PhotoBlog - 影音
本文作者:Kent發佈日期:2011-04-24 21:06:50
在Facebook 專頁按「讚好」,免費影音資訊自動送上
電腦數碼音樂重播專欄 (1)
數碼音樂重播專欄 - 序
15 年沒有執筆寫音響專欄,音響產品的改變不大,音響市場及用家心態卻變了不少。 由 1982 CD 鐳射唱片 (Compact Disc) 推出後,數碼化的音樂和影像已經在市塲 30 年,和我們生活結合。 現在提起電話,打開電視,每天上 facebook, 微博,email 都是數碼化的世界。 由今期開始,筆者希望跟各位分享數碼音樂世界的發展,其中可能是最新技術,也可以是基本認識,甚至是高解像錄音下載,軟硬兼備。 希望這專欄可以多點互動,所以歡迎大家把問題直接電郵至 chinamastering@gmail.com
CAS - Computer As Source
先講音響潮語,這 CAS 不是 ”民安隊”,是由 ”電腦作為訊源” 之解。
電腦及數碼科技的進步,今天愈來愈多樂迷及音響發燒友已採用電腦作為他們的音響播放方式。 當中涉及軟體應用和硬體選擇,五花百門。 用電腦播音樂有幾多好處,有幾多問題出現,會是今後探討分析的重點。
傳統的 CD 播放形式
CD 機播放 CD 可能並不是新鮮事。 由最早期 14bit CD 機至現在大多數 CD 機都配備 24bit / 96kHz 甚至 192kHz 的內部解碼功能。 重點是只要是播放 CD 的,都只擁有 16bit / 44.1kHz 的質數。 音響市場我們常見寫在唱片上的 24 / 96 錄音、192kHz 錄音,其實在製作 CD 過程中,都會被迫打回 CD 制式 - 16bit / 44.1kHz。 這點是現在至未來也改變不了的事實。
高解像錄音 -> CD
筆者第一個 24/96 錄音大約在 1995 年中生產。 由高解像錄音轉回 CD 這過程中,當中有涉及高深數碼學問,但是跟高清音源還有很大差距。 最佳例子便好像高清影像在 Bluray 跟 DVD 的分別。 15 年來我都希望把錄音中最好的質數帶給音響愛好者。 筆者獨立製作的 ”爵士原音” 三部曲便是由 CD -> SACD -> HiRes 所進發。
甚麼是 Bit Depth / 甚麼是 Sampling Rate?
數碼電腦世界並沒有分開音樂、影像、文字。 在電腦語言來說, 所有內容都是由 ”0” 和 ”1” 組成的二元位文字 ( binary number)。 在我們十進制的 ”10”,在二元位文字便是 ”1010”。 未來在 CAS 內容,我們將多次談及二元位文字,所以希望大家多關注。
Binary 二元位 | 0 | 1 | 10 | 11 | 100 | 101 | 110 | 111 | 1000 | 1001 | 1010 |
Decimal 十進制 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 |
數碼 CD 音樂儲存 16bit 代表了有 16 個元位的字元長度 (Word length)。 最大可代表的數字便是 “1111 1111 1111 1111 1111 1111 1111 1111” = 十進制的 ”65536”。 一個真正 24 元位錄音最大可儲存十進制的 “16777216” 數字,簡單來說是較 16bit CD 格式高 256 倍的細節儲存度。 用聲學語言來講,每一 Bit 數碼字元大約可儲存 6dB 動態,CD 格式可有 96dB 動態,而 24bit 格式便擁有 144dB 動態。
Sampling Rate (取樣率)?
由著名 Harry Nyquist 和 Claude Shannon 的基本通訊學理論,把一個訊號以數字化表示。 Nyquist-Shannon 理論說明要完美把一個訊號以取樣,取樣率必須是被取樣訊息中最高頻率的兩倍。
CD 格式取樣率是 44.1kHz (44,100Hz) - CD 格式中最高可儲存的頻率是 22,050Hz。
96kHz 取樣率是 96kHz (96,000Hz) - 96kHz 格式中最高可儲存的頻率是 48,000Hz。
192kHz 取樣率是 192kHz (192,000Hz) - 192kHz 格式中最高可儲存的頻率是 96,000Hz。
人耳的聴覺範圍由 20Hz 至 20,000Hz。 由此可見正常來說 44.1kHz 已經大致足夠。 問題在那裡,未來再談。
|
今期基礎技術重溫
CD 重播中,每一秒鐘的音樂會分成 88,200 個不同的 Samples (兩聲道 x 44,100),而每一個 Sample 也儲存了 16bit 字元長度資料。 Sampling Rate 取樣率越高,每秒鐘音樂會分成 Sample 也越多,可收錄訊號最高頻率也越高。 Bit Depth 字元越長,每個 Sample 可收錄的訊號準確度越高。
筆者自我介紹
Kent Poon 潘建章於 1993 在北美開始學習錄音技術。 1997 加入世界最大音響學會 AES (Audio Engineering Society),是其中最年輕會員之一。 5 年間參與超過 600 多個錄音製作。 Kent 於早期音響雜誌撰寫數碼技術文章而為本地音響愛好者所認識。 1999 年跟加拿大 Focus Audio 合作研製 MT1 / MC1 揚聲器,供應 McGill 大學錄音部門及格蘭美得獎錄音師所採用。 由 2003 回港擔任威達公司技術顧問至 2008 年。 2000 年成為 Weiss 顧問,2010 年在香港成立 Weiss 亞洲區辦事處。 Kent 擁有一私人母帶製作所。 參與的錄音製作包括 2008 年奧運會、趙學而以及爵士原音系列。 執筆之時正聽着 Quincy Jones <From Q with Love> 精選,剛有幸發現了法國 Devalet dpremier 是當世數碼重播之頂峰。 kent@designwsound.com
Last modified: 2011-08-08 02:46:24
發表您對 < 電腦數碼音樂重播專欄 (1) - 序 > 的意見
最新資訊 - 市場
駿韻音響有限公司Wise Sound Supplies Ltd.
