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CD 16 bit 44.1kHz 的由來
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發表於 2012-8-18 15:47:33
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CD 16 bit 44.1kHz 的由來
人的聆聽範圍約在20-20kHz, 根據Nyquist theorem，樣本必須為兩倍，即40kHz以上。最初數碼媒介不多，磁帶成為最佳選擇。磁帶有兩種標準： 525 lines at 60 Hz 及 625 lines at 50 Hz。要滿足此兩種標準便要找出其公倍數。
最後的結果是44.1kHz。這樣525/60及625/50磁帶均可用來錄音了。
60Hz磁帶有35條line不能用，可用的為490條line/frame or 245 line/field
sample rate of 44.1kHz=60 x 245 x 3
50Hz磁帶有37條line不能用，可用的為588條line/frame or 294 line/field
sample rate of 44.1kHz=50 x 294 x 3
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發表於 2012-8-18 15:51:35
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正常的標準 CD (Red Book CD) 音頻格式是 16bit/44.1kHz
其中 16bit 是 Bit Depth (位元深度) - 分辨率
而 44.1kHz 是 Sampling Rate (採樣率) - Bandwidth 頻寬
對於 HDCD (High Definition CD)，它是 20bit/44.1kHz
高質錄音室音頻文件 (Studio Master)，它們可能是 24bit/96kHz 或 24bit/192kHz
增加位元深度，理論上 Dynamic Range (動態範圍) 加大：
16bit - 96dB
20bit - 120dB
24bit - 144dB
Digital Resolution 數碼分辨率：
16bit = 2 ^ 16 = 65,536
20bit = 2 ^ 20 = 1,048,576
24bit = 2 ^ 24 = 1,677,7216
對於 2 Channel (立體聲兩頻道), Data Rate (資料頻率)：
16bit/44.1kHz : 44,100 x 16 x 2 = 1,411,200 bit/s = 1,411.2 kbit/s
20bit/44.1kHz : 44,100 x 20 x 2 = 1,764,000 bit/s = 1,764.0 kbit/s
24bit/96kHz : 96,000 x 24 x 2 = 4,608,000 bit/s = 4,608.0 kbit/s
24bit/192kHz : 192,000 x 24 x 2 = 9,216,000 bit/s = 9,216.0 kbit/s
因此，HR Audio 24bit/96kHz 及 24bit/192kHz 較 Red Book CD 16bit/44.1kHz
會有更多的音頻資料，分辨率更高，動態範圍更大
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3#
發表於 2012-8-18 15:53:54
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請清楚注意
最重要的參數是 bit depth 分辨率 16bit vs 24bit
而 Sampling Rate 採樣率是相對較次要 44.1kHz vs 96kHz / 192kHz
更要清楚，高質素的是指原錄音聲軌 Studio Master Audio Record (24bit/96kHz 及 24bit/192kHz)
並非後期加工的升頻制作 Up-Sampling Audio File 16bit/44.1kHz --> 24bit/96kHz
以 16/44.1 升頻至 24bit/96kHz Audio File 並無加多資料的，只加大 File Size，是假 High Resolution
其聲音質量及動態範圍，均不能與 24bit/96kHz Studio Master 相提並論及比較
所以醒目的朋友，千萬不可被升頻至 24bit/96kHz 的表面數字所誤導
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發表於 2012-8-18 15:55:55
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即使音頻文件已被 upsampled 到 24bit，升頻並不添加任何新的信息，一個 16bit 錄音仍只有 16bit 分辨率
同樣，一個 48kHz 的錄音 (2Channel)，即使它已被 upsampled 到 96kHz， 仍然只是有 24kHz 帶寬 (48kHz/2)
最初使用 upsampling 的原因是簡化後期 DAC 模擬濾波器 (D/A Digital Filter) 的工作
減少失真，從而也提高性能及提升聲音輸出的質量
(1) CD Quality : 16bit/44.1kHz --> 升頻 --> 24bit/96kHz --> DAC
(2) HR Quality : 24bit/96kHz --> DAC
原來的信號源分辨率是完全不同的
CD Audio File 升頻到 24bit，對性能一般的 DAC 仍有幫助的
但對高性能 DAC 則作用不大
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發表於 2012-8-18 15:58:02
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Comparison of Different Resolution Audio Tracks
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發表於 2012-8-18 16:03:25
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太 pro 啦
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發表於 2012-8-18 16:05:12
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raywan 發表於 2012-8-18 15:55
即使音頻文件已被 upsampled 到 24bit，升頻並不添加任何新的信息，一個 16bit 錄音 ...
