[pulseaudio-discuss] [PATCH] protocol-native: set a minimum audio latency for a "phone" input stream

[Arun Raghavan]

On Mon, 2011-06-13 at 19:37 +0300, Tanu Kaskinen wrote:
> On Mon, 2011-06-13 at 11:08 +0800, Lin, Mengdong wrote:
> > I got buffer underruns when moving sinks for a "live" stream. Flag
> > "adjust_latency" will decrease the latency each time when I move the
> > sink and can cause buffer underrun.
> 
> That sounds like a bug. It doesn't make any sense that the latency
> should decrease every time when moving the stream.
> 
> > http://lists.freedesktop.org/archives/pulseaudio-discuss/2011-June/010146.html
> > Is it OK to set a minimum total latency (300ms) for a "live" (phone)
> > playback stream, considering that a BT headset sink has a fixed
> > latency of 128ms?
> 
> No, this is just working around a bug, not fixing it.

What Tanu said -- this sounds like a bug, and Amanda's analysis from
earlier seems reasonable (still need to look at the actual code,
though). If nobody beats me to it, I'll try to verify Amanda's analysis
and try to write up a proper fix.

Cheers,
Arun