真是不可思議…，USB線只有傳遞數位訊號吧?
這麼說，如果有Shark純銀USB2.0線或者古河PCOCC 6N銅做的USB2.0線改進會更多囉?....
不知道前輩們有沒有這種經驗
AndAudio.com
聊耳機、品音樂、論音響、玩電子DIY以及哈拉風花雪月的地方有關於USB2.0傳輸線
23 篇文章
• 第 1 頁 (共 1 頁)
有關於USB2.0傳輸線
由 kalimdor 發表於 週二 6月 01, 2004 12:08 am
今天買了一條powersync的USB2.0線(189元)，號稱「鍍金接頭」，run了8小時後，和KECES廠機附的USB線A/B test，居然發現powersync USB 2.0線高頻較為豐富、人聲泛音較多、顆粒較細緻、高頻延伸較好…
真是不可思議…，USB線只有傳遞數位訊號吧?
這麼說，如果有Shark純銀USB2.0線或者古河PCOCC 6N銅做的USB2.0線改進會更多囉?....
不知道前輩們有沒有這種經驗
真是不可思議…，USB線只有傳遞數位訊號吧?
這麼說，如果有Shark純銀USB2.0線或者古河PCOCC 6N銅做的USB2.0線改進會更多囉?....
不知道前輩們有沒有這種經驗
-
kalimdor - SR40
- 文章: 0
- 註冊時間: 週日 5月 11, 2003 8:21 pm
Re: 有關於USB2.0傳輸線
由 liu168 發表於 週二 6月 01, 2004 5:29 pm
kalimdor 寫:今天買了一條powersync的USB2.0線(189元)，號稱「鍍金接頭」，run了8小時後，和KECES廠機附的USB線A/B test，居然發現powersync USB 2.0線高頻較為豐富、人聲泛音較多、顆粒較細緻、高頻延伸較好…
真是不可思議…，USB線只有傳遞數位訊號吧?
這麼說，如果有Shark純銀USB2.0線或者古河PCOCC 6N銅做的USB2.0線改進會更多囉?....
不知道前輩們有沒有這種經驗
一切都是jitter/noise惹的禍........
- liu168
- SR40
- 文章: 0
- 註冊時間: 週六 4月 24, 2004 12:59 am
由 Luker-Chen 發表於 週二 6月 01, 2004 7:44 pm
狂人 寫:與其說是數位訊號，不如說是硬把類比訊號扭成數位訊號的形狀。
呵呵。
狂人果然是博學......
形容的蠻貼切的,最近才從一個中山的本科網友問出類似的結論!!
確實是一種"脈衝式"的訊號
是會有衰減的問題的!!
不是方波!!
- Luker-Chen
- SR40
- 文章: 0
- 註冊時間: 週三 4月 30, 2003 11:38 am
- 來自: 台北大同
由 yanyufon 發表於 週三 6月 02, 2004 10:51 am
恩..... 
還是不懂, 小弟看著傅利葉級數的公式, 照理在抵達DAC之前的數位訊號中唯一有可能影響的應該只有jitter(time shift) 不是嗎? 其它的應該是到了DAC後做傅利葉轉換時才會將訊號還原, 而還原的signal length 是已經決定了的, 不是0便是1, 除非訊號出錯, 不然轉換出來的波型應該要是一樣的不是嗎? 抱歉小弟只修了大學的DSP 而且只有一點理論, 實作上還有什麼樣的變數會影響了數學轉換波型還請大大指導一下, 小弟真的蠻有興趣的.
還是不懂, 小弟看著傅利葉級數的公式, 照理在抵達DAC之前的數位訊號中唯一有可能影響的應該只有jitter(time shift) 不是嗎? 其它的應該是到了DAC後做傅利葉轉換時才會將訊號還原, 而還原的signal length 是已經決定了的, 不是0便是1, 除非訊號出錯, 不然轉換出來的波型應該要是一樣的不是嗎? 抱歉小弟只修了大學的DSP 而且只有一點理論, 實作上還有什麼樣的變數會影響了數學轉換波型還請大大指導一下, 小弟真的蠻有興趣的.-
yanyufon - SR40
- 文章: 0
- 註冊時間: 週二 2月 11, 2003 7:51 am
- 來自: USA
由 kalimdor 發表於 週三 6月 02, 2004 2:20 pm
JY 寫:我覺得更無法解釋的在於PCM2702的USB傳輸並非即時的...
