I thought that it would be useful to clarify some things about "hi-res" audio. There seems to be a lot of misconceptions about how higher sample rates and bit depths actually work, and how they play into audio quality. I know some of you guys have a deeper understanding of this stuff than me, so please don't hesitate to correct anything I get wrong here. I would like to enter this topic by discussing one of the main misconceptions about hi-res audio, which is: That higher sample rates mean higher resolution, like DVD vs BluRay. Or, another variation of this is: Higher sample rates have more signal information, because 96 samples per second (or whatever) is more samples than 44.1 samples per second, so that means it is more accurate and can sound better. To be clear, those are misconceptions. I see this a lot. Many people are attracted to hi-res because of the flawed (although intuitive) assumption that signal information is lost when we sample it for the conversion to digital. Note: (Edited for clarity) 1) For now I am only talking about the sample rate, not bit depth. Anything less than infinite bit depth does introduce error in the reconstructed signal. I'll talk briefly about bit depth at the end. 2) When discussing Shannon/Nyquist and reconstructing a signal "perfectly", that is only true in the mathematical domain. The actual real world implementation of a reconstruction filter is always an aproximation. So mathematically speaking, it has been shown by the Nyquist-Shannon sampling theorem that you can perfectly reconstruct the original signal (music) from samples, as long as your signal is bandwidth limited at half the sampling rate. What does that mean? CD audio, aka the Redbook standard, chose a sampling rate of 44.1khz (samples per second). This means that you can perfectly reconstruct the original music signal as long as you limit the frequency bandwidth to half that, ie 22.05kz. (Limiting bandwidth is a fancy way of saying that we have to cut off (filter out) any music with frequencies above the limit frequency, 22.05khz for CD audio). This is important to repeat - the original audio audio wave form can be perfectly reconstructed from CD audio if you low-pass filter it at 22.05khz. To reinforce this point, I am cutting and pasting a relevant section from the Wikipedia article on Nyquist-Shannon: "Intuitively we expect that when one reduces a continuous function to a discrete sequence and interpolates back to a continuous function, the fidelity of the result depends on the density (or sample rate) of the original samples. The sampling theorem introduces the concept of a sample rate that is sufficient for perfect fidelity for the class of functions that are bandlimited to a given bandwidth, such that no actual information is lost in the sampling process. It expresses the sufficient sample rate in terms of the bandwidth for the class of functions. The theorem also leads to a formula for perfectly reconstructing the original continuous-time function from the samples." So, finally, the point about a bandwidth limited signal is that if you don't do that, ie if you don't filter out the signal above the bandwidth limit (ie 22.05khz for the Redbook example), when you reconstruct the wave, you get what are called aliases, which are like reflections of the audio signal from above the bandwidth limit into the audible range. We can hear that stuff, so we do need to filter it out as best we can. (BTW, this type of aliasing is not the same thing as what we think of when we talk about graphics cards that do anti-aliasing. That's more about interpolating pixel shades to accommodate the fact that a display has discrete pixels, so diagonal lines look jaggy). Humans don't hear above 20khz, so the Redbook standard makes sense. It sets the sample rate just above what it needs to be, to be able to capture the range of human hearing. So, other than talking about bit depth, we're done, right? Our sampled CD quality music can be converted back to analog perfectly. And that would be true... The only wrench in this plan is that it turns out that it is not so easy to do the do bandwidth limiting perfectly with such tight requirements as those imposed by the choice of the 44.1khz sampling rate - we want to keep everything up to the human hearing limit of 20khz, but then anything above 22.05khz must be filtered out. This makes for a very steep filter (called a brickwall low-pass filter), because there is only a small ~2Khz amount of space from the frequency where we want to hear up to (20khz), to where we need to cut the signal off (22.05kz). It turns out that building really steep filters of this nature introduces its own problems that distort the signal and therefore affect audio quality. If it weren't hard to build the steep filter we would be done. Unicorns and butterflies, audio nirvana. (Well, you still need a very accurate converter to get those analog voltages out of the digital domain, but that's a different problem, not related to sample rates or bit depth). So, to try to avoid having to build such a steep filter, the idea of upsampling, or over sampling (nearly synonymous), was introduced. Upsampling DACs are designed such that they upsample the 44.1khz signal to a higher sample rate. This is done by inserting extra samples in between each existing sample. These extra samples have to be computed to match the shape of the real waveform - this is called interpolation, and is generally handled by an iterative algorithm that successively refines the computed approximation of the waveform. The more iterations the algorithm performs, the more accurate the interpolated samples will be. The interpolation functions converge to perfectly recreating the waveform at infinity (infinite number of iterations). These iterations are often called "taps" in a hardware implementation (so, Yggdrasil/Gungnir Multibit have 18,000 taps or iterations, BifrostMB 9,000). If you upsample the 44.1khz signal to double the sample rate, 88.2khz, now your filter needs to only be at 44.1khz (at half 88.2) for Shannon-Nyquist perfection. And since we know that we actually only care about frequencies up to 20kz, now we have lots of room - we can use a low-pass filter with a gentle downward slope between ~20khz to ~44.1kz, an easy filter to make that doesn't distort things. This solves the problem of making nice easy low-pass filters, but introduces a new problem - the interpolated samples are not a prefect match to the real waveform, because they have to be approximated. So now you have tons of work around designing these upsampling interpolation algorithms (that are also called "filters", godammit). All of the "DAC in a chip" DACs have to do these computations using limited silicon. Higher end stuff may have a dedicated cpu for the interpolation - for example Mike Moffat (@baldr) of Schiit worked together with some other impressive academics to create their own unique "closed form" filter that runs in a DSP chip. ("Closed form", as far as I understand it, which may be wrong, is a mathematical term that means that despite upsampling the original wave into a new wave that is an approximation of the original, they can still recover the original digital samples after their filter is done, before moving on to the bit->voltage conversion stage.) Thus, the only real weakness of the Redbook 44.1khz sample rate is that it is so tight to the frequency limit we care about (20khz) that it made it hard to build DACs, due to the steep filter requirement. If the standard had been 88.2khz, or 96khz, we wouldn't have a more accurate digital representation of our music. The wave form in either case can theoretically come out perfect - but it would have benefited us by making the job of building the low pass filter easy, and thus eliminated the need for tricks like oversampling. Briefly, on bit-depth: In short, bit depth translates into two things - the maximum dynamic range that can be expressed numerically, and noise from the quantization error of the discrete digital bit quantities. 16 bits gives 96db in dynamic range and the quantization noise floor is in theory inaudible... but this is arguable. The higher the bit-depth the more dynamic range, and the lower the noise floor. Technically, CD audio's 16 bit depth is pretty good - for example it exceeds the technical capabilities of vinyl by a wide margin. So, that said, increasing bit depth _does_ mathematically improve the audio signal, mostly by lowering noise floor, so there is some basis for arguing that greater bit depth is better.