Revert of Removing #defines previously used for building without BoringSSL/OpenSSL.

webrtc/api/peerconnectionendtoend_unittest.cc

 /*
  *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include <memory>
 
 #include "webrtc/api/test/peerconnectiontestwrapper.h"
 // Notice that mockpeerconnectionobservers.h must be included after the above!
 #include "webrtc/api/test/mockpeerconnectionobservers.h"
 #ifdef WEBRTC_ANDROID
 #include "webrtc/api/test/androidtestinitializer.h"
 #endif
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/sslstreamadapter.h"
 #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;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
 
 namespace {
 
 const int kMaxWait = 10000;
 
@@ skipping 148 matching lines @@
   GetAndAddUserMedia();
   Negotiate();
   WaitForCallEstablished();
 }
 #endif  // !defined(ADDRESS_SANITIZER)
 
 #ifdef HAVE_SCTP
 // 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;
   rtc::scoped_refptr<DataChannelInterface> caller_dc(
       caller_->CreateDataChannel("data", init));
   rtc::scoped_refptr<DataChannelInterface> callee_dc(
       callee_->CreateDataChannel("data", init));
 
   Negotiate();
   WaitForConnection();
 
   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
   WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
 
   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
   TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
 
   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
   CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
 }
 
 // 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;
 
   // This DataChannel is for creating the data content in the negotiation.
   rtc::scoped_refptr<DataChannelInterface> dummy(
       caller_->CreateDataChannel("data", init));
   Negotiate();
   WaitForConnection();
 
@@ skipping 11 matching lines @@
 
   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
   TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
 
   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
   CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
 }
 
 // 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;
   rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
       caller_->CreateDataChannel("data", init));
   rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
       callee_->CreateDataChannel("data", init));
 
   Negotiate();
   WaitForConnection();
 
   EXPECT_EQ(1U, caller_dc_1->id() % 2);
   EXPECT_EQ(0U, callee_dc_1->id() % 2);
 
   rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
       caller_->CreateDataChannel("data", init));
   rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
       callee_->CreateDataChannel("data", init));
 
   EXPECT_EQ(1U, caller_dc_2->id() % 2);
   EXPECT_EQ(0U, callee_dc_2->id() % 2);
 }
 
 // Verifies that the message is received by the right remote DataChannel when
 // there are multiple DataChannels.
 TEST_F(PeerConnectionEndToEndTest,
        MessageTransferBetweenTwoPairsOfDataChannels) {
+  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
   CreatePcs();
 
   webrtc::DataChannelInit init;
 
   rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
       caller_->CreateDataChannel("data", init));
   rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
       caller_->CreateDataChannel("data", init));
 
   Negotiate();
@@ skipping 101 matching lines @@
 // Verifies that a DataChannel added from an OPEN message functions after
 // a channel has been previously closed (webrtc issue 3778).
 // This previously failed because the new channel re-uses the ID of the closed
 // channel, and the closed channel was incorrectly still assigned to the id.
 // TODO(deadbeef): This is disabled because there's currently a race condition
 // caused by the fact that a data channel signals that it's closed before it
 // really is. Re-enable this test once that's fixed.
 // 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;
   rtc::scoped_refptr<DataChannelInterface> caller_dc(
       caller_->CreateDataChannel("data", init));
 
   Negotiate();
   WaitForConnection();
 
   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
 
   // Create a new channel and ensure it works after closing the previous one.
   caller_dc = caller_->CreateDataChannel("data2", init);
 
   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
 
   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
 }
 
 // This tests that if a data channel is closed remotely while not referenced
 // by the application (meaning only the PeerConnection contributes to its
 // 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;
   rtc::scoped_refptr<DataChannelInterface> caller_dc(
       caller_->CreateDataChannel("data", init));
 
   Negotiate();
   WaitForConnection();
 
   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
   // This removes the reference to the remote data channel that we hold.
   callee_signaled_data_channels_.clear();
   caller_dc->Close();
   EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
 
   // Wait for a bit longer so the remote data channel will receive the
   // close message and be destroyed.
   rtc::Thread::Current()->ProcessMessages(100);
 }
 #endif  // HAVE_SCTP