在buffer過之後的數位(姑且這樣稱USB)訊號還會被線材影響..
我想不是心理作用就是原本的線已經到產生error的程度了..
小弟可以肯定不是心理作用
要求證也很容易，這條線才190元，隨便都買的到。
當然了，我也沒辦法保證每個人都聽的出來
(每個人的線材、電力供應、環境都不同)
至少在小弟的系統上，人聲較通透，高頻的光澤較好…
至於這個「較好」是好多少，就見人見智了。
-
kalimdor - SR40
- 文章: 0
- 註冊時間: 週日 5月 11, 2003 8:21 pm
由 kalimdor 發表於 週三 6月 02, 2004 3:55 pm
liu168 寫:JY 寫:我覺得更無法解釋的在於PCM2702的USB傳輸並非即時的...
在buffer過之後的數位(姑且這樣稱USB)訊號還會被線材影響..
我想不是心理作用就是原本的線已經到產生error的程度了..
那要看看當時usb的工作模式了
有bulk,asyn啥的好像三四種.....有一種是允許error的
我想到的是,這dac的電力吃誰的? usb嗎?
有沒有可能是電源線改善造成的效果?
UAB像一般的DAC一樣外接電源線，USB線只有傳輸數位訊號
當然了，為了好聲，小弟也餵給UAB一條古河的FS-4BTW電源線
，電源線帶給UAB的好處是無庸置疑的，奇怪的是換USB線也有這種類似的效果…-
kalimdor - SR40
- 文章: 0
- 註冊時間: 週日 5月 11, 2003 8:21 pm
由 kaylen 發表於 週三 6月 02, 2004 5:21 pm
JY 寫:我覺得更無法解釋的在於PCM2702的USB傳輸並非即時的...
在buffer過之後的數位(姑且這樣稱USB)訊號還會被線材影響..
我想不是心理作用就是原本的線已經到產生error的程度了..
嗯...最近才剛接觸過有關USB電氣訊號的傳輸方式，
我也是覺得很奇怪...，雖然是以0/1傳送的，卻有經過
通訊協定的封裝，每個傳送的資料之前會先以packet包裝過，
然後拆分成0/1的訊號傳遞至目的端，再將packet解開，
這樣線材會對聲音有影響
- kaylen
- SR125
- 文章: 504
- 註冊時間: 週五 2月 16, 2001 6:21 pm
- 來自: 台北的角落
It's the whole things about jitter.
由 yanyufon 發表於 週四 6月 03, 2004 2:36 am
昨天花了幾個鐘頭找讀了一些文章, 果然是這唯一的傅利葉變數T
所帶來的問題, 也就是Jitter. 影響jitter 的層面非常的廣, 其中影
響CD讀取最大的除了振動外資料讀取後訊號晶片上的邏輯閘rise
and fall time 容忍度反而成了jitter 最大來源, 也就是說晶片的
layout 設計與供應的電源穩定度非常重要. 晶片的Si 純度與邏輯
設計不是我們所能控制的, 但穩定的電源設計可以幫助降低Jitter.
在USB上面這部份在前端訊號已經資料化了(其實硬碟的讀取還是
會造成Jitter) 也就不論, 不過在信號傳輸到USB SAC 時就像類比
一樣外界的電波電磁都會造成在rise & fall time 容忍範圍的位移,
結果一樣造成Jitter, 另外USB線除了兩條信號線還有供電接地線,
就Audio 來說這供電線與信號線同時平行包在一起是最不好的(電磁
感應), 所以就理論來說如果能將這條線斷接或者與GND線並在一起,
DAC由外部供電會有較好的效果. 而信號線有如類比線一樣是AWG# 越低越好(高頻寬).
結論是不管你怎麼搞都會有Jitter, 把音響分的越細(Ex. CDT+DeJitter+DAC+PreAmp+Amp)
你的Jitter 受到影響的機率越大, 不過反過來想, 就算是CD在錄音
時過了電子器材一樣歌手的聲音頻響也早就改的一蹋糊塗了, 再往
前想一點, 歌手每次唱歌哪有每次都一樣的, 既然如此那我們幹麻
還要去在乎Jitter 的準確度呢? 把它當作一種調音的手段就好了,
聲音是感性的, 說不定你心情壞的時候大的Jitter 反而讓你心有戚戚焉 !?