原來如此！對CD格式有多一些認識。THX！
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發表於 2012-8-18 16:38:50
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44.1kHz 是權宜產品，後來錄音室用48kHz，倍數於電話通訊用的8或16Hz。
現時的44.1或48都算是主流產品。
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發表於 2012-8-18 17:36:29
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數字一大堆
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發表於 2012-8-18 17:44:00
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raywan 發表於 2012-8-18 15:55
即使音頻文件已被 upsampled 到 24bit，升頻並不添加任何新的信息，一個 16bit 錄音 ...
Raywan 兄, 請問 DAC 的upsampling 有什麼好處?
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發表於 2012-8-20 07:24:29
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謝謝提供資料。
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發表於 2012-8-20 08:59:42
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raywan 發表於 2012-8-18 15:53
請清楚注意
最重要的參數是 bit depth 分辨率 16bit vs 24bit
非常精简的介绍 ！
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發表於 2012-8-20 10:28:38
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本帖最後由 raywan 於 2012-8-20 11:11 編輯
eddiey4k 發表於 2012-8-18 17:44
Raywan 兄, 請問 DAC 的upsampling 有什麼好處?
升頻DAC製造商聲稱他們的產品，相比傳統的DAC能提高標準CD的音質。
畢竟，不管用什麼DAC將其轉換為模擬數據，CD 的數碼數據是相同的。最流行的理論來解釋升頻DAC的音質出眾是時間塗抹理論 (Time Smearing Theory)。與傳統的DAC不同的是，所有升頻DAC大多採用緩慢滾降 (Slow Roll-Off)的數字重建方法，而不是使用傳統DAC的尖銳滾降 (Sharp Roll-Off) 或磚牆重建 (Brick Wall Reconstruction) 的過濾器。
主要的原因是，在CD上原來的數字音頻數據以至相關時間塗抹已定了，後來的數字音頻播放系統處理根本不可挽回已劣化的時間塗抹。也不可能性可以消除或減少這個時間塗抹。採用緩慢滾降的升頻數字濾波器 (Slow Roll-Off Upsampling Digital Filters) 的特點是為改善音質，但並不是在播放系統中移除或減少時間塗抹或。
一個升頻數字濾波器可以視為一個“變質”的超高採樣數字重建濾波器(Oversampled Digital Reconstruction Filter)但有一個緩慢滾降率 (Slow Roll-Off Rate)。一個升頻的數字重建濾波器是故意違反信號處理理論。重建的模擬波形不再是與原本一樣了。此外加入的超聲資料破壞了基帶音頻信號波形的保真度 (Waveform Fidelity is compromised by the addition of ultrasonic images of the base band audio signal.)。
CD 中44.1 kHz的數字音頻數據的音質，可以由於採用一個“變質”超高採樣及緩慢滾降數字過濾器後，反為可提高音質(可聽性)，取代一個“原好”的快速衰減(Fast Roll-Off)和高阻帶衰減 (High Stop Band Attenuation)數字濾波器。
這是表明，這個“變質”超高採樣數字過濾器的超聲圖像輸出(Ultrasonic Image Output) 是負責提高音質，減少非線性失真 (Non-linear Distortion)，這些非線性，如抖動引起 (Jitter-Induced) 的非線性問題的非線性失真，這些數字失真是獨特且均存在於所有的數字處理系統的DAC中。反觀在模擬數據處理系統中並不存在的。
可以說，相比模擬音源比較常見的失真，人類的聽覺是對某些數字形式的失真更加敏感。
你要
“變質”但較"好聽"的超高採樣數字過濾器
或是
“真實”但較"不順滑"的數字過濾器
故 Up-Sampling DAC 是順應市塲需要的 "好產品"
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發表於 2012-8-20 10:40:35
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Upsampling vs. Oversampling for Digital Audio
http://www.audioholics.com/educa ... g-for-digital-audio
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發表於 2012-8-20 10:43:54
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本帖最後由 raywan 於 2012-8-20 10:46 編輯
Oversampling DACs and Bits
Oversampling is widely used in the DAC. The effects of oversampling at the DAC are advantageous to the design of the analog reconstruction filter that must be built, as we have seen previously. By having a high sample rate out of our DAC we can use a very simple, gentle analog filter to reconstruct our analog filter. This is important since we will be able to design an analog filter that is not only cheap hardware wise, but also has a nice linear phase response over the passband.
