[pulseaudio-discuss] [RFC - MP3 passthrough over A2DP 1/2] mp3 passthrough: core changes

[pl bossart]

> I fully agree, except that I don't understand what's wrong with
> GET_LATENCY. AFAIK, GET_LATENCY returns the time that it will take for
> the next sample that enters the system to get out of the speakers. How
> would GET_TIME differ from GET_LATENCY, and why would it be better?

GET_LATENCY finds out how many bytes are currently buffered, and once
you subtract this from the number of bytes sent to PulseAudio you get
the number of bytes played. Dividing by the sample size gives you the
audio time.
A GET_TIME message would fetch the sample count straight from the
sink, no matter how much was buffered, and no matter what type of data
you handle you can convert to a time value if you divide by the
sampling frequency. If there's no mixing the two messages are
equivalent for PCM data.
Cheers,
-Pierre