2024-11-06
最新資訊 - 市場
新漢建業有限公司市場資訊
2024-11-06
最新資訊 - 影音
Furutech 宣布旗艦級電源排插「NCF Power Vault 」正式在日本國內接受預訂
2024-11-05
最新資訊 - 影音
支援 LC3 及 aptX Lossless,Creative 推出全新藍牙傳輸器「Creative BT-W6」
2024-11-04
最新資訊 - 影音
聽歌、睇戲及打機皆宜,Creative 推出具 HDMI 輸入 USB-DAC「Sound Blaster G8」
2024-11-04
最新資訊 - 影音
TEAC 推出全平衡 / 雙單聲道設計的耳機放大器兼前級放大器「HA-507」
2024-10-31
最新資訊 - 影音
Accuphase 推出全新 A 類合併式放大器 E-800S
2024-10-29
最新資訊 - 影音
全新 KOJO Technology Crystal Ep-G 系列登場
2024-10-29
107 Comments
15 年來我都希望把錄音中最好的質數帶給音響愛好者。<<<===
最喜歡這句...support
Support! Kent is nice and helpful.
Support and looking for more in-depth comments on PC vs Mac, Firewire vs USB, FLAC vs Apple Lossless...
>>>...15 年來我都希望把錄音中最好的質數帶給音響愛好者....
"...最好的質數..." 肯定無打錯字!當中的深層意義可解作為 "....數碼音樂的最好質素...."
^______~
support
Hi Kent,
1. I am surprised you find time to start a thread here and more so in this site instead of DWS and HiFiTrack.
2. "剛有幸發現了法國 Devalet dpremier 是當世數碼重播之頂峰" Would you please elaborate on this so that we may share your "discovery" and the virtues of the premier?
3. I am aware some audiophiles still prefer the sound of analogue LP whilst I have switched over to digital myself. I had the fortunate opportunity the other day to listen to my high resolution tracks via Avant Garde Trio driven by Kondo pre and power amps. I will be listening again next week with over a dozen direct discs played in the Kondo turntable and cartridge. I hope to be able to find out and confirm my own views on analogue LPs versus digital.
Thanks guys.
I always wish to collaboration with more parties. (I am doing so too).
Indeed it is quite difficult for me to write chinese articles these days, but I wish you enjoy them. These articles are published on AV Bi-weekly magazine. (This week is the first time I missed....sorry)
You will find clone of these articles on DWS and other areas too. As time goes by I wish to share with you on my audio journey, and there will be free audio samples, takes or other audio materials for download. This is what I count on web base media.
From my view, Devialet d-Premier is one of the most unique audio products for a VERY long time in hi-end market. I simply cannot find anything better in term of its feature and sound. (of course its sexy look too.) A lot of people may view it as very expansive product. I have opposite view that it's a bargain in hi-end sector. Its amplifier driving ability can compete with the best amplifiers out there, and you cannot separate its DAC and Amp ability. Devialet is not a DAC + AMP, it is merging 2 as one. I am a happy user.
I absolutely have no negative feeling of LP or analogue. Before anything is converted to digital, they are ALL analogue signal. Microphone, mic-preamp, line preamp, are all analogue circuits. LP is one analogue media, open reel is another. I have honor to listen Nat King Cole open reel safety master and Living Stereo masters, I can still remember strongly how do they sound. Capturing and transfer these materials to digital domain do not necessarily means inferior quality as well. We have very good quality A to D and D to A converters these days.
Modern production is not suffered by technologies but people who abuse it. I wish to share more in the future. Speaking of it, I am now writing this with following view. I have already worked over 30 hours, will have to get more things done before tomorrow 8am........guess I am enjoying it.
