Clock Drift between hardware sound card and snd_aloop

[Rob Nertney <rob@xxxxxxxxxxxxx>]

Hi all,

I have successfully built a sound capture card that runs at a given rate off its own clock. I receive the stream and do some post-processing on it, and I want to present this modified data to the user as a capture pcm.

I initially thought that snd_aloop would be the best idea, as I can pipe my post-processed data directly into it. However, the driver is apparently calculating exact timing of what "should" be the rate based off of CPU ticks/jiffies and the requested period/sample rate/etc in the requested output stream.

In my actual hardware I see cycle slips. Underruns and overruns based off of minor adjustments to the card's rate.

Is snd_aloop the appropriate driver to be using? Or should I create a new card, similar to snd_aloop, which is more specifically interrupted (and associated pointer locations, etc. modified) from the hardware card as well?

It seems like I should be able to synchronize snd_aloop's timing with my hardware clock, but I haven't found anywhere in the API to achieve this.

How else can I avoid this clock drift?

Thanks,

Rob

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