Use new TransportController implementation in PeerConnection.
The TransportController will be replaced by the JsepTransportController
and JsepTransport will be replace be JsepTransport2.
The JsepTransportController will take the entire SessionDescription
and handle the RtcpMux, Sdes and BUNDLE internally.
The ownership model is also changed. The P2P layer transports are not
ref-counted and will be owned by the JsepTransport2.
In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport
or SrtpTransport and it implements the public and internal interface
by calling the transport underneath.
Bug: webrtc:8587
Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df
Reviewed-on: https://webrtc-review.googlesource.com/59586
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22693}diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 33bf267..6dbf320 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -2342,8 +2342,10 @@
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
- EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track)->
- CaptureStartNtpTime() > 0, 2 * kMaxWaitForFramesMs);
+ EXPECT_TRUE_WAIT(
+ callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
+ 0,
+ 2 * kMaxWaitForFramesMs);
}
// Test that we can get stats (using the new stats implemnetation) for