亂說的. 
以上只是小弟縱合找到的資料整理消化後的感想, 並非論文, 裡面有
不少小弟並未完全理解其理論(那會搞死小弟 ), 有任合觀念上
的錯誤還請各位大大指教, 也希望能給大家一些有用的 idea.
所帶來的問題, 也就是Jitter. 影響jitter 的層面非常的廣, 其中影
響CD讀取最大的除了振動外資料讀取後訊號晶片上的邏輯閘rise
and fall time 容忍度反而成了jitter 最大來源, 也就是說晶片的
layout 設計與供應的電源穩定度非常重要. 晶片的Si 純度與邏輯
設計不是我們所能控制的, 但穩定的電源設計可以幫助降低Jitter.
在USB上面這部份在前端訊號已經資料化了(其實硬碟的讀取還是
會造成Jitter) 也就不論, 不過在信號傳輸到USB SAC 時就像類比
一樣外界的電波電磁都會造成在rise & fall time 容忍範圍的位移,
結果一樣造成Jitter, 另外USB線除了兩條信號線還有供電接地線,
就Audio 來說這供電線與信號線同時平行包在一起是最不好的(電磁
感應), 所以就理論來說如果能將這條線斷接或者與GND線並在一起,
DAC由外部供電會有較好的效果. 而信號線有如類比線一樣是AWG# 越低越好(高頻寬).
結論是不管你怎麼搞都會有Jitter, 把音響分的越細(Ex. CDT+DeJitter+DAC+PreAmp+Amp)
你的Jitter 受到影響的機率越大, 不過反過來想, 就算是CD在錄音
時過了電子器材一樣歌手的聲音頻響也早就改的一蹋糊塗了, 再往
前想一點, 歌手每次唱歌哪有每次都一樣的, 既然如此那我們幹麻
還要去在乎Jitter 的準確度呢? 把它當作一種調音的手段就好了,
聲音是感性的, 說不定你心情壞的時候大的Jitter 反而讓你心有戚戚焉 !?
亂說的.
以上只是小弟縱合找到的資料整理消化後的感想, 並非論文, 裡面有
不少小弟並未完全理解其理論(那會搞死小弟
), 有任合觀念上
的錯誤還請各位大大指教, 也希望能給大家一些有用的 idea.
-
yanyufon - SR40
- 文章: 0
- 註冊時間: 週二 2月 11, 2003 7:51 am
- 來自: USA
由 yanyufon 發表於 週四 6月 03, 2004 2:38 am
這篇是非常有用的資料, 建議有興趣的讀一下 
http://www.audioasylum.com/audio/tweaks ... /1582.html
*How can digital interconnect cables from a CD transport to an outboard DAC possibly affect the sound? Isn't digital audio just one's and zero's?
A very common misconception about digital signal
transmission with respect to audio is that if the signal does
not get corrupted to the point of losing or changing the 1's
and 0's, that nothing else can go wrong. If the transmission
system had been designed with cost no object, and by
engineers familiar with all the known foibles and problems
of digital transmission of audio signals, then this might
be subtantially true. No differences could rear their ugly
head.
Unfortunately, the systems we ended up with DO NOT remain
unaffected by such things as jitter, where the transistion
from a 1 to a 0 is modulated with respect to time. There are
many ways that jitter can affect the final digtial to analog
conversion at the DAC. Jitter on the transmitted signal can
bleed or feed through the input reciever, and affect the DAC.
How? Current drain on the power supplies due to the changing
signal content and the varying demands made on the power
supply to the logic chips and the DAC. Modulate the power
supply rails, and the DAC will convert at slightly different
times. This is due to the fact that a logical one or zero is detected by the signal swinging through a regoin from around zero volts to around 5 volts. The digital logic chips detect the change at a specific PERCENTAGE of that total voltage swing.
