Chromium Code Reviews
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Chromium Code Reviews
Issue
1371623003:
Delete a connection only if it has timed out on writing and not receiving for 10 seconds. (Closed)
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Created:
5 years, 2 months ago by honghaiz3 Modified:
5 years, 2 months ago Reviewers:
pthatcher1 CC:
webrtc-reviews_webrtc.org, tterriberry_mozilla.com Base URL:
https://chromium.googlesource.com/external/webrtc@master Target Ref:
refs/pending/heads/master Project:
webrtc Visibility:
Public. |
DescriptionDelete a connection only if it has timed out on writing and not receiving for 10 seconds.
BUG=webrtc:5034, webrtc:4937
R=pthatcher@webrtc.org
Committed: https://crrev.com/2b342bf99c9578247940929c02a41ef9ccec6d6e
Cr-Commit-Position: refs/heads/master@{#10119}
Patch Set 1 : #Patch Set 2 : A minor change in GetStats #
Total comments: 35
Patch Set 3 : Address comments #
Total comments: 2
Patch Set 4 : #
Total comments: 2
Patch Set 5 : Addressed comments, fixed Connection Prune method, and added a test #Patch Set 6 : Rename a test #
Messages
Total messages: 36 (16 generated)
Patchset #1 (id:1) has been deleted
Patchset #1 (id:20001) has been deleted
Patchset #1 (id:40001) has been deleted
honghaiz@webrtc.org changed reviewers: + pthatcher@webrtc.org
https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... File webrtc/p2p/base/p2ptransportchannel.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:289: // Currently a channel is considered ICE completed once there is at least one Just remove "Currently" and say "A channel ..." https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:291: // such connection per Network. This would be more clear as "once there is at least one active connection and at most one active connection per network". https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:293: bool has_non_write_timeout_connections = false; I think this would be much more clear as: bool Connection::Active() { return write_state() != WRITE_STATE_TIMEOUT; } if (!had_connection_) { return TransportChannelState::STATE_INIT; } std::vector<Connection*> active_connections; for (Connection connection : connections_) { if (connection->Active()) { active_connections.push_back(connection); } } if (active_connections.empty()) { return TransportChannelState::STATE_FAILED; } for (Connection* connection : active_connections) { // ... rest of the method as is https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:875: !connection->receiving()) { This is basically saying "Don't report stats for connections that we keep around only because they haven't had a minimum lifetime". Why wouldn't we keep around stats for such connections? https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... File webrtc/p2p/base/p2ptransportchannel_unittest.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2049: TEST_F(P2PTransportChannelPingTest, TestDontDeleteRightAwayWhenPrune) { ...WhenPrune => ...WhenPruned https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2069: EXPECT_FALSE(conn2 == nullptr); I think that what we really want to test is that when conn1 is destroyed, then conn2 will begin pinging and can become writable and receiving. Can we make it more complete like that? https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2085: // Now there are two connections, so the transport channel is connecting. We need to check STATE_INIT before there are any connections. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2090: EXPECT_EQ(cricket::TransportChannelState::STATE_COMPLETED, ch.GetState()); To be more complete, I think we need to test the multi-network part of as well. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2092: EXPECT_EQ(cricket::TransportChannelState::STATE_FAILED, ch.GetState()); We should also test that not only Prune() get us in this state, but so does a lose-writability event. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc File webrtc/p2p/base/port.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1233: set_write_state(STATE_WRITE_TIMEOUT); Should just Destroy here? We probably don't want to delay if we can an explicit STUN error repsonse. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1254: void Connection::CheckTimeout(uint32 now) { If this is now called in only one place, why not remove this method and simply call it in that one place? https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1259: // use the connection creation time as the base to compute the timeout. I think this comment ought to say something more like "Don't destroy a connection until N seconds have passed since it was created. We do this to prevent connections from being pruned too quickly during a network change event when two networks would be up simultaneously but only for a brief period." https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1260: uint32 base_for_timeout = std::max(time_created_ms_, last_received()); I think call this "receiving_deadline" would be a bit better: uint32 receiving_deadline = std::max(time_created_ms_, last_received()) + DEAD_CONNECTION_RECEIVE_TIMEOUT. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1264: } This would be more clear with early returns. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1265: } I think this would be more clear with a helper method called something like "Dead": I also think we could combine all my other comments into something like this: void Connection::UpdateState(uint32 now) { ... if (Dead(now)) { Destroy(); } } void Connection::Active() { // TODO: Move from using the write_state_ to using pruned_. return write_state_ != STATE_WRITE_TIMEOUT; } bool Connection::Dead(uint32 now) { if (now < (time_created_ms_ + MIN_CONNECTION_LIFETIME)) { // A connection that hasn't passed its minimum lifetime is still alive. // We do this to prevent connections from being pruned too quickly // during a network change event when two networks would be up // simultaneously but only for a brief period. return false; } if (receiving_) { // A connection that is receiving is alive. return false; } // A connection is alive until it is inactive. return !Active(); // TODO: Move from using the write_state_ to using the receiving state with something like the following: // return (now > (last_received() + DEAD_CONNECTION_RECEIVE_TIMEOUT)); } https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.h File webrtc/p2p/base/port.h (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.h#ne... webrtc/p2p/base/port.h:55: const uint32 DEAD_CONNECTION_RECEIVE_TIMEOUT = 10 * 1000; // 10 seconds. I think there are actually two different things here, that might want different values: 1. "MIN_CONNECTION_LIFETIME", which is the minimum time we will wait before destroying a connection. 10s. 2. "DEAD_CONNECTION_RECEIVE_TIMEOUT", which is the time after which we know we can delete a connection if it hasn't received anything. 30s. Actually, I think we only want #1 for this CL. I don't think we need #2 in this CL.
