[pulseaudio-discuss] bluetooth headset connection and disconnection status
suzukisn at gmail.com
Fri May 4 12:13:10 UTC 2018
>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.
I'm sorry for I don't make it clear enough.
I heard the bit of silence from speaker of the bluetooth headset periodically.
So there's a bit of silence inserted to playback stream using pacat.
Since the pactl thread is occupied in some milli seconds,
pacat is not running during that time.
So I think pacat put some silence packet inserted to the playback stream in it.
2018-05-03 18:50 GMT+09:00 Tanu Kaskinen <tanuk at iki.fi>:
> 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.
E-mail : suzukisn at gmail.com
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