commit 5eb43b4777cfbf02833e552d6d367cf50d39d97e
author Mirko Bonadei <mbonadei@webrtc.org> Mon Jan 18 13:24:40 2021 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Jan 20 10:51:07 2021 +0000
tree 1abb040d68e14fae4a8428af89a7f486b03ac9eb
parent 6dcbcead3da45604b84aef9d9496689fef5ff8f5

Prefix HAVE_SCTP macro with WEBRTC_.

Generated automatically with:

  git grep -l "\bHAVE_SCTP\b" | xargs \
    sed -i '' 's/HAVE_SCTP/WEBRTC_HAVE_SCTP/g'

Bug: webrtc:11142
Change-Id: I30e16a40ca7a7e388940191df22b705265b42cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202251
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33042}

pc/peer_connection_integrationtest.cc

diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index a784126..0005552 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -3705,7 +3705,7 @@
                  kDefaultTimeout);
 }
 
-#ifdef HAVE_SCTP
+#ifdef WEBRTC_HAVE_SCTP
 
 // This test sets up a call between two parties with audio, video and an SCTP
 // data channel.
@@ -3931,7 +3931,7 @@
                  kDefaultTimeout);
 }
 
-#endif  // HAVE_SCTP
+#endif  // WEBRTC_HAVE_SCTP
 
 // Test that the ICE connection and gathering states eventually reach
 // "complete".
@@ -5188,7 +5188,7 @@
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioVideoTracks();
-#ifdef HAVE_SCTP
+#ifdef WEBRTC_HAVE_SCTP
   caller()->CreateDataChannel();
 #endif
   caller()->CreateAndSetAndSignalOffer();
@@ -5208,7 +5208,7 @@
 // Test that transport stats are generated by the RTCStatsCollector for a
 // connection that only involves data channels. This is a regression test for
 // crbug.com/826972.
-#ifdef HAVE_SCTP
+#ifdef WEBRTC_HAVE_SCTP
 TEST_P(PeerConnectionIntegrationTest,
        TransportStatsReportedForDataChannelOnlyConnection) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
@@ -5224,7 +5224,7 @@
   auto callee_report = callee()->NewGetStats();
   EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
 }
-#endif  // HAVE_SCTP
+#endif  // WEBRTC_HAVE_SCTP
 
 TEST_P(PeerConnectionIntegrationTest,
        IceEventsGeneratedAndLoggedInRtcEventLog) {
@@ -5910,7 +5910,7 @@
             callee_track->state());
 }
 
-#ifdef HAVE_SCTP
+#ifdef WEBRTC_HAVE_SCTP
 
 TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
        EndToEndCallWithBundledSctpDataChannel) {
@@ -5978,7 +5978,7 @@
   ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
 }
 
-#endif  // HAVE_SCTP
+#endif  // WEBRTC_HAVE_SCTP
 
 }  // namespace
 }  // namespace webrtc