[討論]硬體和軟體 升頻的聲音

[zzz201]

請問大家對於硬體或軟體升頻 .....印象? 聽感?

以前使用過的器材(DAC)...軟體....
是對於升頻的聲音....覺得多此一舉
聽起來也是改變比改善大.....

直到使用NOVAS DAC 1794 和RME 9632才完全改觀，
而前者是硬體升頻、後者是DSP軟體升頻。

升頻 Upsampling or Oversampling技術
在HIEND很多廠都有採用
而每家都有其獨特的處理....

想有專業背景的高手 或是對喜歡升頻後的聲音 甚至對升頻有疑問的各位 來討論一下

1.軟體升頻的話: 軟體的選擇 優點? 缺點? 甚至是聽感?
2.硬體升頻 優? 劣 ? 聽感?
3.升頻的選擇 兩倍頻(88.2 or 96) 哪個?好聽? 四倍頻 (176.4 or 192) 哪個?好聽?


目前小弟是比較喜歡 透過dac將 44.1 升頻成176.4
前幾天有試過用 ssrc 1.30軟體 將檔案升頻成 176.4
但比起來我還是比較喜歡dac 切換成 176.4出來的聲音
出來的聲音 音場 轉折 甚至是細節
都聽起來比44.1和軟體升頻的喜歡.....

另外還有些疑問
像RME、Lynx 高階數位輸出卡 都有支援DOUBLE WIRE輸出 RME甚至有QUAD 輸出(四倍頻)
這種CHORD、ESOTERIC等很多可DOUBLE WIRE 數位輸入的機子有何差別嗎?
像這種的數位輸出有統一格式嗎? 可以互相通用嗎?
EX.使用RME 、Lynx 輸出 給CHORD 、ESOTERIC 之類的DAC
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Upsampling, Oversampling and Sampling Rate Conversion in General -
引用自http://www.weiss-highend.ch/minerva/documents/minerva-manual.pdf

In consumer audio circles the two terms oversampling and upsampling are in common use. Both
terms essentially mean the same, a change in the sampling frequency to higher values.
Upsampling usually means the change in sampling rate using a dedicated algorithm (e.g.
implemented on a Digital Signal Processor chip (DSP)) ahead of the final D/A conversion (the D/A
chip), while oversampling means the change in sampling rate employed in today’s modern D/A
converter chips themselves.
But let’s start at the beginning. What is the sampling frequency? For any digital storage or
transmission it is necessary to have time discrete samples of the signal which has to be processed.
I.e. the analog signal has to be sampled at discrete time intervals and later converted to digital
numbers. (Also see "Jitter Suppression and Clocking" above)). This sampling and conversion
process happens in the so called Analog to Digital Converter (A/D). The inverse in the Digital to
Analog Converter (D/A).
A physical law states that in order to represent any given analog signal in the digital domain, one
has to sample that signal with at least twice the frequency of the highest frequency contained in
the analog signal. If this law is violated so called aliasing components are generated which are
perceived as a very nasty kind of distortion. So if one defines the audio band of interest to lie
between 0 and 20 kHz, then the minimum sampling frequency for such signals must be 40kHz.
For practical reasons explained below, the sampling frequency of 44.1kHz was chosen for the CD.
A sampling frequency of 44.1kHz allows to represent signals up to 22.05kHz. The designer of the
system has to take care that any frequencies above 22.05kHz are sufficiently suppressed before
sampling at 44.1kHz. This suppression is done with the help of a low pass filter which cuts off the
frequencies above 22.05kHz. In practice such a filter has a limited steepness, i.e. if it suppresses
frequencies above 22.05kHz it also suppresses frequencies between 20kHz and 22.05kHz to some
extent. So in order to have a filter which sufficiently suppresses frequencies above 22.05kHz one
has to allow it to have a so called transition band between 20kHz and 22.05kHz where it gradually
builds up its suppression.

