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Prefix HAVE_SCTP macro with WEBRTC_.

Browse source 
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}
This commit is contained in:
Mirko Bonadei 2021-01-18 13:24:40 +01:00 committed by Commit Bot
parent 6dcbcead3d
commit 5eb43b4777
11 changed files with 26 additions and 26 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
BUILD.gn
View file

@ -261,7 +261,7 @@ config("common_config") {
}
if (rtc_enable_sctp) {
defines += [ "HAVE_SCTP" ]
defines += [ "WEBRTC_HAVE_SCTP" ]
}
if (rtc_enable_external_auth) {

4
pc/BUILD.gn
View file

@ -19,7 +19,7 @@ group("pc") {
config("rtc_pc_config") {
defines = []
if (rtc_enable_sctp) {
defines += [ "HAVE_SCTP" ]
defines += [ "WEBRTC_HAVE_SCTP" ]
}
}
@ -1031,7 +1031,7 @@ if (rtc_include_tests) {
]
if (rtc_enable_sctp) {
defines = [ "HAVE_SCTP" ]
defines = [ "WEBRTC_HAVE_SCTP" ]
}
deps = [

2
pc/connection_context.cc
View file

@ -63,7 +63,7 @@ std::unique_ptr<SctpTransportFactoryInterface> MaybeCreateSctpFactory(
if (factory) {
return factory;
}
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
return std::make_unique<cricket::SctpTransportFactory>(network_thread);
#else
return nullptr;

2
pc/peer_connection_bundle_unittest.cc
View file

@ -755,7 +755,7 @@ TEST_P(PeerConnectionBundleTest, RejectDescriptionChangingBundleTag) {
// (https://bugs.chromium.org/p/chromium/issues/detail?id=827917)
TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
RTCConfiguration config;
#ifndef HAVE_SCTP
#ifndef WEBRTC_HAVE_SCTP
config.enable_rtp_data_channel = true;
#endif
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;

4
pc/peer_connection_end_to_end_unittest.cc
View file

@ -465,7 +465,7 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
EXPECT_NE(encoder_id1, encoder_id2);
}
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
// Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data.
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
@ -735,7 +735,7 @@ TEST_P(PeerConnectionEndToEndTest, TooManyDataChannelsOpenedBeforeConnecting) {
channels[cricket::kMaxSctpStreams / 2]->state());
}
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
TEST_P(PeerConnectionEndToEndTest, CanRestartIce) {
rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =

6
pc/peer_connection_histogram_unittest.cc
View file

@ -497,7 +497,7 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithMdnsCallee) {
expected_fingerprint_callee));
}
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
@ -521,7 +521,7 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
expected_fingerprint |
static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2);
}
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
#endif // WEBRTC_ANDROID
TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) {
@ -628,7 +628,7 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) {
}
#ifndef WEBRTC_ANDROID
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
// Test that the usage pattern bits for adding remote (private IPv6) candidates
// are set when the remote candidates are retrieved from the Offer SDP instead
// of trickled ICE messages.

14
pc/peer_connection_integrationtest.cc
View file

@ -3705,7 +3705,7 @@ TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
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 @@ TEST_P(PeerConnectionIntegrationTest,
kDefaultTimeout);
}
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
// Test that the ICE connection and gathering states eventually reach
// "complete".
@ -5188,7 +5188,7 @@ TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
caller()->CreateDataChannel();
#endif
caller()->CreateAndSetAndSignalOffer();
@ -5208,7 +5208,7 @@ TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
// 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 @@ TEST_P(PeerConnectionIntegrationTest,
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 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
callee_track->state());
}
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
EndToEndCallWithBundledSctpDataChannel) {
@ -5978,7 +5978,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
}
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
} // namespace
} // namespace webrtc

8
pc/peer_connection_interface_unittest.cc
View file

@ -2255,7 +2255,7 @@ TEST_P(PeerConnectionInterfaceTest, TestRejectRtpDataChannelInAnswer) {
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
}
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
// This tests that SCTP data channels can be rejected in an answer.
TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer)
#else
@ -2310,7 +2310,7 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
cricket::GetFirstVideoContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
content =
cricket::GetFirstDataContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
@ -3593,12 +3593,12 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Test that negotiation can succeed with a data channel only, and with the max
// bundle policy. Previously there was a bug that prevented this.
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) {
#else
TEST_P(PeerConnectionInterfaceTest,
DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) {
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
CreatePeerConnection(config);

4
pc/peer_connection_rtp_unittest.cc
View file

@ -1844,7 +1844,7 @@ TEST_F(PeerConnectionMsidSignalingTest, PureUnifiedPlanToUs) {
class SdpFormatReceivedTest : public PeerConnectionRtpTestUnifiedPlan {};
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
TEST_F(SdpFormatReceivedTest, DataChannelOnlyIsReportedAsNoTracks) {
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->CreateDataChannel("dc");
@ -1856,7 +1856,7 @@ TEST_F(SdpFormatReceivedTest, DataChannelOnlyIsReportedAsNoTracks) {
metrics::Samples("WebRTC.PeerConnection.SdpFormatReceived"),
ElementsAre(Pair(kSdpFormatReceivedNoTracks, 1)));
}
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
TEST_F(SdpFormatReceivedTest, SimpleUnifiedPlanIsReportedAsSimple) {
auto caller = CreatePeerConnectionWithUnifiedPlan();

4
pc/rtc_stats_integrationtest.cc
View file

@ -1091,7 +1091,7 @@ class RTCStatsReportVerifier {
rtc::scoped_refptr<const RTCStatsReport> report_;
};
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) {
StartCall();
@ -1254,7 +1254,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsReferencedIds) {
}
}
}
#endif // HAVE_SCTP
#endif // WEBRTC_HAVE_SCTP
} // namespace

2
pc/test/test_sdp_strings.h
View file

@ -60,7 +60,7 @@ static const char kFireFoxSdpOffer[] =
"a=candidate:4 2 UDP 2113667326 10.0.254.2 58890 typ host\r\n"
"a=candidate:5 2 UDP 1694302206 74.95.2.170 33611 typ srflx raddr"
" 10.0.254.2 rport 58890\r\n"
#ifdef HAVE_SCTP
#ifdef WEBRTC_HAVE_SCTP
"m=application 45536 DTLS/SCTP 5000\r\n"
"c=IN IP4 74.95.2.170\r\n"
"a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n"