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| [#61] Blogs: 數碼音樂重播專欄 (1) 我打做我的 digital to analog converter by using analog device sample rate converter . it means it upsample all kinds of 16/44 or 24/96 sample rate to around 96khz. momei hing please try to listen guitar fever track 3 and track 4 cdr version and hires version with your mac  |
batmanames04 正式會員124.xxx.xxx.26 |
2011-05-03 14:39 | |
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| [#62] Blogs: 數碼音樂重播專欄 (1) momei if you like hear no big different with your dac, you need to provide low jitter source to your dac please install amarra into your mac, the software can provide much low jitter source |
batmanames04 正式會員 124.xxx.xxx.26 |
2011-05-03 14:43 |
| [#63] Blogs: 數碼音樂重播專欄 (1) what dac are you using right now? |
batmanames04 正式會員 124.xxx.xxx.26 |
2011-05-03 14:46 |
| [#64] Blogs: 數碼音樂重播專欄 (1) cpu8088, 1. Your answer suggests you are not familiar with the vapour deposition process and which of the tweeters you mentioned are made from it. You could see I asked you with an inquisitive intention to learn more. I understand Richcom and HiFi Gallery people have visited the Focal factory and I thought you might also have been there to provide answer to the question. HAVE YOU EVER BEEN THERE? 2. See if you are also able to tell us whether the fluid magnet used in the bass cones of Focal Utopia is licensed from someone or some company. I ask because Japanese Kondo is using similar technology in its speakers. 3. I do not dispute that the Focal beryllium tweeter is amongst the best if not the best itself. But then you say the Focal one is pure beryllium suggesting the Yamaha is not. Well I won't say that because I don't know. All I know is that an alloy could be better than a pure metal depending on the intended application. We all know expensive diamond necklaces and rings do not use pure gold but karat or alloy because pure gold is too soft to be suitable for the intended purpose. |
momei 正式會員 203.xxx.xxx.61 |
2011-05-03 14:48 |
| [#65] Blogs: 數碼音樂重播專欄 (1) Batmanames04, It appears to me you DAC is the main item causing the issue. If I understand it correctly, if you feed it with 16/44.1, it converts it upwards to 24/96 and either process it to DSD first or converts it directly to analogue output. If you feed it with 24/96, it parallel converts it to ITS OWN 24/96 and then process it similarly as the 16/44.1 track. In other words all the time you are listening to the same character of the DAC irrespective of whether the incoming track is 16/44.1 or 24/96. Furthermore I think if you feed it with a 320 MP3 it will upward convert it into 24/96 and then process it in same manner. Therefore you may not hear much difference either. Originally the test 2 I have in mind was for you to download from 2L two 24/192 vocal tracks and two DXD tracks from the same master. In my case I heard difference immediately between the 2 formats. Also in my case when I was first exposed to CAS, I had 5 CD tracks up converted to 24/96 by the Saracon process. I immediately heard varying degree of differences in all 5. I then had 10 more tracks converted and heard differences in all of them. Your other posts: 1. Jitter is a distortion but honestly I think it has been over-emphasized in many circles. 2. Amarra. I have tried the sample version but I grave reservations about the unwanted effects it introduces. I do not wish to step on its toes openly because after all it is so famous and is a product from Weiss. I have detailed notes prepared during my assessment of it and if you like to have them, please let me know and I will send them by PM. 3. I will listen to tracks 3 and 4 of Guitar Fever as you suggested. I did not even bother to give them a listen when they came out because I did not like the skill of the male singer nor did I think the recording of the guitar would up to my expectation and requirement. 4. I am using interface unit and DAC unit from the same manufacturer in Korea. I refrain from disclosing the name because I do not want to promote them as there are better sounding ones in the market that I have tried though much more expensive. Also there is no technical or repairs support and when problems arise (they do), the buyer got stuck. I do not want to be blamed by buyers. |
momei 正式會員 203.xxx.xxx.61 |
2011-05-03 15:32 |