Thanks! PTAL. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... File webrtc/p2p/base/p2ptransportchannel.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:289: // Currently a channel is considered ICE completed once there is at least one On 2015/09/28 23:07:27, pthatcher1 wrote: > Just remove "Currently" and say "A channel ..." Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:291: // such connection per Network. On 2015/09/28 23:07:27, pthatcher1 wrote: > This would be more clear as "once there is at least one active connection and at > most one active connection per network". Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:293: bool has_non_write_timeout_connections = false; On 2015/09/28 23:07:27, pthatcher1 wrote: > I think this would be much more clear as: > > bool Connection::Active() { > return write_state() != WRITE_STATE_TIMEOUT; > } > > > if (!had_connection_) { > return TransportChannelState::STATE_INIT; > } > > std::vector<Connection*> active_connections; > for (Connection connection : connections_) { > if (connection->Active()) { > active_connections.push_back(connection); > } > } > if (active_connections.empty()) { > return TransportChannelState::STATE_FAILED; > } > > for (Connection* connection : active_connections) { > // ... rest of the method as is Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel.cc:875: !connection->receiving()) { On 2015/09/28 23:07:27, pthatcher1 wrote: > This is basically saying "Don't report stats for connections that we keep around > only because they haven't had a minimum lifetime". Why wouldn't we keep around > stats for such connections? I wanted to keep the same app behavior, but think it is OK to keep them around. I think it will be dropped anyway later in webrtcsession, though. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... File webrtc/p2p/base/p2ptransportchannel_unittest.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2049: TEST_F(P2PTransportChannelPingTest, TestDontDeleteRightAwayWhenPrune) { On 2015/09/28 23:07:27, pthatcher1 wrote: > ...WhenPrune => ...WhenPruned Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2069: EXPECT_FALSE(conn2 == nullptr); On 2015/09/28 23:07:27, pthatcher1 wrote: > I think that what we really want to test is that when conn1 is destroyed, then > conn2 will begin pinging and can become writable and receiving. Can we make it > more complete like that? Done. conn2 will start to ping once conn1 is not receiving. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2085: // Now there are two connections, so the transport channel is connecting. On 2015/09/28 23:07:27, pthatcher1 wrote: > We need to check STATE_INIT before there are any connections. Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2090: EXPECT_EQ(cricket::TransportChannelState::STATE_COMPLETED, ch.GetState()); On 2015/09/28 23:07:27, pthatcher1 wrote: > To be more complete, I think we need to test the multi-network part of as well. Added GetState tests in the MultihomedTest. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2092: EXPECT_EQ(cricket::TransportChannelState::STATE_FAILED, ch.GetState()); On 2015/09/28 23:07:27, pthatcher1 wrote: > We should also test that not only Prune() get us in this state, but so does a > lose-writability event. Prune is used to emulate losing writability. If we want to do an actual lose-writability event, we will need one of the following 1. Wait for 15 seconds until write timeout. That's pretty long for testing. 2. Change the p2ptransportchannel so that we can change the write timeout values. Currently it hard-coded with a constant. 3. Call set_write_state to set to write_timeout. But this is pretty similar to call Prune. Do you want to proceed along one of these? https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc File webrtc/p2p/base/port.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1254: void Connection::CheckTimeout(uint32 now) { On 2015/09/28 23:07:27, pthatcher1 wrote: > If this is now called in only one place, why not remove this method and simply > call it in that one place? Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1259: // use the connection creation time as the base to compute the timeout. On 2015/09/28 23:07:27, pthatcher1 wrote: > I think this comment ought to say something more like "Don't destroy a > connection until N seconds have passed since it was created. We do this to > prevent connections from being pruned too quickly during a network change event > when two networks would be up simultaneously but only for a brief period." Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1260: uint32 base_for_timeout = std::max(time_created_ms_, last_received()); On 2015/09/28 23:07:27, pthatcher1 wrote: > I think call this "receiving_deadline" would be a bit better: > > uint32 receiving_deadline = std::max(time_created_ms_, last_received()) + > DEAD_CONNECTION_RECEIVE_TIMEOUT. Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1264: } On 2015/09/28 23:07:27, pthatcher1 wrote: > This would be more clear with early returns. Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1265: } On 2015/09/28 23:07:27, pthatcher1 wrote: > I think this would be more clear with a helper method called something like > "Dead": I also think we could combine all my other comments into something like > this: > > > void Connection::UpdateState(uint32 now) { > ... > if (Dead(now)) { > Destroy(); > } > } > > void Connection::Active() { > // TODO: Move from using the write_state_ to using pruned_. > return write_state_ != STATE_WRITE_TIMEOUT; > } > > bool Connection::Dead(uint32 now) { > if (now < (time_created_ms_ + MIN_CONNECTION_LIFETIME)) { > // A connection that hasn't passed its minimum lifetime is still alive. > // We do this to prevent connections from being pruned too quickly > // during a network change event when two networks would be up > // simultaneously but only for a brief period. > return false; > } > > if (receiving_) { > // A connection that is receiving is alive. > return false; > } > > // A connection is alive until it is inactive. > return !Active(); > > // TODO: Move from using the write_state_ to using the receiving state with > something like the following: > // return (now > (last_received() + DEAD_CONNECTION_RECEIVE_TIMEOUT)); > } > > > Done. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.h File webrtc/p2p/base/port.h (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.h#ne... webrtc/p2p/base/port.h:55: const uint32 DEAD_CONNECTION_RECEIVE_TIMEOUT = 10 * 1000; // 10 seconds. On 2015/09/28 23:07:28, pthatcher1 wrote: > I think there are actually two different things here, that might want different > values: > > 1. "MIN_CONNECTION_LIFETIME", which is the minimum time we will wait before > destroying a connection. 10s. > > 2. "DEAD_CONNECTION_RECEIVE_TIMEOUT", which is the time after which we know we > can delete a connection if it hasn't received anything. 30s. > > > Actually, I think we only want #1 for this CL. I don't think we need #2 in this > CL. Done.
https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... File webrtc/p2p/base/p2ptransportchannel_unittest.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2092: EXPECT_EQ(cricket::TransportChannelState::STATE_FAILED, ch.GetState()); On 2015/09/29 18:47:49, honghaiz3 wrote: > On 2015/09/28 23:07:27, pthatcher1 wrote: > > We should also test that not only Prune() get us in this state, but so does a > > lose-writability event. > Prune is used to emulate losing writability. > If we want to do an actual lose-writability event, we will need one of the > following > 1. Wait for 15 seconds until write timeout. That's pretty long for testing. > 2. Change the p2ptransportchannel so that we can change the write timeout > values. Currently it hard-coded with a constant. > 3. Call set_write_state to set to write_timeout. But this is pretty similar to > call Prune. > Do you want to proceed along one of these? Is there a way we can just set_write_state(WRITE_TIMEOUT)? https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc File webrtc/p2p/base/port.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1233: set_write_state(STATE_WRITE_TIMEOUT); On 2015/09/28 23:07:27, pthatcher1 wrote: > Should just Destroy here? We probably don't want to delay if we can an explicit > STUN error repsonse. ? https://codereview.webrtc.org/1371623003/diff/90001/webrtc/p2p/base/port.h File webrtc/p2p/base/port.h (right): https://codereview.webrtc.org/1371623003/diff/90001/webrtc/p2p/base/port.h#ne... webrtc/p2p/base/port.h:444: bool Dead(uint32 now) const; Now that I see all of these, I realize they should be lower case: weak(), active(), dead(), to match writable(), receiving(), connected().
