I think everyone is in agreement that x-axis, or timing, plagues digital audio, and that it ought to be able to be dealt with. Getting bit perfect transport of bits is very much achievable and so regenerating the clock data ought to solve the problem better than expecting perfect cabling and interfaces.
The key issues in digital audio to resolve are:
1. How do you eliminate jitter just before, or in the DAC chip itself, to avoid jitter creeping back in subsequent transmission steps. Ethernet makes the most sense to me, because of how it works. Its just that not many audiophile firms have the knowledge to exploit ethernet so they faff about with USB, SPDIF, AES/EBU etc.
2. How high does the sampling rate need to be to make quantisation error immaterial?
3. How do you deal with digital filtering at half the sampling rate. Use DSD? Increase the original sampling rate? Upsample before conversion? Use a filter, not use a filter? If you filter, then what sort of filter?
It is pretty clear to me that the Playback guys mean dealing with these three issues as being 2-D, with the first issue being x-axis and the second and third being y-axis. Where they are claiming to be different is in the x-axis, ie in eliminating jitter. A claim many have made before them.
The only thing they are being clear about is that you should do it in the DAC, not separate from it. I think they are saying they are mapping jitter to change the bits, which isn't new, except that it is usually done in a separate stage to the DA conversion. It is not clear to me why doing it in the DA conversion step is such a good thing as it is generally better to keep the DA conversion step as simple as possible.
But until the Playback guys tell us a bit more about what they are actually doing this is just guesswork.
The key issues in digital audio to resolve are:
1. How do you eliminate jitter just before, or in the DAC chip itself, to avoid jitter creeping back in subsequent transmission steps. Ethernet makes the most sense to me, because of how it works. Its just that not many audiophile firms have the knowledge to exploit ethernet so they faff about with USB, SPDIF, AES/EBU etc.
2. How high does the sampling rate need to be to make quantisation error immaterial?
3. How do you deal with digital filtering at half the sampling rate. Use DSD? Increase the original sampling rate? Upsample before conversion? Use a filter, not use a filter? If you filter, then what sort of filter?
It is pretty clear to me that the Playback guys mean dealing with these three issues as being 2-D, with the first issue being x-axis and the second and third being y-axis. Where they are claiming to be different is in the x-axis, ie in eliminating jitter. A claim many have made before them.
The only thing they are being clear about is that you should do it in the DAC, not separate from it. I think they are saying they are mapping jitter to change the bits, which isn't new, except that it is usually done in a separate stage to the DA conversion. It is not clear to me why doing it in the DA conversion step is such a good thing as it is generally better to keep the DA conversion step as simple as possible.
But until the Playback guys tell us a bit more about what they are actually doing this is just guesswork.