关于采样率－为什么要采用高达192kHz的采样率？

在业界有三套采样率并存：
44.1kHz及其下采样、上采样：22.05kHz, 11.025kHz，88.2kHz, 176.4kHz
48kHz及其下采样、上采样：24kHz, 12kHz,96kHz, 192kHz
8kHz, 16kHz, 32kHz, 64kHz

人耳的听觉是有限的，介于20Hz到20kHz。跟据Nyquist采样定理，采样频率只要超过信号带宽的2倍就不会产生混迭。在数字媒体领域，如音乐CD的规范，都是以44.1kHz作为标准采样率的。因为44.kHz大于20kHz的两倍，所以实际上44.1kHz的采样率是足够用的。

但是现在普遍在工程中都是使用48kHz或者96kHz频率录音，只有在最终母带处理时才会转成44.1kHz的CD格式，这样减少多次采样率转换造成的失真。

而在电脑领域，作为音频硬件codec标准的AC97规范只规定了48kHz。这造成几乎所有的输入、输出信号都要被重新采样（专业术语叫采样率转换，即 SRC）。SRC一般都会造成音质的损失，较简单（即较差）的SRC算法会造成音质明显劣化。但这已经是一个既成事实了。
 
既然44K够了,那为什么还要用192KHZ来录音?

首先，20kHz只是大多数人的听觉门限，也就是说，人耳对于20kHz以上的声音很不敏感。注意不敏感并不意味着完全无法感知。大多数乐器（特别是钢琴和弦乐器）的乐音含有丰富的高次谐波，用音乐术语来说即所谓的上方泛音。截止频率为22.05kHz的CD音频，的确会给听惯了真实乐器的人一种不自然的感觉，尤其在高频部分，因为奈奎斯特截止频率造成更高频率泛音的信号失真。

其次，数字录音通常都需要进行后处理。音频处理会对信号产生进一步的失真，包括信号畸变、频谱混叠，等等。如果录音时仅仅用44.1kHz对原始信号采样，那么在后处理前还得进行上采样（up-sampling），对采样频率进行扩展。由于这种扩展是“假”的，实际上并没有更多有用的原始信号，并且上采样算法的优劣也会影响原录音信号的失真，所以这个做法并不可取。因此，通常的做法是用更高的频率进行采样。

而现在的完全专业数字录音棚中，则不再按CD标准的规范录音、混音以及母带，而是优先使用HD音频规范。即：
采用24Bit 48KHz、24Bit 96KHz、24Bit 192KHz 三种规格进行录音，当然，24Bit 48KHz是一些小的录音棚使用，因为他们的处理器资源有限。而大的录音棚，都清一色的使用24Bit 96KHz和24Bit 192KHz 进行录音。
那么，这样的录音规范，有什么好处？
1.符合HD音频标准，这也是将来的主流标准，制作出的成品，可以直接应用于HDCD、DVD-Audio、蓝光唱片、数字音乐下载业务、数字对媒体播放机业务。
2.完全照顾数字影视视频业务，多声道电影视频都会采用HD音频规范。包括移动便携数字视频设备都用它。
3.完全照顾消费性音频播放业务，比如：因特尔HD-Audio音频标准，AC97音频编码解码，便携MP3/mp4/电话/游戏机最高音频质量播放。
 
目前，专业录音行业的最高质量标准就是：24比特定点位深、192000Hz采样频率，简称“24Bit/192KHz。”。当然，将来这个标准依然会继续提高，向32Bit 384KHz进发也是可能的。
 
实际上，现在的CD唱片市场上卖的产品（正版），最低级别的通常都是HDCD唱片，你买唱片时都会发现基本上都是HDCD标识，也就是一张激光唱片包含两种音轨：普通CD音轨和HDCD音轨。其中CD音轨记录16比特44.1KHz信号（这是这张唱片的兼容内容，照顾早期的CD播放机），HDCD音轨则记录24Bit 96KHz信号（这才是该唱片的主要内容）。普通的CD播放机只能播放CD音轨信号，而HDCD音轨则需要HDCD播放机才能播放（实际上现在的绝大多数 DVD播放机都能播放HDCD，而现在的电脑则更没问题了。）
 
