- Sep 07, 2020
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Tim-Philipp Müller authored
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- Aug 20, 2020
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Tim-Philipp Müller authored
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- Aug 19, 2020
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Matthew Waters authored
Otherwise the app fails to run Part-of: <!25>
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Matthew Waters authored
Fixes possible critical/crash on startup Part-of: <!25>
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Matthew Waters authored
Instead of a generic app-debug.apk Part-of: <!25>
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Matthew Waters authored
As is now required Part-of: <!25>
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Matthew Waters authored
That's what is shipped upstream now. Part-of: <!25>
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Matthew Waters authored
Part-of: <gstreamer/gst-examples!25>
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- Aug 09, 2020
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Carl Karsten authored
Part-of: <!23>
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- Aug 05, 2020
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Sebastian Dröge authored
The default changed back to none because it broke existing code. See gst-plugins-good#749 Part-of: <!22>
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- Jul 31, 2020
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Sebastian Dröge authored
Might miss some signal emissions otherwise, especially the on-negotiation-needed signal. Part-of: <!21>
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Sebastian Dröge authored
Part-of: <!21>
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- Jul 27, 2020
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Seungha Yang authored
g_print* would print broken string on Windows See also gstreamer!258 Part-of: <!20>
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- Jul 03, 2020
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Tim-Philipp Müller authored
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- Jul 02, 2020
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Tim-Philipp Müller authored
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- Jun 29, 2020
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Philippe Normand authored
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: <gstreamer/gst-examples!15>
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- Jun 25, 2020
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Matthew Waters authored
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: <!17>
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Matthew Waters authored
Part-of: <gstreamer/gst-examples!16>
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Matthew Waters authored
- Integrate with the build system. - Some README updates. Part-of: <!16>
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Matthew Waters authored
Original repository location: https://github.com/centricular/gstwebrtc-demos
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- Jun 19, 2020
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Tim-Philipp Müller authored
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Tim-Philipp Müller authored
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- Jun 18, 2020
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We didn't notice this because the logging was broken.
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Reload the SSL context and restart the server if the certificate changes. Without this, new connections will continue to use the old expired certificate.
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First step in making the class able to manage its own state.
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This has changed since the original code was written: https://websockets.readthedocs.io/en/stable/cheatsheet.html#debugging
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It was completely ignored. Also don't de-serialize options. Just parse them directly in `__init__`. Less error-prone.
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- May 14, 2020
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Philippe Normand authored
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- May 12, 2020
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Matthew Waters authored
Part-of: <!14>
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- May 09, 2020
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Jan Schmidt authored
This example sets up a recvonly H.264 transceiver and receives H.264 from a peer, while sending bi-directional Opus audio.
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Jan Schmidt authored
Make sure to unref the transceivers array after use.
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- May 06, 2020
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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.
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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!
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