Hi there, I would like to suggest an idea for improving the latency reporting to userland for audio api:
Can we use the arrival of audio IRQs to trigger the measurement of current buffer positions and high resolution timer timestamp combined with system time, and use these measurements to synchronise a delay-locked-loop (DLL) to provide sub-sample accurate estimates of buffer position in between the arrival of audio IRQs? In this way, the AUDIO_GETIOFFS and AUDIO_GETOOFFS ioctls could return sub-sample accurate buffer positions. I don't think the external facing api would even need to change at all. -- Damien Zammit ZamAudio
