Re: Audio src blocked before session set up
David Woodhouse <[email protected]> Thu, 19 Apr 2018 08:33:40 +0100
| Newsgroups | gmane.comp.telephony.farsight.devel,gmane.comp.gnome.gaim.devel |
|---|---|
| Message-ID | <[email protected]> |
On Thu, 2018-04-19 at 07:04 +0000, Joakim Tjernlund wrote: > On Wed, 2018-04-18 at 22:18 +0100, David Woodhouse wrote: > > I've suddenly started experiencing a problem where my outbound audio in > > Pidgin is missing — Farstream never seems to send any audio, and I > > don't even get the farstream-send-codec-changed signal. > > > > It's sporadic, and relatively hard to reproduce. And of course the more > > I turn up the GST_DEBUG debugging, the harder it gets. > > > > It looks like the problem occurs when pulsesrc gets blocked: > I have somewhat similar issue in sipe, but input instead. If I change to ALSA and > do a test call, then the microphone locks up just as the call connects. > If I recall correctly, USB headsets are needed as well. > Using Pulse it usually works. Yeah, that's the same direction — my microphone locks up. I'm also using a USB headset; perhaps that just affects the latency between Pidgin setting it to PLAYING, and the first sample arriving? If the first sample arrives *before* the rest of the call setup, it seems to be OK. It's failing if the sample arrives *during* the call setup, especially (I think) before _send_src_pad_blocked_callback() is registered by fs_rtp_session_verify_send_codec_bin_locked(). Should fs_rtp_session_verify_send_codec_bin_locked() actually be checking if the pad was *already* blocked, and setting up the codecbin immediately if so? _______________________________________________ Farstream-devel mailing list [email protected] https://lists.freedesktop.org/mailman/listinfo/farstream-devel
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