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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
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2
BUILD.gn
View file

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

4
pc/BUILD.gn
View file

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

2
pc/connection_context.cc
View file

@ -63,7 +63,7 @@ std::unique_ptr<SctpTransportFactoryInterface> MaybeCreateSctpFactory(
if (factory) { if (factory) {
return factory; return factory;
} }
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
return std::make_unique<cricket::SctpTransportFactory>(network_thread); return std::make_unique<cricket::SctpTransportFactory>(network_thread);
#else #else
return nullptr; 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) // (https://bugs.chromium.org/p/chromium/issues/detail?id=827917)
TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) { TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
RTCConfiguration config; RTCConfiguration config;
#ifndef HAVE_SCTP #ifndef WEBRTC_HAVE_SCTP
config.enable_rtp_data_channel = true; config.enable_rtp_data_channel = true;
#endif #endif
config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle; 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); EXPECT_NE(encoder_id1, encoder_id2);
} }
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
// Verifies that a DataChannel created before the negotiation can transition to // Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data. // "OPEN" and transfer data.
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
@ -735,7 +735,7 @@ TEST_P(PeerConnectionEndToEndTest, TooManyDataChannelsOpenedBeforeConnecting) {
channels[cricket::kMaxSctpStreams / 2]->state()); channels[cricket::kMaxSctpStreams / 2]->state());
} }
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
TEST_P(PeerConnectionEndToEndTest, CanRestartIce) { TEST_P(PeerConnectionEndToEndTest, CanRestartIce) {
rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory = 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)); expected_fingerprint_callee));
} }
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) { TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
auto caller = CreatePeerConnection(); auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection(); auto callee = CreatePeerConnection();
@ -521,7 +521,7 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
expected_fingerprint | expected_fingerprint |
static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2); static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2);
} }
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
#endif // WEBRTC_ANDROID #endif // WEBRTC_ANDROID
TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) { TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) {
@ -628,7 +628,7 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) {
} }
#ifndef WEBRTC_ANDROID #ifndef WEBRTC_ANDROID
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
// Test that the usage pattern bits for adding remote (private IPv6) candidates // 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 // are set when the remote candidates are retrieved from the Offer SDP instead
// of trickled ICE messages. // of trickled ICE messages.

14
pc/peer_connection_integrationtest.cc
View file

@ -3705,7 +3705,7 @@ TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
kDefaultTimeout); kDefaultTimeout);
} }
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
// This test sets up a call between two parties with audio, video and an SCTP // This test sets up a call between two parties with audio, video and an SCTP
// data channel. // data channel.
@ -3931,7 +3931,7 @@ TEST_P(PeerConnectionIntegrationTest,
kDefaultTimeout); kDefaultTimeout);
} }
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
// Test that the ICE connection and gathering states eventually reach // Test that the ICE connection and gathering states eventually reach
// "complete". // "complete".
@ -5188,7 +5188,7 @@ TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
ASSERT_TRUE(CreatePeerConnectionWrappers()); ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling(); ConnectFakeSignaling();
caller()->AddAudioVideoTracks(); caller()->AddAudioVideoTracks();
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
caller()->CreateDataChannel(); caller()->CreateDataChannel();
#endif #endif
caller()->CreateAndSetAndSignalOffer(); caller()->CreateAndSetAndSignalOffer();
@ -5208,7 +5208,7 @@ TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
// Test that transport stats are generated by the RTCStatsCollector for a // Test that transport stats are generated by the RTCStatsCollector for a
// connection that only involves data channels. This is a regression test for // connection that only involves data channels. This is a regression test for
// crbug.com/826972. // crbug.com/826972.
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
TEST_P(PeerConnectionIntegrationTest, TEST_P(PeerConnectionIntegrationTest,
TransportStatsReportedForDataChannelOnlyConnection) { TransportStatsReportedForDataChannelOnlyConnection) {
ASSERT_TRUE(CreatePeerConnectionWrappers()); ASSERT_TRUE(CreatePeerConnectionWrappers());
@ -5224,7 +5224,7 @@ TEST_P(PeerConnectionIntegrationTest,
auto callee_report = callee()->NewGetStats(); auto callee_report = callee()->NewGetStats();
EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size()); EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
} }
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
TEST_P(PeerConnectionIntegrationTest, TEST_P(PeerConnectionIntegrationTest,
IceEventsGeneratedAndLoggedInRtcEventLog) { IceEventsGeneratedAndLoggedInRtcEventLog) {
@ -5910,7 +5910,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
callee_track->state()); callee_track->state());
} }
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
TEST_F(PeerConnectionIntegrationTestUnifiedPlan, TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
EndToEndCallWithBundledSctpDataChannel) { EndToEndCallWithBundledSctpDataChannel) {
@ -5978,7 +5978,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout); ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
} }
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
} // namespace } // namespace
} // namespace webrtc } // 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()); 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. // This tests that SCTP data channels can be rejected in an answer.
TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer) TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer)
#else #else
@ -2310,7 +2310,7 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
cricket::GetFirstVideoContent(pc_->local_description()->description()); cricket::GetFirstVideoContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL); ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected); EXPECT_FALSE(content->rejected);
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
content = content =
cricket::GetFirstDataContent(pc_->local_description()->description()); cricket::GetFirstDataContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL); 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 // Test that negotiation can succeed with a data channel only, and with the max
// bundle policy. Previously there was a bug that prevented this. // bundle policy. Previously there was a bug that prevented this.
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) { TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) {
#else #else
TEST_P(PeerConnectionInterfaceTest, TEST_P(PeerConnectionInterfaceTest,
DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) { DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) {
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
PeerConnectionInterface::RTCConfiguration config; PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle; config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
CreatePeerConnection(config); CreatePeerConnection(config);

4
pc/peer_connection_rtp_unittest.cc
View file

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

4
pc/rtc_stats_integrationtest.cc
View file

@ -1091,7 +1091,7 @@ class RTCStatsReportVerifier {
rtc::scoped_refptr<const RTCStatsReport> report_; rtc::scoped_refptr<const RTCStatsReport> report_;
}; };
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) { TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) {
StartCall(); StartCall();
@ -1254,7 +1254,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsReferencedIds) {
} }
} }
} }
#endif // HAVE_SCTP #endif // WEBRTC_HAVE_SCTP
} // namespace } // 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: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" "a=candidate:5 2 UDP 1694302206 74.95.2.170 33611 typ srflx raddr"
" 10.0.254.2 rport 58890\r\n" " 10.0.254.2 rport 58890\r\n"
#ifdef HAVE_SCTP #ifdef WEBRTC_HAVE_SCTP
"m=application 45536 DTLS/SCTP 5000\r\n" "m=application 45536 DTLS/SCTP 5000\r\n"
"c=IN IP4 74.95.2.170\r\n" "c=IN IP4 74.95.2.170\r\n"
"a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n" "a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n"