下面是摘自论坛的一些有趣的讨论，大多数的人（跟我一样）搞不清楚为什么要用高的采样率，比如96kHz,甚至192kHz,认为这是出于对高采样率的迷信，为了迎合盲目的消费者，认为这么高的采样率是浪费，是没有必要的。也有人试图解释这背后的理论性和技术性原因， 包括：1）虽然大多数人听不到20kHz以上的声音，但是不代表所以的人都听不到；2）人耳对瞬时信号（高频）相对于稳态高频而言更容易感知，瞬时高频往往带宽较宽，有时候其带宽可以包括一部分人耳可以听到的频率。虽然低通滤波器可以保留这部分频率，但是总归不如全部保留下来好。
还有人从过采样的角度来分析，认为有减少噪声的作用，还可以降低模拟前置滤波器和模拟后置滤波器的复杂度。我认为，过采样的采样率一般情况下远远比192kHz要高，而且过采样的高采样率仅仅体现在采样过程中，在信号处理和保存时将到了正常采样率（比如44.1，48，96，192，etc），这本身是过采样的问题，跟本主题讨论的问题不太相关。

上面给出的理由还不全面，一旦发现合理的解释我再行更新。


http://www.dsprelated.com/showmessage/96236/1.php
What's the use of a 192 kHz sample rate?
为什么要用192kHz的采样率？192kHz采样率比44.1kHz有什么优势？人耳不是不能够听得到192kHz采样率的全频率吗？
Why does DVD-Audio use 192 kHz sample rate? What's the advantage over 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?

I agree there are a small percentage of humans who can hear above 20 kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a maximum frequency of 96 kHz. There is no known case of any human being able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz
sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just stupid.

So what’s the justification for using 192 kHz? Is it just a total waste of bandwidth and energy? Any proof to the contrary?

Thanks,

Radium


Some objective reasons客观原因:

1) It's used in professional audio to accommodate nonlinear processing such as dynamic range compression.  The gain changes imposed by dynamic range compression are mathematically equivalent to modulation. Modulation produces sidebands.  Those sidebands *may* alias, under the
right conditions.  There have been some published papers on this. Though this may justify 96 kHz, it probably doesn't justify 192.

2) There's some evidence that humans' ability to hear high frequency transient signals exceeds their ability to hear high frequency steady state signals.  Somebody said he can hear steady state tones up to about 16 kHz, but he can hear music that contains tone bursts consisting of 6 cycles of a windowed (appears to be Hann) sinusoid at frequencies up to beyond the limit of his hearing. Though he perceive them as clicks, he definitely hear them.  Again, though this may justify 96 kHz, it probably doesn't justify 192. 

3) Others less verifiable, such as the claimed audibility of pre-ring associated with linear or non-minimum phase anti-imaging filters. Basically the higher sampling rate may allow more options here.  If true, this *might* justify 192 kHz.  Maybe.

Some subjective reasons: 主观原因:

1) Marketing.

2) Bragging Rights.

3) Superstition.

4) Magical Thinking.

5) Self-Delusion.

6) Mass Hypnosis.

Personally I think that something like 64 kHz (possibly 128 kHz for mastering) would have solved all of these problems adequately.

-- Greg

>2) There's some evidence that humans' ability to hear high frequency
>transient signals exceeds their ability to hear high frequency steady
>state signals. 

I forgot to mention:  This can probably be fully explained by the fact that transient signals have a nonzero bandwidth associated with them. If that bandwidth extends down into the audible range, then, of course, the signal becomes audible.  From that standpoint, filtering-out the portion of the transient that is above the audible range should not affect the audibility of what is left.  Still, from a waveform fidelity point of view it might be beneficial to keep more of it.

Greg


I think there is great merit in sampling at 192k/s. These days a 192k/s 24 bit stereo DAC offers excellent noise and distortion specs. Such a high sample rate really makes the analogue filtering a lot easier. A 192k/s ADC is not much more expensive (a difference probably driven more by volume than complexity).

Actually transmitting and storing such sample rates makes no sense at all. 44.1k/s was a bit marginal, when you allow for the impracticality of the filters getting really close to 0.5fs. However, 48k/s should be good enough for any practical purpose.

