[Libav-user] Changing the sample rate of a resampling filter on the fly.

[Peter Spierenburg]

Greetings,


I have an application where I need to control the output sample rate of an audio resampling filter on the fly, with parts-per-million resolution.


I've found an example that gets close at:


https://github.com/andrewrk/libavfilter-example/blob/master/main.c


I've even managed to convince it to accept arbitrary sample rates with parts-per-million resolution. To be fair, I cannot hear the difference between 64000 samples per second and 64000.000001 samples per second. But I can hear the difference between 64000 samples per second and 40100 samples per second.


What I need to figure out is:


- Is it possible to change the resample rate of the aformat filter (lines 61-72) on the fly somewhere in the audio_decode_frame() function?


- Does the aformat filter actually output fractional sample rates, or merely round to the nearest integer rate?


- Is there a filter that is better suited to this task?


Peter.

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