Next 2 days can share with you all Korea HiFi show. Thanks review33.com for such collaboration. Tomorrow night I will listen to Medea+ with Magico Q5, nice.
.
.
.
.
.
Who is she?
Bondy
Lossless CAS platform is definitely the furture of music format but there are a very big risk.....that's how to prevent the illegal copy of such music files. The industry should have a scheme to prevent a second copy such as the SCMS (The Serial Copy Management System) as DAT adopted before.
The topic of illegal copying has been considered for some time by record labels in the music industry. More and more of them are offering high resolution tracks online. Why are they not as worried now as before about illegal copying? One answer is that experience has shown such infringement has not proliferated. Another answer is that copying and sending high resolution tracks around is physically prohibitive. Imagine an album of DXD produced by 2L of Norway has a file of 8 GB, about 1 GB per track, 8 tracks per album. A youngster with a PMP or iPod will decline taking it even if his friend offers it to him because his storage capacity will be overloaded. Alright a 24/96 track only has one quarter the size but is still substantial for his machines. And no one will send it around in mails even if there is capacity for the attachment because the uploading and downloading takes unduly long time. Yes, MP3 tracks are still being copied and sent around but because the price is much cheaper than CD, some prefer to buy selected tracks rather than wait for them to come somewhere.
Incorporating Digital Right Management as suggested is likely to kill high resolution sales because invariably it affects the sound. This will be a stupid move by the industry and there was already prior lesson.
CAS = "Computer As Source" ?! Where did you learn this? @@"
I only know CAS as "Computer Audio Source"
OMG. So surprising an "audio professional" (although only self-proclaimed) making this mistake.
Says who?
Like many acronyms, the term CAS is informally invented by people for pure convenience for the benefits of quick communication. AS far as I know, this is mostly a Hong Kong/Asia term. Countries with English as second or third language tend to make up short forms for convenience. It is not as commonly used by the American/UK audio press for instance.
For all I know, CAS could be "Computer Assisted Surgery" or thousands other representations. So to laugh as someone as being "wrong" about an informally spread term is silly. Just because someone else mentioned it before does not make it the official term.
As far as I am concerned, "Computer Audio Source" does not sound right in English anyway. To me, "Computer based Audio" sounds better; "Computer as Source" also "gels" better grammatically.
The term CAS.
I first learned about it in Kent Poon's web, HiFitrack and Design With Sound. It stands for computer as source.
Then I came across the term elsewhere that is supposed to mean Computer Audio System.
Then we have the web Computer Audiophile.
It is not in Wikipedia.
The "C" stands for computer, that is, using it or a varied form of it, for example, a storage system or a server to play digital tracks.
CAS for computer audio playback not really a standardized or academic abbreviation, no absolute right or wrong term, actually calling such playback based on studio master file from download, can be play back by mobile phone, memory player, universal transport, network streamer and of course computer, as "file based audio playback".
probably a few years later, broadband speed and cost much lower, cloud computing and storage (what amazon doing now), we don't have to purchase physical media or file anymore, we may purchase audio/video subscription fee based on usage or quality (24/96 or above premium quality, 16/44 CD quality and mp3 lossy), you can access to all media stored on server, streaming to device by wifi/3g, so the content not require to store, or not allow to be store and re-transfer (well, if the subscription allow to access all media, why taking extra effort?), just access to playlist and ready to go!!
Last.fm is doing so, just not reaching audiophile quality yet, until records company willing to release Hi-Rez source.
"""AS far as I know, this is mostly a Hong Kong/Asia term. Countries with English as second or third language tend to make up short forms for convenience. It is not as commonly used by the American/UK audio press for instance."""
totally agree. at first when i encountered this term i had no idea what it was.
traditionally computer is the cpu with other items put in one box. these other items like sound card, memory hard disks or storage media are peripherals. one can stores data from a remote storage device and that can be the source. so computer as source is misleading.
i can see the trend will go with pay as you play on demand. it is more like renting instead of owning the software. who still want to have piles of cds and keep your library showing locations/numbering.
there are many software which may help split big wav or flac files using splitter or cue. split files can be transfer to dvd discs as data and then rejoin together at later stage.
Korea HiFi show system.
System 2
Dynaudio with Linn
JBL with Mark Levinson
.
Wilson with Meridian
i hate to ask this question " can computer audio source (cas) really win cd transport for dac?"
I don't invent the term. I didn't know what it means at the first time, nor understand why someone would like to make scene of it.
CAS to me has been "民安隊". I also heard Eric (Editor of AV Magazine) mentioned it. Our government has a web page: www.cas.gov.hk.
People I met (consumer audio world) speaks it as "CAS System". At the first time I thought they mean "CARS System". Singapore, Malaysia audiophiles all speaks this way. They do not pronounce as independent C. A. S. system but "CAS".
