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

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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.

-- 
Tanu



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