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Author SHA1 Message Date
Carl Karsten
e1de93cf40 Update README.md 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/23>
 2020-08-09 20:06:54 +00:00
Sebastian Dröge
bbed24d919 webrtc: Change H264 examples to use aggregate-mode=zero-latency for best compatibility 
The default changed back to none because it broke existing code.

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/22>
 2020-08-05 10:47:55 +03:00
Sebastian Dröge
6378337a0e sendrecv/Rust: Only set pipeline to Playing after connecting to the signals 
Might miss some signal emissions otherwise, especially the
on-negotiation-needed signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/21>
 2020-07-31 12:03:46 +03:00
Sebastian Dröge
3492c81fcf Update Rust examples to latest bindings versions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/21>
 2020-07-31 11:59:58 +03:00
Seungha Yang
61d200a957 Port to gst_print* family 
g_print* would print broken string on Windows
See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
 2020-07-27 16:28:33 +09:00
Tim-Philipp Müller
38d6a5873a Back to development 2020-07-03 02:04:21 +01:00
Tim-Philipp Müller
a8510e63d1 Release 1.17.2 2020-07-03 00:37:47 +01:00
Philippe Normand
234dff8dbb webrtc: Add Janus video-room example 
This Rust crate provides a program able to connect to a Janus instance using
WebSockets and send a live video stream to the videoroom plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/15>
 2020-06-29 14:08:51 +01:00
Matthew Waters
f5d9471639 webrtc/test: check if selenium is available before attempting to add tests 
Fixes the following error

File "/builds/vivia/gst-plugins-bad/gst-build/build/../subprojects/gst-examples/webrtc/check/basic.py", line 5, in <module>
     from selenium import webdriver

ModuleNotFoundError: No module named 'selenium'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/17>
 2020-06-25 22:11:33 +10:00
Matthew Waters
204945b902 webrtc: indent sources 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
 2020-06-25 18:36:22 +10:00
Matthew Waters
e1c3dad258 webrtc: update for move to gst-examples 
- Integrate with the build system.
- Some README updates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
 2020-06-25 18:36:22 +10:00
Matthew Waters
a88e90fa9e Move gstwebrtc-demos into gst-examples 
Original repository location: https://github.com/centricular/gstwebrtc-demos
 2020-06-25 18:36:22 +10:00
Nirbheek Chauhan
d44b2316fa sendonly: Don't assume we're building on UNIX 
Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
 2020-06-25 18:36:18 +10:00
Tim-Philipp Müller
01882c92d1 Back to development 2020-06-20 00:28:41 +01:00
Tim-Philipp Müller
5f8bf174e8 Release 1.17.1 2020-06-19 19:28:16 +01:00
Nirbheek Chauhan
751d06af6f signalling: Fix simple-server script name in Dockerfile 
Fixes https://github.com/centricular/gstwebrtc-demos/issues/202
 2020-06-18 23:34:48 +10:00
Corey Cole
17f84bfd81 fix: python webrtc_sendrecv.py typo 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
0776def18c simple_server: asyncio TimeoutError has moved 
We didn't notice this because the logging was broken.
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
77ae10ab66 simple_server: Restart when the certificate changes 
Reload the SSL context and restart the server if the certificate
changes. Without this, new connections will continue to use the old
expired certificate.
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
4761396d87 simple_server: Abstract out ssl context generation 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
7b96b06752 simple_server: Make the server class loop-aware 
First step in making the class able to manage its own state.
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
b8c1bd1fa3 simple_server: Fix init of websockets log handler 
This has changed since the original code was written:
https://websockets.readthedocs.io/en/stable/cheatsheet.html#debugging
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
78df1ca74c simple_server: Correctly pass health option 
It was completely ignored. Also don't de-serialize options. Just parse
them directly in `__init__`. Less error-prone.
 2020-06-18 23:34:48 +10:00
Sebastian Dröge
180e1ce24c Update dependencies of Rust demos 2020-06-18 23:34:48 +10:00
Philippe Normand
c0f303eacf janus: Remove unused parameters and refactor 2020-05-14 11:04:37 +01:00
Matthew Waters
219415dbf6 add vulkan example for android 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/14>
 2020-05-12 16:24:38 +10:00
Jan Schmidt
255fef3896 webrtc-recvonly-h264: Add a recvonly standalone example. 
This example sets up a recvonly H.264 transceiver and receives
H.264 from a peer, while sending bi-directional Opus audio.
 2020-05-09 19:13:52 +10:00
Jan Schmidt
8da8375986 sendonly: Fix transceivers leak. 
Make sure to unref the transceivers array after use.
 2020-05-09 19:13:52 +10:00
Matthew Waters
7445fc4928 signalling/server: python 3.8 asyncio has it's own TimeoutError 2020-05-06 06:01:57 +00:00
Matthew Waters
3a86a37c03 sendrecv: wait until the offer is set before creating answer 
Pragmatically, an answer cannot be created until the offer is created as
the answer creation needs information from the offer.  Practically, due
to implementation details, the answer was always queued after the set of
the offer and so the call flow did not matter.