I agree cloud base music library will become the majority of mass culture too.
Computer Audio System
Computer As Source
Computer Audiophile System
Computer Audio Server
as you like.
" can computer audio source (cas) really win cd transport for dac?"
u are asking if 16/44 cds can win over 24/88 96 176 192 and above.
theoretically the higher the bit rate and length the closer towards continuous analog as there are less gaps for the algorithms to do the guess work.
similarly in video u are comparing 576i dvd quality with 1080p blu ray video quality. if u consider 1080p better than the 480 or 576 output then the same principle applies to audio too.
hey, I mean cas comparing with cd transport playing 16/44.1 to dac
one more thing, our human ears cannot hear more than 20khz sound , why 24.96 (around 48khz) sound better than 16/44.1?
one more thing, our human ears cannot hear more than 20khz sound , why 24.96 (around 48khz) sound better than 16/44.1?
----------------------------------------------------
If you treat PCM is sampling (snapshot) of analog waveform, at audible frequency, let said 4Khz, 96Khz have double number of sample compared with the one at 44khz sampling rate, more true to original analog waveform.
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.
there are many theories about hearing beyond 20khz
here is a good read
http://www.earthworksaudio.com/wp-content/uploads/th_world_beyond_20khz.pdf
Hi Kent,
Could you share with us the DAC part of Devalet dpremier compare with Weiss DAC2.
"""I mean cas comparing with cd transport playing 16/44.1 to dac"""
assume u have a cd with no scratch and u rip the data to a hard disk with perfect bit which should be the same as u download the same music data file from internet, this should be the same as when you play a perfect cd thru a cd transport.
the difference lies with scratched or imperfectly made cds.
when u rip a scratched cd to hard disk it all depends on your chosen software how to handle the errors and make corrections by guessing. this error correction can be made many times like 20 to 50, for example.
with cd transport it is a different matter. cd transport has limited time frame to make error corrections in may be say 3 to 5 times read and then the inbuilt software will make rougher guess work and transmit the info through to the digital analog converter. a good cd transport may handle error corrections better than crappy ones but no match to more reliably ripped data.
does this make sense?
Batmanames04
Let me say something to complement answers to your question that I understand to be “whether CAS with a higher resolution track could sound better than a CD track from the same source, both played with the same DAC and audio system”.
WOULD KENT POON PLEASE CORRECT ME IF I SAY ANTHING INCORRECT RELATIVE TO YOUR PROFESSION.
Nowadays first hand digital recorders come with ADCs with higher resolution than the CD format. This was what a proprietor of a relatively new record label told me in our correspondence when I asked him why his “master” and DVD-ROM is at 32/192. You are probably aware Professor Johnson of Reference Recording uses 24/176.4, Morten Lindberg of 2L uses 24/352.8 etc.
Because consumers are stuck with the 16/44.1 format in the ubiquitous CD players, the music industry have to “master” down their native or original masters to suit. You may also be aware there are a series of masters coming out from the native masters and mastering, amongst other things, include, “improving” bad recordings to acceptable level; mixing down multi tracks into 2 for CD; and relevant to our topic here, down converting the higher resolution tracks to 16/44.1. Music industry personnel have us believe the higher resolution tracks contain no more musical data than in their CD offspring and recording at higher resolution merely facilitates subsequent mastering that may be easier carried out with flatter slope filters. The majority of consumers are convinced!
Now let us look at what that particular mastering process is doing to the high resolution tracks. A 24/352.8 DXD master has a bit rate of 16, 934 kbps and a file size of roughly 120 MB per 1 minute of music. A CD has a bit rate of 1,411 kbps and a file size of roughly 10 MB. In other words, the CD master only contains 1/12 of the data in the native or original master, meaning 11/12 of data have been filtered out! A CD in comparison with its 24/96 master, has 5/6 of the data filtered out.
Playing high resolution tracks with the computer means playing with the same bit rate and sampling rate as used in the native or original master.
Now you make your own judgement as to whether the 11/12 filtered out data are absolutely useless for our appreciation of the music in the track or perhaps there are overtones, upper harmonics and reverberation that enrich the fundamentals and harmonics within that hearing limit of 20 KHz.
One point to bear in mind is that when “mastered down” from higher resolution, the CD master loses the filtered out data. Therefore even when a CD track is up converted subsequently, the missing data previously filtered out are merely interpolations. The up converted track only resembles the native or original master. How close the resemblance depends on the algorithm used in the upward conversion.
Factually, MP3 involves similar mastering down. You may ask a similar question: whether a CD track could sound better than a MP3 track from the same source, both played with the same DAC and audio system. Say you have a MP3 track of 320 kbps that has about ¼ of the bit rate and file size of the CD. I wish the MP3 sounds as good as the master because I would be able to buy the albums of 2L in the iTunes store at a fraction of the price.