| [#66] Blogs: 數碼音樂重播專欄 (1) momei here is a link about beryllium www.goodsoundclub.com/Docs/Truth_beryllium_diaphragms.doc doing vapour deposit is old technology already. i dont know anything about fluid magnet used in the bass cones of focal utopia. i understand that focal grand utopia is using electro magnetic bass drivers which is not a permanent magnet but it turns into a magnet by induction of electricity. how this termed as fluid is beyond me. the focal beryllium is in sheet form not deposits and bonded together on surface of other materials. when talking about diamond we talk about pure diamond. when we talk about diamond ring we talk about the ring made of gold or brass. 2 different matters. 最後修改時間: 2011-05-03 16:22:29 |
cpu8088 正式會員 115.xxx.xxx.214 |
2011-05-03 16:22 |
| [#67] Blogs: 數碼音樂重播專欄 (1) cpu8088, Thanks for the link to beryllium. You have so much saved in your library and I have asked the right person. I think you are right about the term of electro magnetic and the term I used is incorrect, from inappropriate Chinese term then to English. |
momei 正式會員 203.xxx.xxx.61 |
2011-05-03 16:40 |
| [#68] Blogs: 數碼音樂重播專欄 (1) """it means it upsample all kinds of 16/44 or 24/96 sample rate to around 96khz.""" should be talking about native resolution files comparison. there are n+1 upsampling methods or algorithms |
cpu8088 正式會員 115.xxx.xxx.214 |
2011-05-03 16:49 |
| [#69] Blogs: 數碼音樂重播專欄 (1) cpu8088, The document about beryllium was superb. Thanks again. Do you have any link about electro magnetic? I am curious about what happens if the owner forgets to switch on the power supply for the speakers but starts to play his music? Merely no sound or will any damage be done? Is the owner advised to have it always on, no unplugging or switching off? What is its sonic virtue over conventional permanent magnets? I guess technically the flux could be stronger or it does not lose over time. |
momei 正式會員 203.xxx.xxx.61 |
2011-05-03 17:13 |
| [#70] Blogs: 數碼音樂重播專欄 (1) Batmanames04, I listened just now to tracks 3 and 4 of Guitar Fever and the solo renditions of the same two songs and more. They are not my cup of tea. Joey Tang plays well but I do not like using electric guitars as reference because there is no standard sound out of it, affected by the controls in the guitar itself and those in the amplifier. |
momei 正式會員 203.xxx.xxx.61 |
2011-05-03 17:37 |
| [#71] Blogs: 數碼音樂重播專欄 (1) my brother like the recording much but i am the same as you just for testing |
batmanames04 正式會員 180.xxx.xxx.190 |
2011-05-03 18:13 |
| [#72] Blogs: 數碼音樂重播專欄 (1) You guys are really technical, such many discussions on going. I try to insert my comments. ＞ " can computer audio source (cas) really win cd transport for dac?" If the source is 16bit/44.1Khz, it can be no difference. CD cannot goes above 16/44.1kHz, computer can. ＞ Our human ears cannot hear more than 20khz sound , why 24.96 (around 48khz) sound better than 16/44.1? 24bit has higher resolution than 16bit, and even though there is no real benefits in terms of dynamic range, it helps really much for post processing headroom. This has nothing to do with our listening ability. On 96kHz vs. 44.1kHz, indeed there is no benefits on extend high end range, but this allows different manufacturer to design their own kind of filters, which affects quite high on quality. Well, this is actually going to be next 2 topics of CAS column, stay tune. ＞ Secondly, the Low Pass Filter roll off will push beyond 22khz to 48khz @ 96 sampling frequency, the sound deterioration near the top end (16-20Khz) of the audible range, the sound quality may improve. Right, this is a good idea. ＞ there are many theories about hearing beyond 20khz Yes, but they will not change the case that student learnt from primary school our hearing is 20Hz to around 20kHz. There is nothing evolutionary from David's paper, even though I am using his 50kHz bandwidth microphones. ＞ Could you share with us the DAC part of Devalet dpremier compare with Weiss DAC2? You cannot separate Devialet DAC's with its amplifier, the internal DAC of d-premier has unique I/V conversion which outputs high voltages in Class A. This output stage blends together as the input stage of the amplifier. The Devialet ADH technology works in DSP area to balance between the voltage output (from Class A), and current output (from Class D). In short, the Devialet works at least par with the best amplifiers (pre+power). For example DAC2 with Soulution, Boulder, Vitus, Edge etc. It is very refine and yet well control. ＞ from 20khz to 40khz there is only one octave. Octave usually describes musical note/pitch. 