Add support for the live source for WebRTC 36/238536/7
authorSangchul Lee <sc11.lee@samsung.com>
Wed, 15 Jul 2020 01:29:53 +0000 (10:29 +0900)
committerSangchul Lee <sc11.lee@samsung.com>
Wed, 15 Jul 2020 08:36:18 +0000 (17:36 +0900)
commit3007948f9ce37f3ede7e4014dbc37acb1067f3b3
tree989d7404d33384ad9ae056470fe291936da527e2
parent952f84138452c008ef8257dd1195093d5c59a91f
Add support for the live source for WebRTC

Media streamer provides several APIs to change its states
such as prepare(), play() and pause().
During prepare(), it is required for WebRTC node to exchange
negotiation messages via signaling server controlled by
application. But the implimentation of webrtcbin invokes the
'on-negotiation-needed' callback as soon as the first source
buffer arrives. This patch enables source plugins which do not
have the preroll buffer to work well with WebRTC node.

[Version] 0.1.95
[Issue Type] Improvement

Change-Id: I905351c0fd1a422d8e0c64d79995408c5720ea2e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
include/media_streamer_gst.h
include/media_streamer_node.h
include/media_streamer_priv.h
packaging/capi-media-streamer.spec
src/media_streamer_gst.c
src/media_streamer_gst_webrtc.c
src/media_streamer_node.c
src/media_streamer_priv.c