Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ )
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.
Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: https://chromium.googlesource.com/external/webrtc/+/eaa826c2ee0668cfb4a0dfb66f8d388b65da20f5
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc
index c8a1eb6..c27e53f 100644
--- a/webrtc/api/peerconnection_unittest.cc
+++ b/webrtc/api/peerconnection_unittest.cc
@@ -46,6 +46,12 @@
#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/pc/mediasession.h"
+#define MAYBE_SKIP_TEST(feature) \
+ if (!(feature())) { \
+ LOG(LS_INFO) << "Feature disabled... skipping"; \
+ return; \
+ }
+
using cricket::ContentInfo;
using cricket::FakeWebRtcVideoDecoder;
using cricket::FakeWebRtcVideoDecoderFactory;
@@ -217,7 +223,8 @@
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
- new FakeRTCCertificateGenerator());
+ rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
+ new FakeRTCCertificateGenerator() : nullptr);
return CreateClientWithDtlsIdentityStore(id, constraints, options, config,
std::move(cert_generator), true,
@@ -230,7 +237,8 @@
rtc::Thread* network_thread,
rtc::Thread* worker_thread) {
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
- new FakeRTCCertificateGenerator());
+ rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
+ new FakeRTCCertificateGenerator() : nullptr);
return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr,
std::move(cert_generator), false,
@@ -1464,6 +1472,7 @@
}
void SetupAndVerifyDtlsCall() {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -1488,7 +1497,8 @@
rtc_config.set_cpu_adaptation(false);
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
- new FakeRTCCertificateGenerator());
+ rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
+ new FakeRTCCertificateGenerator() : nullptr);
cert_generator->use_alternate_key();
// Make sure the new client is using a different certificate.
@@ -1684,6 +1694,7 @@
// This test sets up a audio call initially and then upgrades to audio/video,
// using DTLS.
TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -1697,6 +1708,7 @@
// This test sets up a call transfer to a new caller with a different DTLS
// fingerprint.
TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetupAndVerifyDtlsCall();
// Keeping the original peer around which will still send packets to the
@@ -1715,6 +1727,7 @@
// bundle is in effect in the restart, the channel can successfully reset its
// DTLS-SRTP context.
TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -1733,6 +1746,7 @@
// This test sets up a call transfer to a new callee with a different DTLS
// fingerprint.
TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetupAndVerifyDtlsCall();
// Keeping the original peer around which will still send packets to the
@@ -1766,6 +1780,7 @@
// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
// negotiated and used for transport.
TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
@@ -2242,6 +2257,7 @@
// negotiation is completed without error.
#ifdef HAVE_SCTP
TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.SetMandatory(
MediaConstraintsInterface::kEnableDtlsSrtp, true);
diff --git a/webrtc/api/peerconnectionendtoend_unittest.cc b/webrtc/api/peerconnectionendtoend_unittest.cc
index 436e7bf..4110db0 100644
--- a/webrtc/api/peerconnectionendtoend_unittest.cc
+++ b/webrtc/api/peerconnectionendtoend_unittest.cc
@@ -24,6 +24,12 @@
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
+#define MAYBE_SKIP_TEST(feature) \
+ if (!(feature())) { \
+ LOG(LS_INFO) << "Feature disabled... skipping"; \
+ return; \
+ }
+
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
@@ -192,6 +198,8 @@
// Verifies that a DataChannel created before the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
CreatePcs();
webrtc::DataChannelInit init;
@@ -216,6 +224,8 @@
// Verifies that a DataChannel created after the negotiation can transition to
// "OPEN" and transfer data.
TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
CreatePcs();
webrtc::DataChannelInit init;
@@ -247,6 +257,8 @@
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
CreatePcs();
webrtc::DataChannelInit init;
@@ -274,6 +286,8 @@
// there are multiple DataChannels.
TEST_F(PeerConnectionEndToEndTest,
MessageTransferBetweenTwoPairsOfDataChannels) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
CreatePcs();
webrtc::DataChannelInit init;
@@ -395,6 +409,8 @@
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
TEST_F(PeerConnectionEndToEndTest,
DISABLED_DataChannelFromOpenWorksAfterClose) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
CreatePcs();
webrtc::DataChannelInit init;
@@ -421,6 +437,8 @@
// reference count), no memory access violation will occur.
// See: https://code.google.com/p/chromium/issues/detail?id=565048
TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
CreatePcs();
webrtc::DataChannelInit init;
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index e70cbb6..2da3755 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -293,6 +293,12 @@
"a=ssrc:4 cname:stream1\r\n"
"a=ssrc:4 msid:stream1 videotrack1\r\n";
+#define MAYBE_SKIP_TEST(feature) \
+ if (!(feature())) { \
+ LOG(LS_INFO) << "Feature disabled... skipping"; \
+ return; \
+ }
+
using ::testing::Exactly;
using cricket::StreamParams;
using webrtc::AudioSourceInterface;
@@ -2036,6 +2042,7 @@
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
diff --git a/webrtc/api/test/peerconnectiontestwrapper.cc b/webrtc/api/test/peerconnectiontestwrapper.cc
index 3254480..b1eb586 100644
--- a/webrtc/api/test/peerconnectiontestwrapper.cc
+++ b/webrtc/api/test/peerconnectiontestwrapper.cc
@@ -76,7 +76,8 @@
}
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
- new FakeRTCCertificateGenerator());
+ rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeRTCCertificateGenerator()
+ : nullptr);
peer_connection_ = peer_connection_factory_->CreatePeerConnection(
config, constraints, std::move(port_allocator), std::move(cert_generator),
this);
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index c8ae058..1a7a026 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -50,6 +50,12 @@
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediasession.h"
+#define MAYBE_SKIP_TEST(feature) \
+ if (!(feature())) { \
+ LOG(LS_INFO) << "Feature disabled... skipping"; \
+ return; \
+ }
+
using cricket::FakeVoiceMediaChannel;
using cricket::TransportInfo;
using rtc::SocketAddress;
@@ -1844,6 +1850,7 @@
// Test that we accept an offer with a DTLS fingerprint when DTLS is on
// and that we return an answer with a DTLS fingerprint.
TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SendAudioVideoStream1();
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@@ -1872,6 +1879,7 @@
// Test that we set a local offer with a DTLS fingerprint when DTLS is on
// and then we accept a remote answer with a DTLS fingerprint successfully.
TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SendAudioVideoStream1();
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@@ -1901,6 +1909,7 @@
// Test that if we support DTLS and the other side didn't offer a fingerprint,
// we will fail to set the remote description.
TEST_P(WebRtcSessionTest, TestReceiveNonDtlsOfferWhenDtlsOn) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
cricket::MediaSessionOptions options;
options.recv_video = true;
@@ -1924,6 +1933,7 @@
// Test that we return a failure when applying a local answer that doesn't have
// a DTLS fingerprint when DTLS is required.
TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
@@ -1939,6 +1949,7 @@
// Test that we return a failure when applying a remote answer that doesn't have
// a DTLS fingerprint when DTLS is required.
TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
@@ -3915,6 +3926,8 @@
}
TEST_P(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
configuration_.enable_rtp_data_channel = true;
options_.disable_sctp_data_channels = false;
@@ -3927,6 +3940,7 @@
// Test that sctp_content_name/sctp_transport_name (used for stats) are correct
// before and after BUNDLE is negotiated.
TEST_P(WebRtcSessionTest, SctpContentAndTransportName) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetFactoryDtlsSrtp();
InitWithDtls(GetParam());
@@ -3960,6 +3974,8 @@
}
TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
InitWithDtls(GetParam());
std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -3968,6 +3984,7 @@
}
TEST_P(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
SetFactoryDtlsSrtp();
InitWithDtls(GetParam());
@@ -3999,6 +4016,8 @@
// Test that if DTLS is enabled, we end up with an SctpTransport created
// (and not an RtpDataChannel).
TEST_P(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
InitWithDtls(GetParam());
SetLocalDescriptionWithDataChannel();
@@ -4009,6 +4028,7 @@
// Test that if SCTP is disabled, we don't end up with an SctpTransport
// created (or an RtpDataChannel).
TEST_P(WebRtcSessionTest, TestDisableSctpDataChannels) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
options_.disable_sctp_data_channels = true;
InitWithDtls(GetParam());
@@ -4018,6 +4038,7 @@
}
TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
const int new_send_port = 9998;
const int new_recv_port = 7775;
@@ -4059,6 +4080,8 @@
// WebRtcSession signals the SctpTransport creation request with the expected
// config.
TEST_P(WebRtcSessionTest, TestSctpDataChannelOpenMessage) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
InitWithDtls(GetParam());
SetLocalDescriptionWithDataChannel();
@@ -4098,6 +4121,7 @@
// identity generation is finished (even if a certificate is provided this is
// an async op).
TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
EXPECT_TRUE(session_->waiting_for_certificate_for_testing());
@@ -4113,6 +4137,7 @@
// identity generation is finished (even if a certificate is provided this is
// an async op).
TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@@ -4133,6 +4158,7 @@
// identity generation is finished (even if a certificate is provided this is
// an async op).
TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
@@ -4144,6 +4170,7 @@
// Verifies that CreateOffer fails when CreateOffer is called after async
// identity generation fails.
TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtlsIdentityGenFail();
EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
@@ -4156,6 +4183,7 @@
// before async identity generation is finished.
TEST_P(WebRtcSessionTest,
TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(GetParam(),
CreateSessionDescriptionRequest::kOffer);
}
@@ -4164,6 +4192,7 @@
// before async identity generation fails.
TEST_F(WebRtcSessionTest,
TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
CreateSessionDescriptionRequest::kOffer);
}
@@ -4172,6 +4201,7 @@
// before async identity generation is finished.
TEST_P(WebRtcSessionTest,
TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescription(
GetParam(), CreateSessionDescriptionRequest::kAnswer);
}
@@ -4180,6 +4210,7 @@
// before async identity generation fails.
TEST_F(WebRtcSessionTest,
TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
VerifyMultipleAsyncCreateDescriptionIdentityGenFailure(
CreateSessionDescriptionRequest::kAnswer);
}
@@ -4223,6 +4254,7 @@
// Tests that we can renegotiate new media content with ICE candidates in the
// new remote SDP.
TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();
@@ -4252,6 +4284,7 @@
// Tests that we can renegotiate new media content with ICE candidates separated
// from the remote SDP.
TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
InitWithDtls(GetParam());
SetFactoryDtlsSrtp();