Oleksandr Andrushchenko wrote:
We understand that emulated interrupt on the frontend side is completely not acceptable
Allow me to expand on that: Proper synchronization requires that the exact position is communicated, not estimated. Just because the nominal rate of the stream is known does not imply that you know the actual rate. Forget for the moment that there even is a nominal rate; assume that it works like, e.g., a storage controller, and that you can know that a DMA buffer was consumed by the device only after it has told you.
It's possible and likely that there is a latency when reporting the stream position, but that is still better than guessing what the DMA is doing. (You would never just try to guess when writing data to disk, would you?)
and definitely we need to provide some feedback mechanism from Dom0 to DomU.
In our case it is technically impossible to provide precise period interrupt (mostly because our backend is a user space application).
As far as I can see, all audio APIs (ALSA, PulseAudio, etc.) have poll() or callbacks or similar mechanisms to inform you when new data can be written, and always allow to query the current position.
[...] ok, so the main concern here is that we cannot properly synchronize Dom0-DomU. If we put this apart for a second are there any other concerns on having ALSA frontend driver? If not, can we have the driver with timer implementation upstreamed as experimental until we have some acceptable synchronization solution? This will allow broader audience to try and feel the solution and probably contribute?
I doubt that the driver architecture will stay completely the same, so I do not think that this experimental driver would demonstrate how the solution would feel.
As the first step, I would suggest creating a driver with proper synchronization, even if it has high latency. Reducing the latency would then be 'just' an optimization.
Regards, Clemens