20kHz is not a musical note, can't really say only one octave above. ＞ Similarly to 200w to 400w is just 3db difference. not much at all. It is not as simple as this. ＞ The data size doesn't directly reflect the amount of musical information, a 16/44 recording of piano have more musical information than a 24/192 recording of "silence", but the size is much smaller. The data size only reflect the space occupied for storing sample, more room for more information, a store room can store a bicycle or tyres of the MPV but not a MPV, but a big garage can store many bicycle with one or two MPV at the same time. I agree. It is very difficult to say that HiRes is better than CD if they are not engineer in the same process. I can use the most colorful paint to draw picture, but never can compete with Gogh using the worse paint. ＞ why is it so hard to hear similarly in audio with upsampling of 16/44 format to hi rez to approximate true hi rez? It may be difficult with iPOD listening, it may not be difficult for high end system playback. But again listening is very subjective. It is true and only ture to the listener him/herself. ＞ quoted from BobKatz (great recording engineer) Mastering audio the art and the science the ultimate listening test: is it the filtering or the bandwidth? Bob done more than this. He has done extensively on the AES panel and meeting. The case is the filter at high freq. affects the listening, not the extra bandwidth. ＞ My DIY DAC hear almost no different between 16 44.1 and 24/96 material if they are derived from the same source. The source is originally 16/44.1 or 24/96? I can hear differences but that's just me. See my above comment. ＞ if you believe nyquist theorm, a good dac should hear no big different bewteen 16 44.1 and 24 96 material ( provide they are from the same source) No one can yet challenge Nyquist for sure. ＞ the view of Bill Schnee to balance that of Bob Katz who is associated with Daniel Weiss. Bill advocates 24/192. In my book I have heard and read about Bill Schnee more than Bob Katz though I do not know in reality who is more famous. They are both being at the top hill of the audio world. Bill is more practically Bob is well balance between listening and theory. Bill is really getting there to have AES spec. for Blu-ray audio. I talked with him during 2009 and he has been doing so. I am on another camp that computer file will be the one who works in high end audio. ＞ try to audition a high end DAC, your ears can hardly distinguish 16 44.1 and 24 96 recordings This has some grounds. I think I understand where you are heading to. But I can only agree if these are 2 different recordings. For the same one, I can hearing quite a big differences. ＞ I think that it is better to record the same sound with original file in 16bit/44.1Khz and 24bit/96khz into a CD then play it by a CDP. If the originally recording is 16bit/44.1kHz, you record this source with 24/96kHz, you will not gain any improvement, usually worse. ＞ there are reasons why a hi-rez recording will sound different. 1) Down sampling might introduce artifacts. 2)our tweeters can sound different when modulated with signals 20 kHz. I agree with point 1, but I cannot hear above 20kHz. Then I read many lines about listening. You all have to beware that there is no right or wrong in listening. Taste is the matter. Everyone has his/her own like or dislike on various sounds and of course music. It is great to share but no conclusion can be made. |
DWS 正式會員 203.xxx.xxx.150 |
2011-05-03 19:23 |
| [#73] Blogs: 數碼音樂重播專欄 (1) Thanks Kent for going through some many questions and providing answers. TWO MORE QUESTIONS for you please: 1. The benefit of using a fatter slope filter in higher resolution tracks such as 24/96, this I have no problem in understanding. But you still have the track to convert to 16/44.1. When carrying out such conversion what tools do you use? Filters again, algorithm (for example, Nuendo), individual sections of tracks one by one, or a combination of them? Take the example of the Wong Wai Man album that according to the description the native tracks were recorded at 24/96. How did you convert it to 16/44.1? 2. I note from the photo a machine on the left hand side that looks like a tape recorder. Do you still have jobs for working with tapes such as converting tracks on tapes to digital for release as CD? |
momei 正式會員 203.xxx.xxx.61 |
2011-05-03 22:40 |
| [#74] Blogs: 數碼音樂重播專欄 (1) I believed Kent still using trusty Weiss SARACON for redithering and sampling rate conversion, which support DXD and DSD conversion to CD resolution. http://www.weiss.ch/p2d/p2d.html |
hercules 正式會員 112.xxx.xxx.58 |
2011-05-03 23:20 |
| [#75] Blogs: 數碼音樂重播專欄 (1) good info, thanks for sharing |
cubixx 正式會員 210.xxx.xxx.224 |
2011-05-03 23:38 |
| [#76] Blogs: 數碼音樂重播專欄 (1) studio use 24/96 analog to digital converter for pcm recording? |
