Adaptive resampling in gstaudiobasesink.c

[Pavel Hofman]

Hi,

Please do I understand correctly that the slave clock alignment methods 
in gstaudiobasesink.c never touch the actual samples and handle only 
various ways of adjusting pointers into the ringbuffer? Including the 
GST_AUDIO_BASE_SINK_SLAVE_CUSTOM method.

Calculation of the master/slave clock difference is already available in 
gstaudiobasesink.c. Adaptive resampling (like used in PA, alsaloop, 
etc.) would require having two buffers with input and resampled samples: 
in -> resampling -> out. If the out buffer were available, calling e.g. 
src_process of libsamplerate would not be so difficult, IMO.

Please how complicated would be to add the out buffer and let the 
implementations for specific audio outputs (e.g. GstAlsaSink) use it 
instead of the buffer with incoming data? I am afraid it is not trivial 
due to the "inheritance" within the code

struct _GstAlsaSink {
   GstAudioSink    sink;
...

} .

Perhaps the ring buffer would become the "out" buffer and the resampling 
code had to read from another buffer written by the "input" side of the 
plugin.

Thanks a lot for any suggestions and comments.

Best regards,

Pavel.