Add RtpSenderInterface.SetStreams
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index e0a29a3..e6140db 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -61,6 +61,11 @@
// tracks.
virtual std::vector<std::string> stream_ids() const = 0;
+ // Sets the IDs of the media streams associated with this sender's track.
+ // These are signalled in the SDP so that the remote side can associate
+ // tracks.
+ virtual void SetStreams(const std::vector<std::string>& stream_ids) {}
+
// Returns the list of encoding parameters that will be applied when the SDP
// local description is set. These initial encoding parameters can be set by
// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
@@ -112,6 +117,7 @@
rtc::scoped_refptr<FrameEncryptorInterface>)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>,
GetFrameEncryptor)
+PROXY_METHOD1(void, SetStreams, const std::vector<std::string>&)
END_PROXY_MAP()
} // namespace webrtc