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Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"

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This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4.

Reason for revert: Breaks WebRTC Chromium FYI Bots

First failure:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925

Failed tests:
WebRtcDataBrowserTest.CallWithSctpDataAndMedia
WebRtcDataBrowserTest.CallWithSctpDataOnly

Original change's description:
> Fix unsynchronized access to mid_to_transport_ in JsepTransportController
>
> * Added several thread checks to JTC to help with programmer errors.
> * Avoid a few Invokes() to the network thread here and there such
>   as for fetching sctp transport name for getStats(). The transport
>   name is now cached when it changes on the network thread.
> * JsepTransportController instances now get deleted on the network
>   thread rather than on the signaling thread + issuing an Invoke()
>   in the dtor.
> * Moved some thread hops from JTC over to PC which is where the problem
>   exists and also (imho) makes it easier to see where hops happen in
>   the PC code.
> * The sctp transport is now started asynchronously when we push down the
>   media description.
> * PeerConnection proxy calls GetSctpTransport directly on the network
>   thread instead of to the signaling thread + blocking on the network
>   thread.
> * The above changes simplified things for webrtc::SctpTransport which
>   allowed for removing locking from that class and delete some code.
>
> Bug: webrtc:9987, webrtc:12445
> Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33191}

TBR=tommi@webrtc.org,hta@webrtc.org

Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9987
Bug: webrtc:12445
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33204}
This commit is contained in:
Guido Urdaneta
2021-02-09 12:21:02 +00:00
committed by Commit Bot
parent 6e4fcac313
commit 6b143c1c06
11 changed files with 141 additions and 237 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
pc/peer_connection_integrationtest.cc
View File

@ -5969,11 +5969,9 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
ASSERT_EQ_WAIT(SctpTransportState::kConnected,
caller()->pc()->GetSctpTransport()->Information().state(),
kDefaultTimeout);
});
ASSERT_EQ_WAIT(SctpTransportState::kConnected,
caller()->pc()->GetSctpTransport()->Information().state(),
kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
}