For people who say supersonic sound can't play a part in a listening experience, trying being in a room with a high intensity of supersonic energy. Under some conditions (I'm not clear which) you can sense it, even though you can't hear it. It actually feels like something loud that you can't hear is going on. It’s a very odd feeling. That said, I've never found any evidence that this plays a part in any musical
experience. I see no reason to try to capture that energy in a recording, unless you feel your dog should enjoy a greater musical experience.

Steve



> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over 44.1
> kHz? Humans can't hear the full range of a 192 kHz sample rate?

I think that the answer is aliasing avoidance. Take it this way:

- The audio band pass is limited to 16KHz, say 20KHz to get some extra marging for the most perfect ears on earth.

- As far as I know ANY audio digitization circuit uses a low pass filter at around 20KHz, so even a 192Ksps ADC or DAC will be band pass limited to 20KHz signals, as there is absolutely no need to manage audio signals with a higher frequency.

- If you use 44Ksps then you must insure that there is no power above 44/2=22KHz thanks to M. Nyquist, so your low pass filter must have a very sharp transition. As the filter will never be perfect you will get aliases. For example even if you use a 12th order filter (already difficult and
expensive to build) then the attenuation will be "only" 72dB/octave, meaning that a 16KHz low pass filter will have an attenuation of only 50dB or so at 22KHz. And 50dB is not enough for good listeners as a -50dBc "noise" is clearly audible.

- However if you use a 192Kbps sampling rate then the required performances on the low pass filter are drastically relaxed. This filter can keep a corner frequency at 16 or 20KHz, but even a 6th order filter will provide a at 86dB attenuation at 192/2=96KHz...

And as a 192Ksps sampling rate is far cheaper to build than a very very good low pass filter... That's the beauty of oversampling...

Does it make sense ?

Cheers,
Robert Lacoste
www.alciom.com
The mixed signal experts



>Does it make sense ?
>
>Cheers,
>Robert Lacoste
>www.alciom.com
>The mixed signal experts
>

Not a lot. As far as I'm aware there are NO ADCs that sample at the data rate of the output signal. For example the 44.1ksps ADC in my PC samples at 2.8224MHz. When you sample at that rate it is  trivially easy to make a gently sloping analogue lowpass filter that guarantees a lack of alias products. All further filtering and decimation is done digitally, where it is easy. THAT is what oversampling is all about, not using a 192ksps sampling rate.

d
--
Pearce Consulting
http://www.pearce.uk.com


Oversampled conversion does not require one to *store* information at the oversampled rate.

--
Oli


>Oversampled conversion does not require one to *store* information at
>the oversampled rate.

Fully right, but it is a low cost solution if you want to avoid the cost of a digital low pass filter & decimator...

Robert


> Fully right, but it is a low cost solution if you want to avoid the cost of
> a digital low pass filter & decimator...

The cost is in any case low, and a little extra cost in the acquisition chain is amply repaid in reduced storage cost of all the copies.

Jerry
--


On May 5, 8:14 am, rajesh <[e-mail]> wrote:
>
> I don't mean literal repetition.Take for example a sine wave
> of 10khz and sample it with 1 mhz. One does think of
> samples getting repeated are at least they are close.

okay, rajesh, i'll try to clarify (or muddy) the waters here:

for a single sine wave (if it is already established that this is what you're looking at - a single sine wave), three sufficiently close samples (and spaced by 44.1 kHz or 1 MHz sampling rate are both
sufficiently close) will contain all the information you need.  Of course if there are errors (like quantization errors) in those three samples, you'll get a different sine wave reconstructed.

so rather than looking at a single sine wave at 10 kHz, let's consider a general waveform, but with one restriction of generality: a waveform *bandlimited* to just under 10 kHz.  we will call that bandlimit,
"B".  now, if you're sampling at 1 MHz, it wouldn't be precise to say that there are samples getting repeated, but it is true that there is redundancy.  that is 50 times oversampled because it only needs
samples once every 1/20th millisecond.    you don't have 49 copies of each necessary sample, but the 49 samples in-between each 50th sample can be constructed from only the knowledge of those samples spaced by 1/20th millisecond.  so there is a *sense* of repeated samples (in
that in both cases of repeated samples and this oversampled case, 49 outa 50 samples is redundant), but it is, in fact, not the case.