Another important issue relevant to the topic under discussion: We are often told our hearing is between 20 Hz to 20K Hz at best and based on Nyquist 20.05KHz (half of 44.1) is already beyond our hearing, thus CDs are more than good enough for practically all of us.
Don’t trust people like me in forums who may merely be talking crap. Listen to what your admired engineer, the mastermind for the Invicta DAC has to say in the Owner’s Manual that you referred me to. There are too many pages there to be reproduced here and I merely paraphrase the key issue. The human brain, as supported by tests that he quoted, is able to sense and fill in sound above 20 KHz, up to 50 KHz. Listeners are able to sense fullness if sounds above 20 KHz are captured in the recording and retained in the playback. And there are listeners able to detect differences when higher resolution tracks are played in comparison with CD tracks from the same masters.
Well observational listening to pick out the difference is another topic. If you want to talk about it, please tell us your previous encounters with listening to CD tracks and CAS high resolution tracks and why you have doubt about any difference between the two. We then start from there.
I must make clear I do not sell anything or help anybody to sell anything, thus no ulterior motive to convince people to use the computer and buy tracks or DAC.
thanks momei, i need sometime to digest it
there are master recordings made with 24/192 and convert to cd format 16/44. u will notice that half of 192 is 96 and half of that is 48. so from 48 to 44 there will be another downscale.
better fit is from 176 to 88 and then to 44.
also note that there are some so called hi rez downloadable files in the format of 24/88 or 24/96 actually upscaled from 16/44. they are cheats and not necessarily mixed from master recordings.
from 20khz to 40khz there is only one octave. similarly to 200w to 400w is just 3db difference. not much at all.
"A CD has a bit rate of 1,411 kbps and a file size of roughly 10 MB. In other words, the CD master only contains 1/12 of the data in the native or original master, meaning 11/12 of data have been filtered out! A CD in comparison with its 24/96 master, has 5/6 of the data filtered out. "
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.
Yes Hercules I take your point.
The crux of the matter is whether the data filtered out represent utter silence that is totally useless (over 7 GB in the case of a 2L album) or therein are overtones and harmonics that enrich the fundamentals and harmonics within the 20 to 20 KHz hearing limit.
Well I belong to the camp accepting the enrichment concept. New equipment, for example, both the Wadia S-7i CD player and McIntosh preamps incorporate up conversion processes on the incoming CD format of 16/44.1. Therefore audiophiles have to decide for themselves whether these are merely marketing gimmicks or they actually will bring about improvements in sound.
in video u will view all black or all white using 576i format data the same as 1080p format data, irrespective there is upconvert of 576i to 1080p. however when there are motions and multi colour images it is a different story. most people can see true 1080p better than upscaled 578i to 1080p.
why is it so hard to hear similarly in audio with upsampling of 16/44 format to hi rez to approximate true hi rez?
quoted from BobKatz (great recording engineer) Mastering audio the art and the sciencethe ultimate listening test:
is it the filtering or the bandwidth?
in December 1996, I sought to systematically find reasons for sonic differences between sample rates, performing a listening test. with the
collaboration of members of the Pro Audio maillist. The question we wanted to answer was. Does high sample rate audio sound better ( or different )
because of increased bandwidth, or because of less-instructive filtering? we developed a test that would eliminate all variables except bandwidth. Other major
factors were held constant, sample rate, filter design, DAC. and Jitter.
The test were devised was to take a 96khz recording, and compare the effect on it of 2 different low pass filters. The volunteer design team consisted of
Ernst parth (filter code), matthew xavier moral (shell), rusty scott (filter design), and bob katz (coordinator and beta tester). We created a digital audio
filtering program with two impeccably-designed filters which are mathematically identical, except that one cuts off at 20khz.
the filters were designed for overkill, with exemplary characteristics : double precison diethered, FIR linear phase, 255-tap, > 110dB stopband attenuation, and <
.01dB passband ripple.
Fort the first listening test, I took a 96khz orchestral recording, filtered it and laid both versions into sonic solutions DAW for comparison. I expected to hear
radical differences between the 20khz and 40khz filtered material. BUT I COULD NOT HEAR ANY DIFFERENT! Next, I compared the 20khz filtered against "no filter" (of course , the
material has already pass through two 48khz filters in the converters) Again I could not hear any difference! The intention
was to listen double blind ; but EVEN sighted, 10 additional listeners who took part in the tests (one at a time) heard no difference between the 20khz digital filter
and no filter. And if no one can hear a difference sighted, why proceed to a blind test?
I then tried different types of musical material, including a close-milked recording of castanets ( which have considerable ultrasonic information). but there was still no audible
difference. I then created a test which put 20khz filtered material into one channel of my Stax electrostatic headphones and the time- aligned wide bandwidth material
into the other channel. I was not able to detect any image shift-there was always a perfect mono center at all frequencies in the headphones! This must be a pretty darn good filter!