PTAL. https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... File webrtc/p2p/base/p2ptransportchannel_unittest.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/p2ptransp... webrtc/p2p/base/p2ptransportchannel_unittest.cc:2092: EXPECT_EQ(cricket::TransportChannelState::STATE_FAILED, ch.GetState()); On 2015/09/29 19:51:32, pthatcher1 wrote: > On 2015/09/29 18:47:49, honghaiz3 wrote: > > On 2015/09/28 23:07:27, pthatcher1 wrote: > > > We should also test that not only Prune() get us in this state, but so does > a > > > lose-writability event. > > Prune is used to emulate losing writability. > > If we want to do an actual lose-writability event, we will need one of the > > following > > 1. Wait for 15 seconds until write timeout. That's pretty long for testing. > > 2. Change the p2ptransportchannel so that we can change the write timeout > > values. Currently it hard-coded with a constant. > > 3. Call set_write_state to set to write_timeout. But this is pretty similar to > > call Prune. > > Do you want to proceed along one of these? > > Is there a way we can just set_write_state(WRITE_TIMEOUT)? Yes although set_write_state() is protected. We can change it to public and add comments saying it is for testing. But What Prune does is really just call set_write_state(WRITE_TIMEOUT) (plus a few other things). Maybe we just revisit this when changing the behavior of Prune? https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc File webrtc/p2p/base/port.cc (right): https://codereview.webrtc.org/1371623003/diff/80001/webrtc/p2p/base/port.cc#n... webrtc/p2p/base/port.cc:1233: set_write_state(STATE_WRITE_TIMEOUT); On 2015/09/28 23:07:27, pthatcher1 wrote: > Should just Destroy here? We probably don't want to delay if we can an explicit > STUN error repsonse. Ah. can destroy here. https://codereview.webrtc.org/1371623003/diff/90001/webrtc/p2p/base/port.h File webrtc/p2p/base/port.h (right): https://codereview.webrtc.org/1371623003/diff/90001/webrtc/p2p/base/port.h#ne... webrtc/p2p/base/port.h:444: bool Dead(uint32 now) const; On 2015/09/29 19:51:32, pthatcher1 wrote: > Now that I see all of these, I realize they should be lower case: weak(), > active(), dead(), to match writable(), receiving(), connected(). Done. Also changed Weak to weak in p2ptransportchannel.
https://codereview.webrtc.org/1371623003/diff/110001/webrtc/p2p/base/port.cc File webrtc/p2p/base/port.cc (right): https://codereview.webrtc.org/1371623003/diff/110001/webrtc/p2p/base/port.cc#... webrtc/p2p/base/port.cc:1256: Destroy(); Don't we still need the call to set_state(STATE_FAILED)?
lgtm, assuming you fix the STATE_FAILED thing
https://codereview.webrtc.org/1371623003/diff/110001/webrtc/p2p/base/port.cc File webrtc/p2p/base/port.cc (right): https://codereview.webrtc.org/1371623003/diff/110001/webrtc/p2p/base/port.cc#... webrtc/p2p/base/port.cc:1256: Destroy(); On 2015/09/29 22:56:52, pthatcher1 wrote: > Don't we still need the call to set_state(STATE_FAILED)? Done.
The CQ bit was checked by honghaiz@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1371623003/130001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1371623003/130001
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: Try jobs failed on following builders: win_compile_dbg on tryserver.webrtc (JOB_FAILED, http://build.chromium.org/p/tryserver.webrtc/builders/win_compile_dbg/builds/205)
The CQ bit was checked by honghaiz@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1371623003/150001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1371623003/150001
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: Try jobs failed on following builders: win_x64_rel on tryserver.webrtc (JOB_FAILED, http://build.chromium.org/p/tryserver.webrtc/builders/win_x64_rel/builds/9696)
The CQ bit was checked by honghaiz@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1371623003/170001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1371623003/170001
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: This issue passed the CQ dry run.
Patchset #5 (id:130001) has been deleted
Patchset #5 (id:150001) has been deleted
Patchset #5 (id:170001) has been deleted
The CQ bit was checked by honghaiz@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1371623003/190001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1371623003/190001
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: This issue passed the CQ dry run.
Made a change to the Connection Prune method in order to fix a CQ test error and added a test to verify a connection can be pruned again.
The CQ bit was checked by honghaiz@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1371623003/210001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1371623003/210001
Message was sent while issue was closed.
Committed patchset #6 (id:210001) manually as 2b342bf99c9578247940929c02a41ef9ccec6d6e (presubmit successful).
Message was sent while issue was closed.
Patchset 6 (id:??) landed as https://crrev.com/2b342bf99c9578247940929c02a41ef9ccec6d6e Cr-Commit-Position: refs/heads/master@{#10119} |
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