Re: FsAppTransmitter / dummy call in nullprpl
David Woodhouse <[email protected]> Sun, 10 Dec 2017 21:02:24 +0000
| Newsgroups | gmane.comp.telephony.farsight.devel,gmane.comp.gnome.gaim.devel |
|---|---|
| Message-ID | <[email protected]> |
On Mon, 2017-11-06 at 19:40 +0100, Jakub Adam wrote: > Hi David, > > after commenting out > > "wait-for-connection", FALSE, > > in g_object_set() of the file sink (which is a property GstFileSink doesn't have) > Pidgin started saving raw audio into /tmp/pidgin.s16 and I was also able to import > the file to Audacity and play it back correctly. Hm... so it looks like I can't actually get Farstream to do any of the codec work for me. That's in FsRtpConference, not FsRawConference. If I attempt to set the remote codecs to Opus, it isn't going to work. So I probably need the "opusenc ! appsink" and "appsrc ! opusdec" parts to be handled by my new FsTransmitter. It isn't just "FsAppTransmitter". I'm thinking of following the precedent set by FsShmTransmitter and abusing the host and username fields of the FsCandidate, except not with filenames as FsShmTransmitter does it, but with pipeline descriptions to be handled by gst_parse_launch(). That also resolves the issue of communicating the appsink/appsrc between the FsTransmitter and the application (which Pidgin in particular is going to make complex with its abstractions). The application (the PRPL) can just precreate those, then refer to them as named elements in the pipeline descriptions it passes in to the transmitter — something like "opusenc ! chimecallappsink0." and "chimecallappsrc0. ! opusdec" for "host" and "username" in the FsCandidate respectively... Does that seem reasonable? _______________________________________________ Farstream-devel mailing list [email protected] https://lists.freedesktop.org/mailman/listinfo/farstream-devel
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