batmanames04 正式會員 203.xxx.xxx.117 |
2011-05-04 09:24 |
| [#77] Blogs: 數碼音樂重播專欄 (1) ＞ 1. The benefit of using a fatter slope filter in higher resolution tracks such as 24/96, this I have no problem in understanding. 96kHz provides 48kHz bandwidth. We can design much smoother filter from around 20kHz to 48kHz. Compares in 44.1kHz, we need to design a very steep filter from 20kHz to 20.5kHz. ＞ But you still have the track to convert to 16/44.1. Only if you are making a CD. In CAS you do not need to. One of the very big advantages of CAS. ＞ When carrying out such conversion what tools do you use? Filters again, algorithm (for example, Nuendo), individual sections of tracks one by one, or a combination of them? Take the example of the Wong Wai Man album that according to the description the native tracks were recorded at 24/96. How did you convert it to 16/44.1? Nuendo is a recording/mixing software. Our approach is to record everything in 24/96 or above, and then throughout the whole post production procedure to maintain in the highest resolution. Sometimes it is higher than 24bit because some of the post processing (edit, fades, individual tracks volume, automation) are performed in 32bit or 64bit floating point. After mixing procedure, we have a stereo mixes in 24/96 or higher. We use this mix for mastering also in highest resolution. After the mastering is done, we have a hiRes master files. 1) If the recording is intended for CD production, we need to convert it back to 16bit/44.1kHz (Honestly I lot of engineers do not know there is a procedure require to make here. They just save the file as 16bit/44.1kHz) 2) If the recording is intended for DVD, Bluray, we have HiRes native master files. 3) If the recording is intended for SACD, we need to convert it to DSD too. The above 1), 2) & 3) procedure requires attention on listening because there is no technical one way ticket. We have quite some tools for SRC (sampling rate conversion), but these days we just use one solution as hercules mentioned. ＞ 2. I note from the photo a machine on the left hand side that looks like a tape recorder. Do you still have jobs for working with tapes such as converting tracks on tapes to digital for release as CD? Yes, that is the Studer A820. The Roll Royce of analogue open reel transport. Digitizing analogue open reel and restoration is an expertise for specialist. And people know how much I love analogue sound. |
DWS 正式會員 203.xxx.xxx.150 |
2011-05-04 10:59 |
| [#78] Blogs: 數碼音樂重播專欄 (1) Many thanks Kent for the detailed answers. |
momei 正式會員 203.xxx.xxx.61 |
2011-05-04 11:10 |
| [#79] Blogs: 數碼音樂重播專欄 (1) ＞ I believed Kent still using trusty Weiss SARACON for redithering and sampling rate conversion, which support DXD and DSD conversion to CD resolution. Yup. ad time. =) Quotations: George Massenburg "Now That I've used it on everything I do, it is the only resampler I can listen to..." Bob Katz "Saracon is not only the world's best audio sample rate converter. It's also a great batch renamer, format converter, interleaver, de-interleaver, word length reducer and all-round utility tool!" Gus Skinas who in charges Super Audio Center in USA using it for SACD production. Okay besides AD, let's go back to technical. That was around 2001 Weiss design team Uli Franke and Rolf Anderegg have developed SFC3 (Weiss pro catalogue has a hardware SFC2 box for 4ch. SRC). The SFC3 intends to cover DSD conversion because the popularity of SACD during that period of time. Later on the computer native (CPU) grows so faster and the team designed to produce its first software. SA (sampling) RA (rate) CON (conversion) By using native software solution. The disadvantage of the program is not a real time process. Actually it can be faster than real time depends on the CPU speed. This means as times goes by, the faster CPU, the faster processing speed. Another advantage of the program is the purely digital calculation inside the computer, it surpasses the limitation of AES/EBU (24bit max.) and provides as low as -180dBFS THD. One of the earlier paper is open to public: http://freerider.dyndns.org/anlage/Diplomarbeit-DSD2PCM.pdf After I started using SARACON, I sold out all other SRCs in my lab. ＞ studio use 24/96 analog to digital converter for pcm recording? They surely have the ability to do so. But most are recorded in 24bit/48kHz, some do 24/96, rarely some work in 24/192. |
DWS 正式會員 203.xxx.xxx.150 |
2011-05-04 11:20 |
| [#80] Blogs: 數碼音樂重播專欄 (1) a photo for you guys to relax  |
batmanames04 正式會員 61.xxx.xxx.217 |
2011-05-04 15:53 |
