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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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