HOWEVER the power supply gets modulated, it will
affect the DAC. One version of this has been popularly
refered to as LIM or Logic Induced Modulation by the
audiophile press. See:
"Time Distortions Within Digital Audio Equipment Due to Integrated Circuit Logic Induced Modulation Products"
AES Preprint Number: 3105 Convention: 91 1991-10
Authors: Edmund Meitner & Robert Gendron
Many of the logic chips in a digital audio system behave
very poorly with respect to dumping garbage onto the rails
and even worse, onto the ground reference point. Even as I
post, logic manufacturers such as TI are advertising the
benefits of their latest generation of logic chips that
reduce ground bounce. The circuitry itself generates it's
own interference, and this can be modulated by almost
anything that also affects the power supply or ground.
The amount of jitter that it takes to affect the analog
output of the signal used to be thought of as fairly high,
somewhere on the order of 1,000 to 500 pS worth. Now, the
engineers on the cutting edge claim that in order for jitter
to be inaudible and not affect the sound of the signal, it
may have to be as low as 10 to 20 pS. That's for 16 bit
digital audio. That's a very tiny amount of jitter, and
easily below what most all current equipment is capable of.
Computer systems never convert the 1's and 0's to time
sensitive analog data, they only need to recover the 1's
or 0's, any timing accuracy only has to preserve the bits,
not how accurately they arrive or are delivered. So in this
regard, computer systems ARE completely different than
digital audio systems.
Look into digital audio more thouroughly, and realize that
the implementations are not perfect or ideal, and are
sensitive to outside influences. Just because they could
have been and should have been done better or more nearly
perfect does not mean they were! People are not hearing
things, they are experiencing the result of products designed
to a cost point that perform the way they do in a real
world because of design limitations imposed by the consumer
market price conciousness all the mid-fi companies live and
die by.
With digital cables, there are three things that are paramount:
proper impedance, proper cable termination, and wide bandwidth.
It may be that a particular cable more nearly matches a systems
actual impedance. The other factor, proper termination includes,
but is not limited to, the actual electrical termination inside
the components, as well as the connector on the end of the
cable. If the connector is NOT a perfect 75 ohm, 110 ohm, or
whatever, it will cause minor reflections in the cable, which
makes our old friend JITTER raise it's ugly head again.
The third factor, bandwidth, is only an issue because both the
AES/EBU and the SP/DIF interface formats were designed before
Sony/Phillips knew all there was to know about digital problems,
and they require PERFECT unlimited bandwidth cables in order for
the transimission systems to be free of jitter. The more you limit
the bandwidth, the more jitter. This is a known engineering
fact, and an AES paper was given about this very subject not
too long ago.
"Is the AES/EBU/SPDIF Digital Audio Interface Flawed?"
AES Preprint Number:3360 Convention: 93 1992-10,
Authors: Chris Dunn & Malcolm O. J. Hawksford
The effective data rate of SP/DIF is about 3 Mhz, and the
design of the transmitters and recievers is abysmal. Maybe
if everything else was done right, then cables, etc. wouldn't
matter. So much was done wrong or cost cut till it screwed
up that they do come into the picture.
A good web source for info on jitter is located at:
http://www.digido.com/jitteressay.html
AND:
http://www.audioprecision.com/publicati ... tm#Digital Audio Transmission
I recommend a couple of options for digital interconnect cables in my DIY Interconnect Note.
http://www.audioasylum.com/audio/tweaks ... /1582.html
*How can digital interconnect cables from a CD transport to an outboard DAC possibly affect the sound? Isn't digital audio just one's and zero's?
A very common misconception about digital signal
transmission with respect to audio is that if the signal does
not get corrupted to the point of losing or changing the 1's
and 0's, that nothing else can go wrong. If the transmission
system had been designed with cost no object, and by
engineers familiar with all the known foibles and problems
of digital transmission of audio signals, then this might
be subtantially true. No differences could rear their ugly
head.
Unfortunately, the systems we ended up with DO NOT remain
unaffected by such things as jitter, where the transistion
from a 1 to a 0 is modulated with respect to time. There are
many ways that jitter can affect the final digtial to analog
conversion at the DAC. Jitter on the transmitted signal can
bleed or feed through the input reciever, and affect the DAC.
How? Current drain on the power supplies due to the changing
signal content and the varying demands made on the power
supply to the logic chips and the DAC. Modulate the power
supply rails, and the DAC will convert at slightly different
times. This is due to the fact that a logical one or zero is detected by the signal swinging through a regoin from around zero volts to around 5 volts. The digital logic chips detect the change at a specific PERCENTAGE of that total voltage swing.