As a last resort, I went back to the list and asked maillist participant robert bristow johnston to design a special Dirty filters
with 0.5dB ripple in the passband. Finally, with this filter, I was able to hear a difference... it added a boxy, veiled, "
gritty" quality that resembles the sound of some of the cheaper cd players we all know.
After I conducted my test, several others have tried this filtering program, and most have reached the same conclusion: the filter is inaudible. one maillist participant, eelco grimm, a netherlands based writer
and engineer, performed the test and reported no audible differences using a sonic solutions system, yet he and a colleague passed a blind test between filtered and non filtered using an augan workstation. He did
not compare the sound of the 20khz versus 40khz filters, so we are not sure if he was hearing the filter or the bandwidth ( i suspect the filter). We are not certain, but perhaps the reason eelco uniquely reported a sonic difference is that the sonic system produced sufficient jitter to mask the other differences, which must be very subtle indeed!
! Be aware that two other 48khz filters in the chain may have obscured the audible effect of the test filter, so it is very difficult to design a perfect test.
this 1996 test seems to show that a " perfect 20khz filter" can be designed. regardless of whether eelco s group did reliably hear bandwidth differences, it should be clear by now that differences people hear between sample rates more likely due to filter design than to supersonic bandwidth. Ironically, it was necessary to make a high sample rate recording in order to prove that high sample rates may not be necessary
My DIY DAC hear almost no different between 16 44.1 and 24/96 material if they are derived from the same source
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)
1996 thats 15 years ago. there was no diamond beryllium or other tweeters with exotic materials. could those old tweeters output properly the frequencies above 20khz? how good was the room treatments for the tests? if it was one that absorbed a lot of high frequencies of course there was less chance to hear properly.
"""a good dac should hear no big different bewteen 16 44.1 and 24 96 material"""
i must be wrong. how can a dac hear things? i thought human can hear but a dac?
from wiki:
Theorems have two components, called the hypotheses and the conclusions. The proof of a mathematical theorem is a logical argument demonstrating that the conclusions are a necessary consequence of the hypotheses, in the sense that if the hypotheses are true then the conclusions must also be true, without any further assumptions.
The concept of a theorem is therefore fundamentally deductive, in contrast to the notion of a scientific theory, which is empirical.
Batmanames04,
Merely to mention 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.
Anyway, back to the subject matter. If you use a PC and Foobar 2000 or WASAPI, then I get stuck because I have never used them. If you use MAC and iTunes then follow test 1 below.
Test 1: You need to open MIDI, play a track of 24/96 and check whether you have set your computer to recognize you DIY DAC and also whether the setting displays 24/96 and not 16/44.1 whilst you play the 24/96 track. If the recognition and setting remains at 16/44.1, then it means you have always played your 24/96 tracks at 16/44.1, thus you heard no difference. If you have already done this previously and the recognition and setting is correct, please come back for test 2.
try to audition a high end DAC, your ears can hardly distinguish 16 44.1 and 24 96 recordings
momei
I am using PC with foobar kernel streaming mode and the screen showing sampling rates16 44.1 and 24 96.
If the file is playback from computer then the sound quality maybe affected by the quality of the computer.
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.
then it will be comparing cdp vs pc, not a good control experiment
may be you can use pioneer dv ax 10 universal player output 16/44.1 and 24/96 for comparison
CPU8088,
Regarding beryllium tweeters: not intending to challenge you but merely to tell you the fact. In the 70's Yamaha already made beryllium tweeter for its NS1000 loudspeakers. They were available in Hong Kong and some audiophiles bought them.
The Yamaha tweeters are distinguishable from that of current Focal ones in that the Y ones are convex whereas the F ones are concave. I have read posts here about your undue favour towards Focal speakers. Could you tell us what process Focal uses to make the tweeters? Vapour deposition or what? Apparently the Focal tweeters are thinner than the Yamaha ones. I do not have information what process Yamaha used and I guess it might be heat formation, that is, putting the metal powder in a mould and heat it to the melting temperature in an oven. Talking about Focal Utopia, I suspect, based on what I deduced from listening, the tweeter has dispersion restrictions merely because of the design of the tweeter cabinet that has the width and depth the same as the woofer cabinet. Other designs such as the Karma Exquisite, the KEF Fat Lady all have the tweeter housing tapered to the smallest size to help dispersion. Or in the B&W and Tannoy Royal designs, place it on top.
this thread is about digital music not tweeters
anyway yamaha of the old was using beryllium vapour deposit not pure beryllium what focal is doing.
the diamonds of b and w and kharma are vapour deposits with diamond flakes bond together.
dispersion pattern, baffle and cabinet designs are different subjects. what i questioned was what speakers used in 1996 for the test to output over 20khz?
