[討論]硬體和軟體 升頻的聲音
發表於 : 週六 3月 28, 2009 5:20 am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
-----------------------------------------------------------------------------------------------------------------------------------------------------------------
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.
---------------------------------------------------------------------------------------------------------------------------------------------------------------
PS.小弟如發文內容有誤 麻煩指教
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
-----------------------------------------------------------------------------------------------------------------------------------------------------------------
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.
---------------------------------------------------------------------------------------------------------------------------------------------------------------
PS.小弟如發文內容有誤 麻煩指教