Another reason for oversampling is to reduce the effects of quantization noise. By oversampling, we can spread any quantization noise over a larger bandwidth while keeping our signal of interest in the same band. Our filter will serve to cut out the out-of-band quantization noise while keeping our original signal and thereby increasing our SNR. For each factor of four that we oversample by, we gain 6dB of noise lowering. 6dB represents approximately one bit of information. By oversampling, we can theoretically drop one bit for every 4x increase in sample rate.
The question of number of bits is another thing to consider. Does carrying extra bits increase the amount of information in our signal? Unfortunately, once we have sampled our signal, nothing can be done to increase the amount of information we have to work with. What carrying more bits does is that it prevents the loss of information. DSP algorithms and filters require additions, multiplications, and other math functions. If we are able to carry more bits in the results of these operations, we lose less information by chopping off fewer bits. Every truncation of a result will add noise to our signal. But now we can see that by balancing the number of bits we carry in our computations and by the amount we oversample, we can reduce the effect of this truncation in word length. One thing to note is that many products claim 24-bit word lengths, but yet only process internally at 20 bits.
What Does This All Mean - Will it Sound Better?
So the question remains whether upsampling or oversampling actually make music sound 'better'. How much do we need? We have seen the main motivation behind oversampling and how it allows us to use simpler digital and analog filters as well as helping us with quantization noise. The effects of upsampling are greatly debated. While it is true that upsampling does help us in attenuating the amount of jitter caused by sampling errors and an inaccurate clock, whether this jitter is audible or not is a point of contention. There is no doubt that wide bit words and super-high sampling rates that are touted by the latest products are largely marketing. Oversampling has been around for a very long time and has been used extensively in audio products to not only improve sound quality through 'better' filtering but to make these same products much cheaper. Upsampling, on the other hand, is relatively newer and debated greatly. The effects of upsampling are no doubt overstated. By carefully designing the sampler, ADC, digital processing path, and oversampling DAC, the upsampling and asynchronous rate transfer can, in my opinion, be avoided.
The Purists Point of View
There are basically two points of view regarding this upsampling an oversampling. The audio 'purists' want no additional processing on their signal and want whatever comes in from the source to come out as analog. They talk about zero oversampling DACs and such that are completely filter free both in the analog and digital domain. That is one extreme that some may argue is the purest since it avoids any digital artifacts and it's quality relies on human perception by arguing that the human ear in itself acts as a brickwall filter after 20 kHz. Whenever we get into debates of human perception, the math and theory go out the window. Does it sound better without all the digital processing and filtering even with the image of the signal sitting just past fs/2? The energy past 22.05kHz is still present and you are still sending it to the speaker's tweeter. How will the tweeter react to such out-of-band frequencies that are present? Furthermore, sending such a signal that is not limited in bandwidth could cause stability problems with wide-bandwidth amplifiers that have a high unity-gain crossing. The overall system's signal-to-noise- ratio will be adversely affected as well. The DAC will also introduce frequency spurs all over the place. If we don't filter them at all, what will their presence do to the sound? It's a complicated problem and such a minimalist approach could introduce more non-linearities and negative effects, more so than the digital processing ever would.
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發表於 2012-8-20 11:34:07
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又學到嘢不過好深要慢慢消化
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