but most 33 members know you like jm lab beryillium tweeter
a CD contains 16 bit samples with a 44.1 kHz sample rate. this allows theoretically for a dynamic range of 16*6= -96 dBFS and 44.1/2 = 22.05 kHz as the highest possible frequency.
recordings are often made with a greater bit depth (24) and a higher sample rate.
one of the benefits of computer based audio is that you are not in need of recordings down sampled to CD format. uou can play the original recording at its original bit depth and sample rate (if your sound card allows for it).
a 24/96 recording allows for a dynamic range of 24*6= -144 dBFS. this sounds impressive but CD’s -96 dBFS is very soft and the noise floor of your gear e.g. -110 dBFS will be the limiting factor.
probably listening to the decay of instruments might reveal a subtle difference.
96 kHz has a Nyquist frequency of 48 kHz.
Sounds impressive too but our hearing may stop somewhere at 20 kHz (when we are young).
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.
often no audible differences between CD audio and higher resolutions are reported on the internet. similarly some claim no audible differences between CD audio and high bitrate MP3. as 1+1=2 some conclude that there isn’t a difference between MP3 and Hi-rez audio!
2)our tweeters can sound different when modulated with signals 20 kHz.
????
similarly some claim no audible differences between CD audio and high bitrate MP3
??? (can mp3 reach 20khz bandwidth
Batmanames04,
It seems better for me to know a bit more about your DIY DAC and the tracks you used by which you found no difference than to ask you to go straight into Test 2.
Does your DAC support 24/192 or even DXD? Does your computer out port support 24/192 and what interface or connector you use to play those tracks? The port and interface could be bottle necks and is 24/192 shown on the monitor screen to confirm such resolution is being played? I am trying to see whether by listening to 24/192 you may be able to hear the difference. Also please list two to three tracks you used previously, the source and the genre (type of music).
I forgot where I read about the story. Morten Lindberg (proprietor of the 2L label in Norway) took some of his tracks to the US to meet up with equipment manufacturers and importers and played them two different resolutions. At the end they all preferred the higher resolution version and took copies of those.
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
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
what dac are you using right now?
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.
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
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.
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.
"""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,
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.
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.
my brother like the recording much
but i am the same as you just for testing
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.
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?
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
good info, thanks for sharing
studio use 24/96 analog to digital converter for pcm recording?
> 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.
Many thanks Kent for the detailed answers.
> 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.
a photo for you guys to relax
Hi Kent,
As you are a MacBook Pro user and also selling Weiss DAC at Fortress, would you try out the new Thunderbolt and tell us whether it has any advantage in terms of sound reproduction over Firewire 800 that you encourage audiophiles to use. Fortress sells the computer and you may plug it with Thunderbolt into your demo gears there the sound of which are familiar with, and tell us whether there is any improvement over Firewire 800.
Thanks
Is Thunderbolt backward-compatible with USB and FireWire?
Third-party vendors will sell adapters, available sometime this spring, that let you connect USB, FireWire 400, and FireWire 800 devices to Thunderbolt ports. Thunderbolt won’t make these legacy devices any faster, however—they’ll still be limited to the performance of their built-in components. For example, a FireWire 800 device still won’t be able to transfer data faster than 800 Mbps.
-------------------------------------------------
I expect the thunderbolt port running in firewire 800 compatible mode to work with Weiss DAC will make no difference as ordinary firewire port.
The real difference and advantage can't tell until real thunderbolt device available.
Sorry that I missed this post.
CAS blog #4 is about the INTEL Cebit show in German. Weiss in invited to demonstrate in Intel booth with MAN202 prototype (which uses INTEL technology).
We also touch base with Intel if Thunderbolt will provide better quality, but we doubt it. MAN202 is also using Apple iPAD as remote control.
Since Thunderbolt is compatible with USB/Firewire/Display Port, just get the matching cable will be the same.
Faster bandwidth does not necessary means better quality, but we will see.
and Batman hing,
thanks for posting STS company. They are our good customers. Actually this workflow can be enhance when the ADC2 is working in 96kHz and maintain at 96kHz until finally Pow-R with SRC back to 44.1kHz.
The main reason is both DNA-1 and DS-1 are working in double sampling rate (88.2/96). This means each of them will upsamples the 44.1/48 signal before processing and downsamples back to 44.1/48 at its output.
Dear Kent,
Can you let us know which specifice Intel technology with Man202 use? I thought it will be a dedicated computer running Linux?
Dear ackcheng,
MAN202 uses Intel SSD, Intel chipset and other technologies. Indeed it is what you describes. Next week we will have an important meeting in Munich to solid the idea of MAN202 core technology as platform.
A platform means it can be an on going development path on audio DSP, which Daniel's team has been best at. We also have to set goals in order to have products ready for the market in different time schedule.
