Sebastian Dröge
6233eb0ff3
common: Stop using GQuark-based GstStructure field name API
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432 >
2024-09-26 19:21:29 +03:00
Guillaume Desmottes
ed54734825
examples: set perfect-timestamp=true on opusenc
...
Fix audio streaming on Chrome, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6512 >
2024-04-02 22:08:31 +00:00
Nirbheek Chauhan
033a71e405
webrtc examples: Use webrtc.gstreamer.net
...
Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3802 >
2023-02-04 13:37:02 +00:00
Matthew Waters
b134433e0b
examples/webrtc-sendrecv: add some dot file dumps on async-done and error messages
...
Just as a helpful thing if debugging is needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3823 >
2023-01-30 05:22:59 +00:00
Olivier Crête
b7c0e8bc84
webrtc examples: Force regular non-MULTIOPUS
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Using MULTIOPUS breaks with most browsers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675 >
2023-01-04 12:02:25 +00:00
Sebastian Dröge
d10981f7b9
examples: webrtc: Add bus handling to the Android and C sendrecv examples
...
Without a bus, messages will just pile up and errors are not handled at
all. Also without handling the LATENCY messages the latency configured
on the pipeline will be wrong.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609 >
2022-12-20 13:02:08 +02:00
Jan Schmidt
8177588250
examples/sendrecv: Remove extra unref of webrtcbin
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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
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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
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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
