Quote:
Originally Posted by Firam
That doesn't really answer his question. I would also like to know what resampleing is. So what if it is moved from 44.1k to 96k? How does that change the sound?
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Resampling is when you change the sampling rate of the digital music stream. CD's are sampled at 44.1KHz and we can resample that to other sampling rates, like 48 KHz or 96 KHz. There are different ways to resample, the basic way to upsample is to simply place a bunch of zeros inbetween the original samples. This will preserve the original signal fine but it gets a little messy when you want to resample by non-integer multiples. They will sometimes use interpolation to fill the zero samples as well.
The idea behind resampling for the Creative soundcards is due to the fact that the native sampling rate of the card's processors is 48KHz. So, when the soundcard plays back a CD, it resamples it to 48KHz, processes it (even if it is just doing pass through), and then outputs the audio (I don't know if it resamples it back down to 44.1 before sending it to the output. The digital out of the Creative cards is in multiples of 48KHz so there doesn't seem to be any reason for the card to downsample). Many people feel that the card does a poor job of upsampling the signal to 48KHz, so they prefer to use software resamplers to do this before the signal is sent to the card. By sending the card a 48KHz stream, the card will no longer need to resample.
The reasons for upsampling in CD players and DACs are along a different line of thinking. The output from a DAC is not a perfect reproduction of the original analog signal. Sometimes it will be a stairstep (zero-order hold DAC) or it will have some degree of interpolation. However, by applying an analog filter, we could theoretically perfectly reproduce the original analog signal even from a zero-order hold DAC. The problem is that the required analog filter would use an infinite amount of components. Instead, the analog filter is an approximation, where the more complex (and costly) the filter is the more accurate the results. So there are some tricks that one could do that will decrease the complexity of the analog filter without sacrificing accuracy. One way is to have a digital filter applied to the ditigal signal before it is sent to the DAC. This way, the analog filter can be a simpler filter. Another way is to upsample the signal. Upsampling will never add information to the original signal, but you can upsample without degrading the signal. Zero-padding (where the extra samples are just zeros inserted between the original samples) does not degrade the signal for example. The advantage is that by upsampling, you increase the bandwidth of the signal. But the added bandwidth is not used because the original signal did not have any information in that region. For example, CD's have frequencies from 20Hz-20Khz so if we upsample from 44.1Khz to 88.2Khz, then we now have a region from 20Khz-44.1Khz of unused frequencies. The filter now can be relaxed so that if there were signals in the unused frequencies, they would be affected by the filter, but not in the way needed to be properly reproduced. But since we do not have any information in the added regions, we can let the filter be anything in that region. The result is that we only need to control the filter now over a smaller bandwidth region instead of the entire bandwidth. This allows them to use less complex filters to retain the same level of accuracy.
So by upsampling, you can create a DAC that is more accurate than a given straight 44.1KHz DAC for the same money and resources.