm writing this e-mail for a question about an article on your homepage.



Actually I must say I'm NOT fluent about technical terms at all. But I came to understand a little bit about technical things thanks to an explosive argue about an article by your editor JEJ, in an on-line audiophile community in Korea.



Below is the article

-----------------------------------------------------------------------

http://www.hometheaterhifi.com/technical-articles/427-a-secrets-technical-article.html?start=3



"I tested some 10 kHz and 20 kHz sine waves that were recorded at several word lengths (16 bit or 24 bit) and sampling frequencies (44.1 kHz, 96 kHz, 192 kHz), analyzing them in a software sequencer."



"The dots represent finite voltage values that are fed in sequence as a stream to the DAC, which then produces a stair-stepped output, after which a low-pass reconstruction filter smooths out the signal. What I want you to notice is how jagged the lines are at standard Redbook CD 16/44. The DAC and reconstruction filter's job is to make these jagged lines more sinusoidal, so that it will be like the music that was recorded, which is also sinusoidal."



Fig 1. http://www.hometheaterhifi.com/images/stories/april-2008/vinyl-vs-cd-10-khz-sine-wave-16-44---16-96---16-192---24-44---24-96---24-192-large.gif



"The dots represent finite voltage values that are fed in sequence as a stream to the DAC, which then produces a stair-stepped output, after which a low-pass reconstruction filter smooths out the signal. What I want you to notice is how jagged the lines are at standard Redbook CD 16/44. The DAC and reconstruction filter's job is to make these jagged lines more sinusoidal, so that it will be like the music that was recorded, which is also sinusoidal."



Fig.2 http://www.hometheaterhifi.com/images/stories/april-2008/vinyl-vs-cd-20-khz-sine-wave-16-44---24-96---24-192-large.gif

-----------------------------------------------------------------------



and JEJ made a comment like below

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written by JEJ , October 04, 2009

A Couple of Points said - And JEJ, it's really quite misleading to show DAC output before reconstruction filtering had done its job. NO ONE hears the sine waves shown above. -

It's not misleading. You just scanned the paragraphs too quickly. I specifically mentioned in paragraph 5 - The lines connecting the dots represent the signal that is fed to the output stage before any filters are applied. - and - The filter's job is to make these jagged lines more sinusoidal, so that it will be like the music that was recorded, which is also sinusoidal.

-----------------------------------------------------------------------



Here's the point.



Many guys, many of whom are engineers, are talking that the explanation above about DAC processing is quite misleading, and not based on exact concept about digital audio processing. They say that first figure in Fig. 2 shows one "generated" by a program, and the lines connecting the dots is just simply connecting the dots and shows NOT the sinusoidal waveform, NOR the reconstructed wave. And because JEJ is explaining on the comment below that "the lines connecting the dots" are fed to the DAC, and go through the fillter, JEJ is making quite a mistake about DAC processing and Nyquist theorem.



That may be the exactly the same point raised by a guy called Joshua on a comment below

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Re: Straight Lines

written by Josuah , October 08, 2009

I don't believe the graphs are true representations of the signal before the low-pass filter because the dots were connected with straight lines instead of connecting them using the mathematical formulas described on the Wikipedia page. Specifically the last bullet point under "Mathematical basis for the theorem" which I quoted earlier, with respect to the Dirac comb function.

http://en.wikipedia.org/wiki/ Nyquist–Shannon_sampling_theorem#Mathematical_basis_f

or_the_theorem

44.1kHz does not lack the required information for sound reproduction below 22kHz, but you can deconstruct/reconstruct more accurately if your process is performed at a higher sample rate. It is somewhat analogous to performing a chain of multiplication and division operations using decimal places even though your original numbers are integers. By using the decimal places, you help avoid rounding errors. This is mentioned under the "Practical considerations" section of the Wikipedia page.

http://en.wikipedia.org/wiki/ Nyquist–Shannon_sampling_theorem#Practical_considerat

ions

Dan Lavry has a nice paper about this:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

-----------------------------------------------------------------------



But, Another guy in our community is having another explanation. He says that since JEJ said he tested some sine waves that were "recorded" (rather than generated), JEJ might have used some waves he himself recorded using ADC and the filter may refer to a kind of ADC filter, NOT the low-pass reconstruction filter.



most of the guys I mentioned above are apparently criticizing strongly about the explanation above. and the conversation is, actually many times somewhat.. harsh !



So, I thought I need to request the author to answer this issue.



In summary, the point is whethere



1. JEJ made a mistake about the DAC processing concept. He tested some sine waves "generated" by a program, and the dots are actually the signals fed to the DAC but the lines connecting the dots is never the signal and means nothing.



or



2'. JEJ tested some sine waves he himself recorded using an ADC or digital sound signals, and the the lines connecting the dots represent the signals without any kind of recording filter.



Many guys say 1 is the case, and one guy says 2 may be the case.



and the guy insisting No.2 asked me to add another question, which is below



-------------

I'm very confused now. Please let me know what kind of 'FILTER' you talked about. Is that a kind of anti-aliasing filter used in the recording stage or a low-path filter included in the DAC machine?