Note that so far we have talked about the so called anti-aliasing filter which filters the audio signal
ahead of the A/D conversion process. For the D/A conversion, which is of more interest to the
High-End Hi-Fi enthusiast, essentially the same filter is required. This is because after the D/A
conversion we have a time discrete analog signal, i.e. a signal which looks like steps, having the
rate of the sampling frequency.
Such a signal contains not only the original audio signal between 0 and 20kHz but also replicas of
the same signal symmetrical around multiples of the sampling frequency. This may sound
complicated, but the essence is that there are now signals above 22.05kHz. These signals come
from the sampling process. There are now frequencies above 22.05kHz which have to be
suppressed, so that they do not cause any intermodulation distortion in the amplifier and speakers,
do not burn tweeters or do not make the dog go mad.
Again, a low pass filter, which is called a „reconstruction filter“, is here to suppress those
frequencies. The same applies to the reconstruction filter as to the anti-aliasing filter: Pass-band
up to 20kHz, transisition-band between 20kHz and 22.05kHz, stop-band above 22.05kHz. You may
think that such a filter is rather "steep", e.g. frequencies between 0 and 20kHz go through
unaffected and frequencies above 22.05kHz are suppressed to maybe 1/100'000th of their initial
value. You are right, such a filter is very steep and as such has some nasty side effects.
For instance it does strange things to the phase near the cutoff frequency (20kHz) or it shows
ringing due to the high steepness. In the early days of digital audio these side effects have been
recognized as beeing one of the main culprits for digital audio to sound bad.
So engineers looked for ways to enhance those filters. They can’t be eliminated because we are
talking laws of physics here. But what if we run the whole thing at higher sampling rates? Like
96kHz or so? With 96kHz we can allow frequencies up to 48kHz, so the reconstruction filter can
have a transition band between 20kHz and 48kHz, a very much relaxed frequency response
indeed. So let’s run the whole at 96kHz or even higher! Well – the CD stays at 44.1kHz. So in
order to have that analog lowpass filter (the reconstruction filter) to run at a relaxed frequency
response we have to change the sampling frequency before the D/A process. Here is where the
Upsampler comes in. It takes the 44.1kHz from the CD and upsamples it to 88.2kHz or 176.4kHz
or even higher. The output of the upsampler is then fed to the D/A converters which in turn feeds
the reconstruction filter.
All modern audio D/A converter chips have such an upsampler (or oversampler) already built into
the chip. One particular chip, for instance, upsamples the signal by a factor of eight, i.e. 44.1kHz
ends up at 352.8kHz. Such a high sampling frequency relaxes the job of the reconstruction filter
very much, it can be built with a simple 3rd order filter.
So, how come that upsamplers are such a big thing in High-End Hi-Fi circles? The problem with the
upsamplers is that they are filters again, digital ones, but still filters. So in essence the problem of
the analog reconstruction filter has been transferred to the digital domain into the upsampler
filters. The big advantage when doing it in the digital domain is that it can be done with a linear
phase response, which means that there are no strange phase shifts near 20kHz and the ringing
can also be controlled to some extent. Digital filters in turn have other problems and of course
have quite a few degrees of freedom for the designer to specifiy. This means that the quality of
digital filters can vary at least as much as the quality of analog filters can. So for a High-End Hi-Fi
designer it is a question whether the oversampling filter built into the D/A chips lives up to his/her
expectations. If not, he/she can chose to design his/her own upsampler and bypass part of or the
whole oversampler in the D/A chip. This gives the High-End Hi-Fi designer yet another degree of
freedom to optimize the sonic quality of the product.
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PS.小弟如發文內容有誤 麻煩指教

[Mira]

1.軟體升頻的話: 軟體的選擇 優點? 缺點? 甚至是聽感?
2.硬體升頻 優? 劣 ? 聽感?
3.升頻的選擇 兩倍頻(88.2 or 96) 哪個?好聽? 四倍頻 (176.4 or 192) 哪個?好聽?

1. 主觀聽感上沒有優劣，純粹看自己是否喜歡。
2. 同上。
3. 升頻的比較是沒有意義的，不是所有的環境都適合這種設定。只有在無法達到正確輸出的環境之下，才有迫切使用的必要。

升頻部分記得 NOVAS DAC 1794 並沒有針對此點作太多改善與文章，而是著墨於其它部分就是。就自己所知，極高採樣率下 DAC 晶片性能會不升反降，精度亦會下降。而且，這個世界上存在絕對精準的時鐘嗎？88.2 除以 44.1 就一定是 2 倍？反之也一定是 2 倍？真的「整倍數」了嗎？還有忘了在哪裡看到的，有一個有趣的情況是，一本英國音響雜誌雖然極力推崇升頻。但其一邊極力推薦的同時，卻又在聽感測試的時候，將音質總評分數評的很低，這值得琢磨。升頻會加入噪音、失真，雖然部份失真在人耳聽覺以外的高頻地帶，但這些失真能量會波及聽覺以內的頻率，而且會加入本來沒有的極高頻噪音，所以沒任何好處。說不定自身以為聽到的細節，實際上只是噪音而已。

支持升頻與不支持升頻兩派直到現在都還在爭吵著，仍沒有一個公認的結論。除了理論的部分，最重要的還是聆聽者本人的看法，自己喜歡就好。針對個人需要作出適當的選擇才是正確的，盲目地追求或推崇所謂「專業」的功能或規格，並不是專業的人會做的。明白及找出自己需要什麼，才是比較好的作法。

[AFAIK]

1. 主觀聽感上沒有優劣，純粹看自己是否喜歡。
2. 同上。
3. 升頻的比較是沒有意義的，不是所有的環境都適合這種設定。只有在無法達到正確輸出的環境之下，才有迫切使用的必要。

