[alsa-devel] External samplerate changes, UAC2 clock topologies

Daniel Mack daniel at caiaq.de
Fri Jun 4 14:46:35 CEST 2010


when I implemented the basics for parsing the UAC2 clock functions, I
came across some principal questions I would like to discuss.

As I wrote in the commit log, UAC2 supports a number of clock units
which can be combined to complex topologies. The single unit types are

 - clock sources (can be fixed, variable, programmable and synced to the
   USB clock)
 - clock switches, which let the host decide which one out of many input
   pins appears on its output pin
 - clock multipliers, which have a numerator and a denominator, and both
   can be read-only or read-write, respectively

A clock source has controls for its frequency and its validity,
a clock selector has a control to control its current state, and
a clock multiplier has controls for its numerator and denominator

The driver is currently able to walk the graph of connected units in
order to find CLOCK_SOURCE descriptors as end-leafs. This is mandatory
as all requests for sample rate queries have to be sent to this unit

This raises the first question, as the driver can only know which sample
rates are supported in the currently selected combination of selectors.
Once the user changes any of them (which is currently already possible
as clock selectors are exported as mixer controls), everything can
change. And this can even happen while the stream is active, because
there's currently no check about whether the clock selector is part of
the actively used path.

The second issue I see is that a clock can lose its validity. A
real-life example is an external S/PDIF connected device which provides
a clock and which is suddenly disconnected. Firmwares are expected to
notify the host about such cases, and these messages are trivial to
dispatch. However, I wonder how the driver should react on this. From
a user's perspective, it would be best to just make the driver find
another clock path which reports a valid clock source endpoint, changes
the sample rate accordingly and continues streaming. There would be
a gap in the stream of course, but at least it would not kill the
applications or require major exception handling in userspace.

I wonder which approaches are actually possible to implement, which
details in the ALSA core would need to be extended, and so on.

Any oppinions? Has this been done before for any other audio hardware
supported by ALSA?


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