You said, "I think that if we were to go to 500 kHz sampling, 24 bit, no 'FILTER' would be necessary because typical studio microphones don't respond beyond 20 kHz, and some even roll off at 15 kHz."

-----------



Please help end our long-lasting, sometimes boring or harsh argue in a positive way. Thank you .



Sincerely.
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김영선 2012-05-08 14:00:10
답글

고생하셨습니다. 답변을 받으시면 좋겠네요-<br />

김영민 2012-05-08 14:07:51
답글

문법이나 표현을 조금 고쳐서 수정하였습니다.

김영선 2012-05-08 14:18:54
답글

huge mistake 는 조금 심한 듯 하기도 하네요. 걍 mistake 정도로 하시는게 어떨까 싶고.

이종남 2012-05-08 14:34:27
답글

2. JEJ tested some sine waves he himself recorded using an ADC, and the the lines connecting the dots represent the signals through a kind of ADC filter. <br />
<br />
를 이런식으로 바꾸었으면 합니다.. <br />
<br />
2. JEJ tested some sine waves he himself rec

이종남 2012-05-08 14:35:50
답글

recording filter를 recording anti-aliasing filter로 쓰면 더 확실하겠네요...

이종남 2012-05-08 14:53:54
답글

하나 더 추가 하자면..<br />
<br />
You said that "I think that if we were to go to 500 kHz sampling, 24 bit, no 'FILTER' would be necessary because typical studio microphones don't respond beyond 20 kHz, and some even roll off at 15 kHz". <br />
<br />

김영민 2012-05-08 15:17:22
답글

이종남님/ 수정하였으면 좋겠다고 제시해주신 2번 문장의 digital sound signals 라는 게 어디에 걸리는 건지 명확하지 않아서.. 말씀하신 대로라면 sine wave 에 병치되는 문장구조인데, 즉<br />
JEJ tested some<br />
ㄱ. sine waves he himself recorded using an ADC<br />
ㄴ. or digital sound signals<br />
라는 뜻인데, 맞는건지요?

이종남 2012-05-08 15:26:06
답글

잘 쓰신 것 같습니다.. 가장 핵심적인 것만... 찝어 주신 것 같고요...<br />
<br />
저도 메일을 한번 쓸까 하다가.. 아무래도 객관성이 떨어진다는 소리가 나올까 망설이고 있었습니다..<br />
솔직히.. 귀찮기도 하고요... <br />
<br />
그나저나 가끔 보면 김영민님. 저보다도 더........ 집요한데가 있어요... ^^

조은석 2012-05-08 15:33:50
답글

I'm very confused now. Please let me know what kind of 'FILTER' you talked about. Is that a kind of anti-aliasing filter used in the recording stage or a low-path filter included in the DAC machine?

조은석 2012-05-08 15:34:38
답글

You said, "I think that if we were to go to 500 kHz sampling, 24 bit, no 'FILTER' would be necessary because typical studio microphones don't respond beyond 20 kHz, and some even roll off at 15 kHz." <br />

이종남 2012-05-08 15:38:20
답글

흠.. 게다가. 문법적으로도 완벽한.. 좋은 문장을 조은석님이 만들어 주셨네요....

김영민 2012-05-08 15:45:15
답글

다음과 같은 내용을 서두에 추가하고, 고쳐서 다시 보냈습니다. 설마 스팸처리 안되겠죠? 고친 내용은 본문을 다시 수정하였습니다. 영어공부하자는 의미도 있으니 뭐든지 사소한 문법실수도 지적해 주심 감사하겠습니다. 제 직종이 영어도 잘 하면 좋은 직종이라..<br />
---<br />
Sorry to bother you again !<br />
<br />
The guy who are insisting the summary No.2 wh

김영민 2012-05-08 15:54:37
답글

이종남님/ 저는 판을 굉장히 좁히고 명료하게 하는 데 집요한 편이랍니다. 키우면 감당을 못 하거든요..

민영기 2012-05-08 16:30:09
답글

넓히고 좁히고...<br />
<br />
현 상황에 딱 맞는 표현이네요. 영민님도 핵심을 잘 짚으시는걸 보니 전보다 훨 예리해지신 듯 합니다.<br />
보기가 좋습니다. ^^

김현희 2012-05-08 16:56:15
답글

이론에 약하고 지엽적이기도 한 토론에 김영민님의 정리가 정주행하는데 큰 도움이 되고 있네요. ^^

최규호 2012-05-08 17:47:37
답글

김영민 님의 수고로 메일을 보내게 되었네요. <br />
<br />
영어 잘하는 분들을 뵈면 참 부럽습니다.<br />
<br />
감사합니다. 수고하셨습니다.

이승태 2012-05-08 20:51:53
답글

수고하셨습니다...^^ <br />
<br />
열정도 열정 나름이라는 생각이 드는군요.

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