Hi, Would there be benefit in separating the DMA and the transfer of audio samples to the sound card hardware from the delay and hw position feedback. For example, If the playback position is 0, the hardware has probably already transferred one period of the sound card. So, one cannot write to playback position 2 and expect it to be output to the sound card if the period size is 1024. For transfer purposes, one only needs to know that DMA transfer is complete on period X so that period X can now be over-written. I believe that this is what the DMA interrupt indicates. One needs to reliably detect a missed interrupt. It one took this approach, one might be able to hide the "missed interrupt" problem. When the interrupt finally did arrive, we would know where the next free period X is, and continue writing samples to that, irrespective of what went wrong before. This would allow for auto-recovery of the audio stream without having to stop and start the ring buffers again. One then needs to determine how one can get accurate "delay" values back from all sound card types. The "delay" value is used for applications like ensuring that audio and video play in sync. The problem ones at the moment appear to be the intel8x0 and hd-audio ones. There may be benefits from making the "delay" value return in nanoseconds instead of samples, particularly if gettimeofdate() is being used to try to produce accurate "delay" values. Kind Regards James _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
