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webrtc: add a few comments on bundle and src pad exposure

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This commit is contained in:
Matthew Waters 2018-11-26 16:21:19 +11:00
parent 6f91a191de
commit a42fdbb012
1 changed files with 18 additions and 1 deletions
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19
ext/webrtc/gstwebrtcbin.c
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@ -62,7 +62,21 @@
*
* On the receiving side, RTPTransceiver's are created in response to setting
* a remote description. Output pads for the receiving streams in the set
* description are also created.
* description are also created when data is received.
*
* A TransportStream is created when needed in order to transport the data over
* the necessary DTLS/ICE channel to the peer. The exact configuration depends
* on the negotiated SDP's between the peers based on the bundle and rtcp
* configuration. Some cases are outlined below for a simple single
* audio/video/data session:
*
* - max-bundle (requires rtcp-muxing) uses a single transport for all
* media/data transported. Renegotiation involves adding/removing the
* necessary streams to the existing transports.
* - max-compat without rtcp-mux involves two TransportStream per media stream
* to transport the rtp and the rtcp packets and a single TransportStream for
* all data channels. Each stream change involves modifying the associated
* TransportStream/s as necessary.
*/
/*
@ -2908,6 +2922,9 @@ _connect_output_stream (GstWebRTCBin * webrtc,
g_free (pad_name);
gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
/* The webrtcbin src_%u output pads will be created when rtpbin receives
* data on that stream in on_rtpbin_pad_added() */
}
typedef struct