同意

不過升頻也不全然沒有意義
像是 dac 內部也作升頻的動作，好處是可以降低 quantization noise
不過想想，用硬體作可以作到多好？ 記得看過一篇文章，Wadia 內部的升頻不過是用 B-Spline 作 interpolation
而其他一般 dac chip 用的計算就更簡單了...
意思是說，相較之下，用好一點的硬體（dsp chip、電腦）這部份的演算法可以弄到很強大

這方面有些不錯的資料可以看看：
http://www.hotechaudio.com/db%5CArticle ... 233354.DOC


下面這個蠻有參考價值，是 anagram tech 寫的（這家公司就是 Orpheus 的母公司）
裡面還提到 jitter 的一些觀念，還有如何處理 jitter 的方法
http://www.anagramtech.com/getdoc.php?d ... 5M-PL-100A
http://www.anagramtech.com/products/har ... -hifi/q5m/

主要就是通過自家的 upsampling 技術
如此一來可以解決 jitter 問題、繞過 dac 內部的 oversampling 部份、使用更好的演算法
不過這都是 anagram tech 說的，我沒用過，不知道啦

[AFAIK]

至於像是 ssrc 這類的
有一套很不錯，都沒有人提到...
http://sox.sourceforge.net/
也有人幫她寫成 foobar2000 的外掛
http://www.hydrogenaudio.org/forums/ind ... opic=67373

在最近的 downsampling 表現出色，看波形是超完美的 XD （VHQ Linear phase）
http://src.infinitewave.ca/

另外，SSRC有個重大的問題（這套很久沒有人維護了！？）
http://www.hydrogenaudio.org/forums/ind ... t&p=590201
從 22050 升頻 到 96000 Hz 時候，有嚴重的 artifacts 出現

[class6]

謝謝提供的資源.

請問您知道
allow aliasing
steep filter
phase response
這些選項的意義嗎?

謝謝.

[AFAIK]

謝謝提供的資源.

請問您知道
allow aliasing
steep filter
phase response
這些選項的意義嗎?

謝謝.

是抓了 foobar2000 外掛嗎？
那個檔案中有個說明檔，裡面有些解釋（不是很清楚，加減看一下吧）
東西寫的一長串，懶得貼在這裡了


要不要分享一下跟 ssrc 還有其他的 resampler 相比較的聽感阿

[chanlin]

個人有 Monarchy 48/96 upsampler ，這部只有 48 與 96 KHz 的輸出選項，接 44.1 不昇頻都不行。
也有 Novas The DAC1794，這部是 CS8414 收到後，後面一顆 chip 昇到 96K 處理吧！記得是這樣。
還有一部舊的白豆腐 de-jitter ，同軸入可以用 96KHz 。

用 DAC1794，
以前曾搞了許久接來接去的比較，映象中用 Monarchy 去 upsamplng 變化不是很明顯，
加入 de-jitter 以比較，目的是想排除數位線傳輸與輸出/接收晶片在高頻上的限制，
當時喇叭靠牆很近，音場並不完整，只比音質變化。
其差異不如換條類比訊號線或電源線。
當初比較時的音響設定環境比較差吧，不同數位線 (但都是 Belden 的) 是分不出差異的。

另一部 Lite DAC-38，略明顯地不喜歡加料 (upsampling) 的聲音。加料後記得是 3D 感會 "些微" 被破壞。


用 X-Fi 切換 44.1, 48, 88.2, 或切換不同使用模式，其影響都較明顯立現。
哪種好，個人覺得與錄音有關，
好錄音 44.1 對味，其他怪怪的。
一般錄音， 48 或 96 用軟體 upsampling 也不錯聽。
( X-Fi 是指數位輸出)

[class6]

自己查的結果:

allow aliasing 允許鋸齒
也就是關掉反鋸齒. 不要勾 除非慢到受不了.


steep filter 陡峭的濾波器.
將重新取樣的帶寬(band-width) 從95%變成99%. (偶不知道這是啥?)
這個值可以用命令列. 設為 74% ~ 99.7% .


phase response 時間軸上抖動的控制
http://www.andaudio.com/phpbb3/viewtopic.php?f=26&t=69162&start=0
http://www.av199.com/viewthread.php?tid ... eD1&page=1


某網友的說法:
linear...............for 交響樂
minimum...........for 流行歌曲
steep...............for 清唱 獨奏

intermediate是介於linear和minimum之間.

如果使用命令列的話.
phase response其實是可以從0設到100. 有100段的微調.
( 0 = minimum, 25 = intermediate, 50 = linear )

[Higuma]

TMNEXT兄在

viewtopic.php?f=26&t=69162

這串討論有提到一點,可惜我看不太懂,更詳細的可能還是得問他本人吧.

[class6]

TMNEXT兄似乎是不太想再過問音響版的紛紛擾擾了.

隨緣吧.
基本上使用是OK了啦.