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Sox Sideshow
BillT:
--- Quote from: Hendrik on August 01, 2016, 09:52:05 am ---If the algorithm is really smart it can move the noise into higher frequencies by using noise shaping, which is only possible if you have a higher sample rate.
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Doesn't that only affect noise in the ADC-DAC system? In real world recordings noise will be dominated by acoustic or pre-amp noise i.e. noise generated before the ADC, so noise shaping is going to be rather academic, especially if you use 18 bits or more.
flac.rules:
--- Quote from: BillT on August 01, 2016, 10:36:57 am ---Doesn't that only affect noise in the ADC-DAC system? In real world recordings noise will be dominated by acoustic or pre-amp noise i.e. noise generated before the ADC, so noise shaping is going to be rather academic, especially if you use 18 bits or more.
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I am sure many will argue that having a high quality DAC in general is pretty academic when it comes to the final playback quality.
pschelbert:
--- Quote from: dtc on August 01, 2016, 09:14:24 am ---A couple of informal comments.
In the early days, Phillips wanted to increase the sound quality of its CD players. There are many reports that they did this by putting in 3 extra samples between the 44.1KHz samples and those extra samples were is fact simply zeros. I do not know the details of what they did, but since there are so many reports of them adding 0's I would not dismiss it out of hand. In general we think of upsampling as some sort of interpolation, but there is at least some though that Philips did it differently.
We should distinguish between early upsampling and todays DACs. Early upsampling CD players did upsample to 96KHZ and 192KHz, for example. I still have one that has a switch for 96 or 192. That was definitely a way to use a less sharp filter and to move the filter noise to higher frequencies. When these CD players were also used with external sources or when stand alone DACs used these techniques, they also used the internal clock and thus were a way to lower jitter associated with the input.
Most modern DACs use Sigma Delta techniques to upsample to extremely high sample rates in the MHz range using 1 bit or a few bits. Some DACs upsample internally to some intermediate frequency like 192 KHz or 384 KHz before doing to the Sigma Delta process. Sox can be used to bypass that process, if you think Sox is better at it than the DAC. But that also requires knowing if the DAC does this intermediate upsampling and, if it does, what sample rate it upsamples to internally. For example, using Sox to upsample to 192 KHz when the DAC upsamples to 352 KHz would probably hurt the process. This depends entirely on the particular DAC. Likewise, upsampling to 192 KHz may have little or no effect if the DAC does the Sigma Delta process on the original 44.1 KHz signal.
Discussing moving the filtering to higher frequencies is appropriate to DACs that upsample to 96 KHz and 192 KHz, for example. However, it is only part of the process with Sigma Delta DACs, since the noise shaping of those DACs is often not a simple slow row off low pass filter. It is often a finely detailed low pass filter and many DACs offer different versions of those filters. The filters definitely move the side effects of the filter to higher frequencies but they also deal with other effects, like time smearing.
The discussion above about the time smearing algorithms that Schiit uses should not be simply dismissed as marketing hype. They claim to be using very different techniques than most other manufacturers and they have good credentials for their claims. That does not mean they are the best that can every be done, but they should not be dismissed. Their DACs with the new techniques are receiving very good reviews by a lot of users.
Bottom line, for me, is that this topic is worthwhile and it is also more complicated than most people realize. The Sox upsampling may or may not be useful depending on what the DAC does internally and the whole Sigma Delta process and the subsequent filters may be far more important than the initial software upsampling.
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Hi
if you insert zeros and then go through an interpolation filter (convolution) or you calculate the intermediate samples in a different way, is an implementation issue. Mathematics behind is the same.
What you need in any case is an antialisingfilter in the digital domain, for any upsampling to remove the alias frequencies. This is mandatory in any type of upsampling, no matter how the converter otherwise works. If you upsample a CD, the antialiasfilter must cut off to lets say -100dB to -130dB above 22050Hz.
Peter
pschelbert:
--- Quote from: Elvis133 on August 01, 2016, 11:01:39 am ---I am sure many will argue that having a high quality DAC in general is pretty academic when it comes to the final playback quality.
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However its true if you have a DAC from one of the big fours (TI, AKM, ESS, AD) you have not to worry, they know what they do and its execellent performance of all these chips. They filter correct and some have even user selectable filters (minphase, linear, sharp cut off, slow cut-off etc.)
Peter
kstuart:
--- Quote from: Elvis133 on August 01, 2016, 11:01:39 am ---I am sure many will argue that having a high quality DAC in general is pretty academic when it comes to the final playback quality.
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That would be fine if we listened to words, but in listening to music in 2016, there is more of a difference between DACs than any other part of the audio chain. (And this difference is no longer a matter of pricing.)
--- Quote ---However its true if you have a DAC from one of the big fours (TI, AKM, ESS, AD) you have not to worry, they know what they do and its execellent performance of all these chips.
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The chips are fine as far as they go, the implementation can make a significant difference.
People seem to have a blind trust that someone selling a device has a wide and deep knowledge of electronics and also has not cut any corners in order to increase profit margin (cf Bay Bridge bolts, Challenger o-rings, etc.).
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