now we *do* know that oversampling does, in the virtually ideal case, reduce noise and this (along with noise-shaping) is one of the neat properties of sigma-delta conversion.  for an N-bit converter, the theoretical roundoff noise is

    roundoff noise energy = (1/12)*( (full scale)*2^(1-N) )^2

that energy must be the area under the curve of the noise power spectrum.  note that the sampling rate is not a function of that.  If the roundoff noise is truly random in nature (a bad assumption for
very small signals, but not so bad for signals closer to full scale), then we think that the noise is white or flat, from -Fs/2 to +Fs/2. so, if the area under that constant function is this constant

    area = (1/12)*( (full scale)*2^(1-N) )^2

and the width is (+Fs/2 - (-Fs/2)) = Fs, then the height is

     1/Fs * (1/12)*( (full scale)*2^(1-N) )^2

so, as Fs gets larger, the height if this noise level (the amount of noise per Hz) gets lower, and if you can sacrifice some of the spectrum above your bandlimit, you can filter out all of the noise
from your bandlimit to Fs/2 and the level of noise has been reduced by a factor of B/(Fs/2) which is the reciprocal of the oversampling factor, Fs/(2B).  this is not unheard of, but it comes into play when
there is a limit to the number of bit in the word, N.  if that is the case (you have a very fast converter with fewer bits) you can make meaningful samples with word width wider than the A/D converter.  but
you have to oversample by a factor of 4 to get one extra meaningful bit.  that's how the math works out.  (this is not assuming noise-shaping.)

now, in audio practice, in the studio they get super-high quality A/D converters with, say, around 24 bits and sampling at a much higher sampling rate, maybe 192 kHz.  this is for initial recording, mixing,
editing, effects, etc.  i don't disapprove of them throw extra bits at this whether the need is disputed or not.

but eventually they are going into a CD of 16 bit words, two channels, and a 44.1 kHz sample rate.  that's 1411200 bits per second flying out at you. (or a DVD or SACD with a lot more bits per second.)  now, if the sample rate is increased (thus increasing the bit rate), how are we gonna "compare apples to apples"?  if the bit rate increase with the sample rate, and if your bit error rate measured as bits of error/second remains constant, of course it will sound better as you increase Fs.  if it's 1 MHz sampling vs. 20 kHz sampling and you drop one sample per second in both cases, the 1 MHz data will care a lot less (in fact, 50 times less) than the 20 kHz data.  you are knocking out a larger portion of the data in the 20 kHz case.

but what if the portion was the same?  what if, in the 1 MHz case, you lost 50 samples per second compared to the 20 kHz case where you lose 1 sample per second.  which is worse?  in a recording or transmission environment, which is the case?  can you expect an equally noisy channel to have fewer bit errors per bit of data for the high sampling rate case?  what would be the mechanism for that?

r b-j



Suppose you want to digitize audio in the frequency range up to 20kHz.

If you digitize with a sampling rate of 48000, you need to use a filter which stops all frequencies above 24000Hz, but allows through all frequencies below 20000Hz.

However, if you digitize with a sampling rate of 192000Hz, you need a filter which stops all frequencies above 96000Hz, but allows through all frequencies below 20000Hz.

This is, in principle, an easier filter to build; and it's possible that may result in the quality of the filtered analog (pre-digitization) audio being better for the 192000Hz sampling than it is for the 48000Hz sampling.

So I think there are two separate questions:

1) do devices sampling at 192000Hz result in better digital files than devices sampling at 48000Hz? To answer this question, you need to use different sampling devices, as well as different sampling rates.

2) if you sample at 192000Hz, does it sound better if you store and play back the data at 192000Hz compared with downsampling to 48000Hz and storing and playing back at 48000Hz? To answer ths question, only one device is needed.

I think in practice your comment addresses the second question, but not the first.

Tim
 
 
还有一些讨论：
The Difference Betweeen 96khz & 192khz
http://www.tomshardware.com/forum/49110-6-difference-betweeen-96khz-192khz