HOWEVER the power supply gets modulated, it will
affect the DAC. One version of this has been popularly
refered to as LIM or Logic Induced Modulation by the
audiophile press. See:
"Time Distortions Within Digital Audio Equipment Due to Integrated Circuit Logic Induced Modulation Products"
AES Preprint Number: 3105 Convention: 91 1991-10
Authors: Edmund Meitner & Robert Gendron
Many of the logic chips in a digital audio system behave
very poorly with respect to dumping garbage onto the rails
and even worse, onto the ground reference point. Even as I
post, logic manufacturers such as TI are advertising the
benefits of their latest generation of logic chips that
reduce ground bounce. The circuitry itself generates it's
own interference, and this can be modulated by almost
anything that also affects the power supply or ground.
The amount of jitter that it takes to affect the analog
output of the signal used to be thought of as fairly high,
somewhere on the order of 1,000 to 500 pS worth. Now, the
engineers on the cutting edge claim that in order for jitter
to be inaudible and not affect the sound of the signal, it
may have to be as low as 10 to 20 pS. That's for 16 bit
digital audio. That's a very tiny amount of jitter, and
easily below what most all current equipment is capable of.
Computer systems never convert the 1's and 0's to time
sensitive analog data, they only need to recover the 1's
or 0's, any timing accuracy only has to preserve the bits,
not how accurately they arrive or are delivered. So in this
regard, computer systems ARE completely different than
digital audio systems.
Look into digital audio more thouroughly, and realize that
the implementations are not perfect or ideal, and are
sensitive to outside influences. Just because they could
have been and should have been done better or more nearly
perfect does not mean they were! People are not hearing
things, they are experiencing the result of products designed
to a cost point that perform the way they do in a real
world because of design limitations imposed by the consumer
market price conciousness all the mid-fi companies live and
die by.
With digital cables, there are three things that are paramount:
proper impedance, proper cable termination, and wide bandwidth.
It may be that a particular cable more nearly matches a systems
actual impedance. The other factor, proper termination includes,
but is not limited to, the actual electrical termination inside
the components, as well as the connector on the end of the
cable. If the connector is NOT a perfect 75 ohm, 110 ohm, or
whatever, it will cause minor reflections in the cable, which
makes our old friend JITTER raise it's ugly head again.
The third factor, bandwidth, is only an issue because both the
AES/EBU and the SP/DIF interface formats were designed before
Sony/Phillips knew all there was to know about digital problems,
and they require PERFECT unlimited bandwidth cables in order for
the transimission systems to be free of jitter. The more you limit
the bandwidth, the more jitter. This is a known engineering
fact, and an AES paper was given about this very subject not
too long ago.
"Is the AES/EBU/SPDIF Digital Audio Interface Flawed?"
AES Preprint Number:3360 Convention: 93 1992-10,
Authors: Chris Dunn & Malcolm O. J. Hawksford
The effective data rate of SP/DIF is about 3 Mhz, and the
design of the transmitters and recievers is abysmal. Maybe
if everything else was done right, then cables, etc. wouldn't
matter. So much was done wrong or cost cut till it screwed
up that they do come into the picture.
A good web source for info on jitter is located at:
http://www.digido.com/jitteressay.html
AND:
http://www.audioprecision.com/publicati ... tm#Digital Audio Transmission
I recommend a couple of options for digital interconnect cables in my DIY Interconnect Note.
-
yanyufon - SR40
- 文章: 0
- 註冊時間: 週二 2月 11, 2003 7:51 am
- 來自: USA
由 JY 發表於 週四 6月 03, 2004 4:15 am
The PCM2702 is a single chip digital-to-analog converter offering two D/A output channels and an integrated USB 1.0 compliant interface controller. The newly developed [b]SpAct
『這裡、那裡、以及彼端。我們所身處的這裡，既不是天堂也非地獄。只是暫時的停留，是一個學校，或是一個轉戾點。我們在這裡學習，等待進步。我們不為什麼而活，但是我們不怕死亡。』 － ヤюЪ
-
JY - SR325
- 文章: 1550
- 註冊時間: 週四 11月 15, 2001 12:40 pm
- 來自: 台中市,ON
23 篇文章
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