[alsa-devel] How to properly handle variable samplerate?

[Petr Kulhavy]

Hi,

I'm developing an audio stream player application with ALSA output, the 
player has a large software buffer. The trouble is that the stream can 
potentially change the sampling rate (e.g. streaming RTP from a VLC 
playlist created from different MP3 songs). How to properly handle this?

I'm thinking of the following concept:
The codec feeds data into the software buffer together with markers for 
sample rate changes. The ALSA playback is asynchronous, using a callback 
function feeding the data from the software buffer into ALSA. If a 
samplerate marker is hit the new rate is set.

Can the sample rate be set from a callback? What other constraints are 
there? And how does the HW FIFO handle samplerate changes?
Or is there a better way of implementing?

Thanks a lot!
Petr

-- 
Petr Kulhavy, MSc
System Architect

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petr at barix.com <mailto:petr at barix.com> | Skype: brain.barix

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