Just would like to have a simple explanation about the "resampling" thing in ESS DAC's and the absence of that in AKM DAC's.
Does that mean that ESS "manipulate" the sound that it is fed with to make it "sound better" ?
I thought if you "upsample" or "downsample" or "resample" stuff, then you necessarily add/substract information from original signal... Thus the sound cannot be as it is meant to be.
Or, as this refers to "synchronous digital inputs" does that mean that the input signal coming in 44.1 is resampled to 48khz because that is what the DAC accepts (or the other way round) ?
Just to reiterate the excellent info provided by the previous responders to your questions, and to elaborate a bit only in order to combat my reputation for concision and brevity in my posts. There are three distinct resampling processes that you may be lumping into one in your mind. Firstly, there is the sample rate conversion that the Operating System of the source may do, either to convert the sample rate of the source files/stream to match the sample rate capability of the DAC to which they are fed (either internal DAC or external), or to enable mixing of sounds with different sample rates (by conversion to nearest multiple of 48kHz), or else to limit the bitrate of the digital stream output to S/PDIF Coax or TosLink. Any such sample rate conversion is done by the OS audio layer when it receives data from the music player software and prior to sending the resampled audio data to the DAC device driver. I assume this sample rate conversion is done by the OS using its own routines and the CPU hardware, and when done for mixing sound channels or for output to S/PDIF it is independent of the particular DAC chip. This is where the whole "bitperfect" hoopla comes in, and people who are into HiRes audio fret about it more. Consumers have done a whole lot of digging into how to prevent the OS from resampling and instead send the bitperfect data directly to the DAC. It turns out this is difficult and frustrating, and involves the OS, the music player software and the DAC's device driver. Under Microsoft Windows, you have to get the music player to use exclusive mode of ASIO or WASAPI, and under Android it is even more difficult to bypass the resampling of the Audio layer. I have not bothered looking into it so far, except that the software on some of my DAPs is designed for bitperfect feeding of the DAC chip/s.
Secondly, there is the "resampling" (more resembling a reclocking) that Amir referred to in his Topping D70s review. This resampling happens, if it does, on the DAC chip side during the audio data transfer from the source to the DAC chip or a receiver chip like AKM4118. This is the ASRC done by the ESS DAC chips (which include a hardware ASRC unit) to the USB/Coax/TosLink stream received by them, but not by the AKM DAC chips. Post #37 in the D70s review thread explains that the resampling is more artful than just using the hardware ASRC for clock domain synchronization, and is more like an on-the-fly improvement of the syncing of source and internal DAC clocks. This is done to correct timing errors (jitter) in the audio bitstream, and gives the ESS DAC chips their superior jitter performance on Coax and TosLink inputs relative to the AKM DAC chips. In Post #4 Vincent Kars showed how use of ASRC improves jitter performance. While the above is true, in practice the jitter performance of AKM DAC chips is still excellent, and the jitter is inaudible, so nothing to worry about.
Thirdly, there is the upsampling/oversampling performed by all modern delta sigma DAC chips, including the ESS, AKM and Cirrus Logic families, an upsampling that happens within the DAC chip itself as it processes the audio data that it received from the source. As Vincent Kars mentioned, this upsampling is at the very basis of their superior performance compared with alternative DAC chip architectures such as R2R ladder resistors. The delta sigma modulation takes advantage of the engineering happenstance that the state-of-the-art analog relative time resolution of electrical circuit clocks is far more accurate than relative voltage level resolution of the analog output circuits. The delta sigma modulation upsamples along the time axis, and on the face of it downsamples the signal amplitude (though it does not actually lose amplitude resolution but instead maps it into the higher time resolution). This is followed by digital noise shaping and an analog low-pass filter that gets rid of the ultrasonic noise caused by very rapid oscillation of the DAC chip's unfiltered analog output among the coarsened amplitude levels. This is drastic manipulation of the data bits, very far from leaving the data untouched, yet maintaining audio signal fidelity quite well, but the upshot of it all is that it works very well, and recent generations of delta sigma DACs have proven quite superior in transparent performance and cheaper to make, as compared with the best examples of alternative DAC architectures.
I don't want any up/re/downsampling. Can I have that on a device that money can buy (decently) ?
With substantial research, you can avoid the software digital resampling performed by the OS (the first type of sample rate conversion). The second type, the ASRC performed for sync improvement, is actually beneficial in reducing jitter in Coax and TosLink connections, and there is no sense in going out of your way to avoid it even if you had the option to do so. I do not think it actually changes the sample rate; my understanding of it is that it just adjusts the sampling instants to more reliably read the digital audio data from the input stream of the DAC. The third type of digital resampling, the upsampling performed in modern delta sigma DAC chips, is also beneficial and is the enabler for the fidelity with which they convert digital to analog. The use of a NOS DAC or selecting a NOS filter on a DAC defeats upsampling capability, and with lower bit-rate audio data, it will audibly degrade the sound fidelity (but may be perceived by its afficionados as improving the musical quality). With very high bit-rate audio data (HiRes files or streaming), a NOS filter will still degrade the sound fidelity, but the degradation would be inaudible. If you are bent on avoiding this third type of up/re/downsampling, you can do so by choosing a R2R DAC instead of a ESS/AKM/Cirrus Logic one. Take a look at the performance and price of an excellent recent discrete R2R DAC reviewed by WolfX-700:
Measurements of Musician Pegasus R2R DAC.
So, to summarize, as the others have pointed out, your Aune X8 is audibly transparent, and should cause no worry about its manipulation of the audio data. Enjoy your Aune X8.
But the DAC also has a digital output.
If I hook that to another DAC, the input of the second DAC shows the same as the input as the first DAC.
But maybe that digital ouput is just a "passthrough" ?
Yes it is a passthrough (except converted to S/PDIF format), and unrelated to the data manipulation that would be done internally by the DAC chip if you were to use its analog output instead. High sample-rate/bit-rate input data to any DAC unit would be downsampled to the maximum allowed S/PDIF "passthrough" (96kHz/24-bit or 192kHz/24-bit, I forget which; mfr may set lower limit), which content owners stipulated to prevent bitperfect copying of HiRes audio. HDMI output would not limit the bitrate, but mandates use of HDCP, in order to prevent bitperfect copying.
