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9 commits

Author SHA1 Message Date
Jan Schmidt
8177588250 examples/sendrecv: Remove extra unref of webrtcbin 
The code now constructs webrtcbin with a floating ref and then
gives it to the pipeline. The extra unref is one too many.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3436>
 2022-11-19 19:51:54 +11:00
Jan Schmidt
f2ae481a69 examples/webrtc: Configure payload types 
MR 2398 broke the webrtc sendrecv example
by not configuring the payload types, so both audio and video streams
get sent on payload 96.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
 2022-11-19 13:12:58 +11:00
Matthew Waters
d586c2cc28 examples/webrtc: don't use factory_make_full() for enums 
They are not currently translated into their respective enum values and
will produce an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3210>
 2022-10-18 01:30:37 +00:00
yatinmaan
2c1e61ea16 webrtc: Split WebRTCICE into base classes and implementation. 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
 2022-07-26 13:51:11 +00:00
Nirbheek Chauhan
0007fa38e0 webrtc-sendrecv: Fix create-answer caps negotiation 
We need to parse the payload type map provided by the offer SDP and
set those values on the payloader, otherwise webrtcbin will create
a recvonly answer SDP and we won't send anything to the browser.

Fixed it for both C and Python sendrecv examples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
3c0d582b7c webrtc_sendrecv.py: Add picture-id-mode to rtpvp8pay 
This doesn't just make TWCC stats perform better, it also fixes
stuttery video playback in Chrome.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
583408c312 webrtc_sendrecv.py: Implement all negotiation modes 
Earlier, the example only supported one negotiation mode:
* Browser client is running, gstreamer starts a call and sends offer

Now these three modes are also supported:
* Browser client is running, gstreamer starts a call and sends an
  offer request
* gstreamer connects and waits for browser client to start a call and
  send an offer
* gstreamer connects and waits for browser client to start a call and
  send an offer request

The following features are still missing:
* Data channel support
* TWCC support + stats logging

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
d6799b069a webrtc: Update Makefile for building webrtc-sendrecv 
This now needs the RTP library.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Thibault Saunier
41ed155bdf Move files from gst-examples into the "subprojects/gst-examples/" subdir 2021-09-24 16:15:58 -03:00
Renamed from webrtc/sendrecv/gst/webrtc-sendrecv.c (Browse further)