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https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
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31e836f4d6
This makes the new payloaders closer to the old ones, and makes usage in webrtcbin easier. Also properly configure default PT of subclasses. Previously any PT that was set for these subclasses via g_object_new() would be overridden by the default one during construction. Additionally, do SSRC collision handling while queueing output packets. This is the more natural place as that's where the SSRC is actually used, it happens potentially earlier and also allows to drain any pending packets before the SSRC change in the caps. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/557 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693> |
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| .. | ||
| aws | ||
| hlssink3 | ||
| mpegtslive | ||
| ndi | ||
| onvif | ||
| quinn | ||
| raptorq | ||
| reqwest | ||
| rtp | ||
| rtsp | ||
| webrtc | ||
| webrtchttp | ||