The current code also hid a bug in webrtcbin where ice candidates would be
generated before the answer had been created which is against the JSEP
specification.

Change to the correct call flow for exemplary effect.
 2020-05-06 06:01:57 +00:00
Matthew Waters
615813ef93 check/validate: a few more tests and improvements 
Tests a matrix of options:
- local/remote negotiation initiator
- 'most' bundle-policy combinations (some combinations will never work)
- firefox or chrome browser

Across 4 test scenarios:
- simple negotiation with default browser streams (or none if gstreamer
  initiates)
- sending a vp8 stream
- opening a data channel
- sending a message over the data channel

for a total of 112 tests!
 2020-05-06 06:01:57 +00:00
Matthew Waters
c3f629340d check: first pass at a couple of validate tests 2020-05-06 06:01:57 +00:00
Matthew Waters
bc821a85d4 tests: first pass at some basic browser tests 2020-05-06 06:01:57 +00:00
Matthew Waters
37cf0dffb5 add __pycache__ to .gitignore 2020-05-06 06:01:57 +00:00
Costa Shulyupin
56a03add78 html: charset 
Avoid warning:
The character encoding of the HTML document was not declared.
The document will render with garbled text in some browser configurations
if the document contains characters from outside the US-ASCII range.
The character encoding of the page must be declared in the document
or in the transfer protocol.
 2020-04-16 17:53:17 +02:00
Costa Shulyupin
8c4345da7d android, mp-webrtc-sendrecv, sendonly: cleanup 
webrtc-unidirectional-h264.c: removed empty lines

android: removed unused var
 2020-04-16 17:34:11 +02:00
Costa Shulyupin
133a1593ee android, sendrecv: add missing break in switch case statements 2020-04-16 17:34:11 +02:00
Costa Shulyupin
2557eab9d5 gst-indent 2020-04-14 14:40:37 +03:00
Costa Shulyupin
ca96b6de86 gst-indent 2020-04-14 14:40:37 +03:00
Costa Shulyupin
804c0c2f5e gst-indent 2020-04-14 14:40:37 +03:00
Sebastian Dröge
65db695212 Set TURN server in Rust sendrecv example too 
Previously it was only in the multiparty example.
 2020-03-24 12:57:17 +02:00
Jan Schmidt
5bf67feae8 sendrecv: Add a switch for remote-offerer 
Add a switch to the command line utility that makes it request
the initial offer from the peer instead of generating it.

Modify the webrtc.js example to support a new REQUEST_OFFER
message, and generate the offer when receiving it.
 2020-03-05 03:03:17 +11:00
Nirbheek Chauhan
8b90dd949a Cerbero has moved from gnutls+openssl to only openssl 2020-03-03 08:03:31 +05:30
Jan Schmidt
c8e79c9671 webrtc-sendrecv.py: Add a stun server 
Fixes https://github.com/centricular/gstwebrtc-demos/issues/160
 2020-02-21 14:01:58 +11:00
Jan Schmidt
d2236266dc Android: Update build for android example 2020-02-07 23:00:20 +11:00
Sebastian Dröge
699b830213 Update Rust examples to async-tungstenite 0.4 2020-02-01 15:21:08 +02:00
Jan Schmidt
1f1233064f janus: Add picture-id-mode=2 to VP8 payloading 
This writes an extended header and Picture-ID into each RTP packet
which makes Janus able to detect which frames are keyframes and
to request replacement keyframes.
 2020-01-28 00:05:59 +11:00
Jan Schmidt
d8e7687132 janus: Add options near the top 
Add some script configuration options to choose
between VP8 and H.264 near the top, to modify the video input
source, and to enable/disable RTX support
 2020-01-28 00:05:59 +11:00
Sebastian Dröge
42c6eac7f1 Update dependencies of Rust examples and simplify slightly 2020-01-23 08:36:21 +02:00
Jan Schmidt
3cabee61c7 Add python Janus videoroom streaming example. 
Added with permission and copyright @tobiasfriden and @saket424
on github. See https://github.com/centricular/gstwebrtc-demos/issues/66
 2020-01-15 10:47:27 +11:00