[pulseaudio-discuss] bluetooth headset connection and disconnection status

Tanu Kaskinen tanuk at iki.fi
Thu May 3 09:50:37 UTC 2018

On Wed, 2018-05-02 at 08:34 +0900, Shinnosuke Suzuki wrote:
> Hi,
> > It's true that when the server processes the "list cards" command from
> > pactl, packets to/from parec/pacat are not processed during time, but
> > sending the card information shouldn't take a long time, so it sounds
> > strange that you'd observe audio drop-outs. Does your application
> > perhaps itself stop processing the audio to/from pacat/parec while it's
> > running pactl?
> The problem which I have is audio stops only while pactl is running.
> Audio is gradually lagging behind.I mean audio don't drop-out.
> Audio stream seems to be delayed every time calling pactl.
> My application calls pactl per three seconds.The process time for pactl
>  is not so log time. But process time seems to be accumulated by calling pactl.

You're saying that the audio stops, but that there's no drop-out. Those
two things mean the same thing to me, so it's unclear to me what the
nature of your problem is then. By "drop-out" I mean a situation where
there's a bit of silence inserted to playback, or a bit of audio is
missing from a recording stream.

If you are observing changes in end-to-end latency, then that's a
problem in your application. As you're building a telephony app, I
assume the audio pipeline is such that there are two computers both
running your application, and the two application instances stream
audio over the internet. Audio gets from one computer's microphone to
the speakers of the other computer. Have you considered the fact that
the microphone and the speakers run using different clocks? Some clock
drift is unavoidable. Either the speakers consume audio slightly faster
than the microphone produces it, in which case there will be occasional
underruns, or the speakers consume audio slightly slower than the
microphone produces it, in which case the latency will gradually
increase. You'll need to monitor the latency and try to keep it
constant either by resampling or simply by adding/removing samples.



More information about the pulseaudio-discuss mailing list