We also will discuss the AFI1 development. AD/DA etc. Samuel Groner will lead the analogue circuit designs (He designed DAC202, OP1-BP, and coming OP1-JFET soon.
Dear ackcheng, (and other Weiss friends).
I think you will have more ideas about everything from this interview: http://designwsound.com/dwsblog/2011/05/interview-daniel-weiss/
Thanks! That's ver informative!
So many probing questions from the knowledgeable interviewer; a "must read" for any audiophile interested in computer audio IMHO.
With the technology available with Weiss, he should try to use SHARC as the DSP processor instead of intel chip. This will make subsequent processing software alot more powerful. Also make the competitor difficult to catch up. I am afraid that he will sacrifice the quality for profits.... As mentioned in the interview, even if the Dac is of poorer quality, it will sell because of the brand name.
Just some random thoughts only. Hope you don't mind!
Dear ackcheng, you mis-understood his point on the "poor quality" DAC.
What he means is nows today people are rating the DAC chipset more than anything else in DA designs. So that even Weiss products do not use the best "specification" DAC chipsets, it will still sell because it will reach the quality standard in our own perspectives, and more often better than other companies using the best "specification" chipset.
We all know how much it costs for a DAC chipset, but solid engineering always more important.
Further about the Sharc DSP. We have those in ALL professional DSP boxes. The best example is SFC2 (sampling rate conversion). It uses Sharc DSP calculates in 40bit fix and floating points.
Few years ago we launched our first software based software "Saracon". Saracon does not only much cheaper (1/5 prices) than SFC2 because lacks of all hardware costs. Saracon calculation is double precision 64bit floating points. It gets rid of AES/EBU standard of max. 24bit (-144dBFS). Therefore Saracon is objectively a better product (PCM to PCM in any rate reaches -180dBFS THD)
Daniel doesn't need to try using Sharc, he is consultant of TI of using Sharc in audio applications. What actually we are looking is similar to Nvidia GPU calculation in graphic card. One of this card can provide floating point calculation better than hundreds of SHARC DSPs, very scary spec. The world fastest company by China is making use of GPU calculation instead of CPU. But this is overkill for audio application.
Intel CPU calculates in 80bit floating points. Since it does not run Windows/Mac OS, the customize Linus has most direct access to resources and execution.
I do concern your comment (but always respectfully) because Daniel is not a business man and does not focus on profits at all. He will be last man to sacrifice quality.
> Just some random thoughts only. Hope you don't mind!
Oh no, not at all. Thanks for given me a chance to explain and provide more information too.
Greeting from Nagoya, Japan.
Thanks Kent for your clarification,
"Daniel doesn't need to try using Sharc, he is consultant of TI of using Sharc in audio applications"
Actually, I noticed that he uses SHARC in his Powerhouse so I thought using this technology in MAN202 would be a natural progression! Well, may be there are something better around the corner!
Using GPU may even be better than CPU for the kind of calculations we need. Something like CUBA can hack the GPU for this purpose! Someone is already working on it at DIY level!
Hello ackcheng, thanks for your comment.
Powerhouse is Weiss latest product using Sharc, AND it is a product which cannot see its life in public. =....(
Luckily, Weiss has customized clients, especially in audio research field. I am sure you notice the Virtual Haydn project by McGill's Wieslaw Woszczyk.
http://www.music.mcgill.ca/thevirtualhaydn/
That was time Daniel and I discuss "WHEN" computer surpasses hardware DSP.
There are 2 types of DIYers - one thinks he/she smarter than all commercial brands. The other is doing something really special but have no interest in commercial direction.
It is a pity that the powerhouse project is killed. But it is good to know that you guys have a clear direction! Well, I'm a professional, so now as a customer, I will leave the tough part to professional like you! DIY is just for fun, for me!
powerful sharc processor , same as the one used in weiss medea ( please correct if I am one)
old weiss medea ( ) with sharc processor ( exactly same version sharc processor as accuphase DF 35 digital crossover)
DS1-MK3 has 5 x Sharc DSPs.
BW102
BW102
BW102: Modular system
hardware controller
This modular controller for the DSP moduler racks. Later on there is computer and provides software controls with a Mouse and Keyboard.
Good old times.....20 years ago. The DSP still supports up to 24bit/96kHz and you can still see it in almost all top mastering studios.
Hi Kent,
Have you updated the firmware of your Devialet D-Premier to the newest 5.3 version?
It's said that there will be very obvious sonic improvement:
http://www.6moons.com/audioreviews/devialet2/1.html
no, but it is a very very good product.
what monitor speakers you using right now...Yamaha? Using pro tools?
Since university I have to learnt the curve of the NS10, I never have Yamaha speakers.
My current one is ATC-SCM100ASL. Thought of a while to update.....
We have protools but not my usual DAW setting.